2011-04-21 Leif Madsen * Asterisk 1.6.1.25 Released. * AST-2011-005: File Descriptor Resource Exhaustion * AST-2011-006: Asterisk Manager User Shell Access 2011-03-17 Leif Madsen * Asterisk 1.6.1.24 Released. (Asterisk 1.6.1.23 was released a day earlier, but a bug existed in the patch for AST-2011-003 so this is a reissue of that release.) * AST-2011-003: Resource exhaustion in Asterisk Manager Interface * AST-2011-004: Remote crash vulnerability in TCP/TLS server 2011-02-21 Leif Madsen * Asterisk 1.6.1.22 Released. * AST-2011-002: Multiple array overflow and crash vulnerabilities in UDPTL code 2011-01-17 Leif Madsen * Asterisk 1.6.1.21 Released. * AST-2011-001: Stack buffer overflow in SIP channel driver 2010-05-17 Leif Madsen * Asterisk 1.6.1.20 Released * This is the last bug maintenance release for the 1.6.1 branch. For continued support please move to the 1.6.2 branch. This branch will continue to receive security related issues for one additional year. 2010-05-13 Leif Madsen * Asterisk 1.6.1.20-rc2 Released * This will be the last maintenance release in this branch. Please move to the latest 1.6.2.x release for continued issue support. See http://www.asterisk.org/asterisk-versions for more information. 2010-05-12 20:02 +0000 [r262802] Paul Belanger * main/loader.c, main/cli.c, /: Merged revisions 262800 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r262800 | pabelanger | 2010-05-12 15:59:16 -0400 (Wed, 12 May 2010) | 8 lines Notify CLI when modules is loaded / unloaded (closes issue #17308) Reported by: pabelanger Patches: cli.modules.patch uploaded by pabelanger (license 224) Tested by: pabelanger, russell ........ 2010-05-12 18:07 +0000 [r262747] David Vossel * /, apps/app_meetme.c: Merged revisions 262744 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r262744 | dvossel | 2010-05-12 13:01:20 -0500 (Wed, 12 May 2010) | 17 lines Merged revisions 262662 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r262662 | dvossel | 2010-05-12 12:00:04 -0500 (Wed, 12 May 2010) | 11 lines fixes app_meetme dsp error We attempted to detect silence after translating a frame from signed linear. This caused a flooding of errors. To resolve this the code to detect silence was moved before the translation. (closes issue #17133) Reported by: jsdyer ........ ................ 2010-05-12 16:29 +0000 [r262515-262658] Tilghman Lesher * /, apps/app_privacy.c: Merged revisions 262656 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r262656 | tilghman | 2010-05-12 11:23:26 -0500 (Wed, 12 May 2010) | 8 lines Ensure the arguments are initialized. Also miscellaneous CG cleanup. (closes issue #16576) Reported by: uxbod Patches: 20100505__issue16576.diff.txt uploaded by tilghman (license 14) Tested by: uxbod ........ * /, include/asterisk/causes.h: Merged revisions 262513 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r262513 | tilghman | 2010-05-11 16:25:05 -0500 (Tue, 11 May 2010) | 7 lines Move cause 200 to cause 26, as specified in Q.850. Also cleanup the formatting and add a few more that seem like good candidates. (closes issue #16157) Reported by: wimpy ........ 2010-05-11 20:02 +0000 [r262426] Paul Belanger * pbx/pbx_config.c, /: Merged revisions 262419 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r262419 | pabelanger | 2010-05-11 15:40:37 -0400 (Tue, 11 May 2010) | 8 lines Improve logging by displaying line number (closes issue #16303) Reported by: dant Patches: issue16303.patch.v2 uploaded by pabelanger (license 224) Tested by: dant, lmadsen, pabelanger ........ 2010-05-11 19:58 +0000 [r262424] Jason Parker * /, res/Makefile: Merged revisions 262422 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r262422 | qwell | 2010-05-11 14:57:24 -0500 (Tue, 11 May 2010) | 18 lines Merged revisions 262421 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r262421 | qwell | 2010-05-11 14:55:42 -0500 (Tue, 11 May 2010) | 11 lines Use a less silly method for modifying a flex-generated file. The sed syntax that was used wasn't actually valid, causing some versions to choke. This is the method that is used in 1.6.x+ for similar changes. (closes issue #16696) Reported by: bklang Patches: 16696-sedfix.diff uploaded by qwell (license 4) Tested by: qwell ........ ................ 2010-05-11 19:30 +0000 [r262416] Paul Belanger * /, channels/chan_sip.c: Merged revisions 262414 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r262414 | pabelanger | 2010-05-11 15:26:17 -0400 (Tue, 11 May 2010) | 8 lines Improve logging information for misconfigured contexts (closes issue #17238) Reported by: pprindeville Patches: chan_sip-bug17238.patch uploaded by pprindeville (license 347) Tested by: pprindeville ........ 2010-05-11 17:25 +0000 [r262339] Tilghman Lesher * apps/app_voicemail.c, /, Makefile.rules: Merged revisions 262330 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r262330 | tilghman | 2010-05-11 12:23:51 -0500 (Tue, 11 May 2010) | 9 lines Merged revisions 262321 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r262321 | tilghman | 2010-05-11 12:22:07 -0500 (Tue, 11 May 2010) | 2 lines Fix issue #17302 a slightly different way (mad props to Qwell) ........ ................ 2010-05-10 18:52 +0000 [r262238] David Vossel * /, channels/chan_console.c: Merged revisions 262236 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r262236 | dvossel | 2010-05-10 13:36:10 -0500 (Mon, 10 May 2010) | 11 lines fixes crash in chan_console There is a race condition between console_hangup() and start_stream(). It is possible for console_hangup() to be called and then the stream thread to begin after the hangup. To avoid this a check in start_stream() to make sure the pvt-owner still exists while the pvt lock is held is made. If the owner is gone that means the channel hung up and start_stream should be aborted. ........ 2010-05-10 16:38 +0000 [r262154] Tilghman Lesher * /, Makefile.rules: Merged revisions 262152 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r262152 | tilghman | 2010-05-10 11:36:25 -0500 (Mon, 10 May 2010) | 17 lines Merged revisions 262151 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r262151 | tilghman | 2010-05-10 11:34:21 -0500 (Mon, 10 May 2010) | 10 lines Allow compilation on Mac OS X 10.4 (Tiger) (closes issue #17297) Reported by: jcovert Patches: 20100506__issue17297.diff.txt uploaded by tilghman (license 14) (closes issue #17302) Reported by: jcovert ........ ................ 2010-05-09 02:17 +0000 [r261915-262104] Tilghman Lesher * autoconf/ast_ext_lib.m4, autoconf/ast_c_compile_check.m4, autoconf/ast_c_define_check.m4, /, configure, include/asterisk/autoconfig.h.in: Merged revisions 262102 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r262102 | tilghman | 2010-05-08 21:14:04 -0500 (Sat, 08 May 2010) | 5 lines Cleanup a bit more by getting rid of useless version defines. Also make library detection use passed CFLAGS. (closes issue #17309) Reported by: stuarth ........ * /, configure, configure.ac: Merged revisions 262048 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r262048 | tilghman | 2010-05-07 21:40:01 -0500 (Fri, 07 May 2010) | 2 lines Use CPPFLAGS to pass PTHREAD_CFLAGS for vpb only ........ * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 261913 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r261913 | tilghman | 2010-05-07 15:35:17 -0500 (Fri, 07 May 2010) | 14 lines Use the detected pthread building flags in every place, instead of hardcoding -lpthread. We nicely detect the right flags on each system for building Asterisk with pthreads, then ignore it for every other build option that requires us to build with pthreads. This caused some items to return a false negative. Also cleanup some minor naming issues that caused "library library" redundancy in the output. (closes issue #17303) Reported by: stuarth Patches: 20100507__issue17303.diff.txt uploaded by tilghman (license 14) Tested by: stuarth ........ 2010-05-06 20:13 +0000 [r261738] Jeff Peeler * apps/app_voicemail.c, /: Merged revisions 261736 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r261736 | jpeeler | 2010-05-06 15:11:53 -0500 (Thu, 06 May 2010) | 15 lines Merged revisions 261735 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r261735 | jpeeler | 2010-05-06 15:10:59 -0500 (Thu, 06 May 2010) | 8 lines Only allow the operator key to be accepted after leaving a voicemail. Or rather disallow the operator key from being accepted when not offered, such as after finishing a recording from within the mailbox options menu. ABE-2121 SWP-1267 ........ ................ 2010-05-06 17:08 +0000 [r261611] Jason Parker * sounds/Makefile, /: Merged revisions 261609 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r261609 | qwell | 2010-05-06 12:06:40 -0500 (Thu, 06 May 2010) | 11 lines Merged revisions 261608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r261608 | qwell | 2010-05-06 11:56:02 -0500 (Thu, 06 May 2010) | 4 lines Use the versioned MOH tarballs, now that we have them. This makes for more reproducibility. Prompted by a discussion in #asterisk-dev ........ ................ 2010-05-06 15:42 +0000 [r261562] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 261560 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r261560 | tilghman | 2010-05-06 10:39:10 -0500 (Thu, 06 May 2010) | 8 lines Permit more lines within a SIP body to be parsed. The example given within the related issue showed 120 lines, which was mostly a result of the body being XML. (closes issue #17179) Reported by: khw ........ 2010-05-06 Leif Madsen * Asterisk 1.6.1.20-rc1 Released 2010-05-06 14:02 +0000 [r261497] Russell Bryant * /, main/heap.c: Merged revisions 261496 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r261496 | russell | 2010-05-06 08:58:07 -0500 (Thu, 06 May 2010) | 40 lines Fix handling of removing nodes from the middle of a heap. This bug surfaced in 1.6.2 and does not affect code in any other released version of Asterisk. It manifested itself as SIP qualify not happening when it should, causing peers to go unreachable. This was debugged down to scheduler entries sometimes not getting executed when they were supposed to, which was in turn caused by an error in the heap code. The problem only sometimes occurs, and it is due to the logic for removing an entry in the heap from an arbitrary location (not just popping off the top). The scheduler performs this operation frequently when entries are removed before they run (when ast_sched_del() is used). In a normal pop off of the top of the heap, a node is taken off the bottom, placed at the top, and then bubbled down until the max heap property is restored (see max_heapify()). This same logic was used for removing an arbitrary node from the middle of the heap. Unfortunately, that logic is full of fail. This patch fixes that by fully restoring the max heap property when a node is thrown into the middle of the heap. Instead of just pushing it down as appropriate, it first pushes it up as high as it will go, and _then_ pushes it down. Lastly, fix a minor problem in ast_heap_verify(), which is only used for debugging. If a parent and child node have the same value, that is not an error. The only error is if a parent's value is less than its children. A huge thanks goes out to cappucinoking for debugging this down to the scheduler, and then producing an ast_heap test case that demonstrated the breakage. That made it very easy for me to focus on the heap logic and produce a fix. Open source projects are awesome. (closes issue #16936) Reported by: ib2 Tested by: cappucinoking, crjw (closes issue #17277) Reported by: cappucinoking Patches: heap-fix.rev2.diff uploaded by russell (license 2) Tested by: cappucinoking, russell ........ 2010-05-06 07:35 +0000 [r261452] Tzafrir Cohen * channels/chan_dahdi.c, /: Merged revisions 261451 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r261451 | tzafrir | 2010-05-06 10:27:31 +0300 (ה', 06 מאי 2010) | 4 lines When failing to configure, don't destroy 'cfg' twice Fixes a crash when some config section had an incorrect channel config. ........ 2010-05-05 19:14 +0000 [r261317] Paul Belanger * /, channels/chan_sip.c: Merged revisions 261314 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r261314 | pabelanger | 2010-05-05 14:43:03 -0400 (Wed, 05 May 2010) | 19 lines Merged revisions 261274 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r261274 | pabelanger | 2010-05-05 12:42:22 -0400 (Wed, 05 May 2010) | 12 lines Registration fix for SIP realtime. Make sure realtime fields are not empty. (closes issue #17266) Reported by: Nick_Lewis Patches: chan_sip.c-realtime.patch uploaded by Nick Lewis (license 657) Tested by: Nick_Lewis, sberney Review: https://reviewboard.asterisk.org/r/643/ ........ ................ 2010-05-04 23:55 +0000 [r261097] Tilghman Lesher * main/channel.c, /: Merged revisions 261095 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r261095 | tilghman | 2010-05-04 18:51:52 -0500 (Tue, 04 May 2010) | 18 lines Merged revisions 261093-261094 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r261093 | tilghman | 2010-05-04 18:36:53 -0500 (Tue, 04 May 2010) | 7 lines Protect against overflow, when calculating how long to wait for a frame. (closes issue #17128) Reported by: under Patches: d.diff uploaded by under (license 914) ........ r261094 | tilghman | 2010-05-04 18:47:08 -0500 (Tue, 04 May 2010) | 2 lines Add a tiny corner case to the previous commit ........ ................ 2010-05-04 18:57 +0000 [r260926] Jeff Peeler * apps/app_voicemail.c, /: Merged revisions 260924 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r260924 | jpeeler | 2010-05-04 13:51:28 -0500 (Tue, 04 May 2010) | 18 lines Merged revisions 260923 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260923 | jpeeler | 2010-05-04 13:46:46 -0500 (Tue, 04 May 2010) | 12 lines Voicemail transfer to operator should occur immediately, not after main menu. There were two scenarios in the advanced options that while using the operator=yes and review=yes options, the transfer occurred only after exiting the main menu (after sending a reply or leaving a message for an extension). Now after the audio is processed for the reply or message the transfer occurs immediately as expected. ABE-2107 ABE-2108 ........ ................ 2010-05-04 15:51 +0000 [r260745-260804] Jason Parker * /, build_tools/make_build_h: Merged revisions 260802 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r260802 | qwell | 2010-05-04 10:49:57 -0500 (Tue, 04 May 2010) | 9 lines Merged revisions 260801 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260801 | qwell | 2010-05-04 10:49:27 -0500 (Tue, 04 May 2010) | 1 line Fix fallout from removing from configure script. Pointed out by philipp64 on #asterisk-dev ........ ................ * /: Fix merge props 2010-05-03 17:37 +0000 [r260741] Paul Belanger * Makefile, /: Merged revisions 260661-260662 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260661 | pabelanger | 2010-05-03 12:41:30 -0400 (Mon, 03 May 2010) | 10 lines non-root make install PREFIX=/tmp fails. Prepend libdir when executing mkpkgconfig allowing non-root installs to work. (closes issue #17268) Reported by: pabelanger Patches: issue17268.patch uploaded by pabelanger (license 224) Tested by: pabelanger ........ r260662 | pabelanger | 2010-05-03 12:54:41 -0400 (Mon, 03 May 2010) | 3 lines Should have removed /usr/lib/ part. Thanks Qwell. ........ 2010-05-03 14:59 +0000 [r260572] Leif Madsen * doc/HOWTO_collect_debug_information.txt: Merged revisions 260570 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r260570 | lmadsen | 2010-05-03 09:58:23 -0500 (Mon, 03 May 2010) | 9 lines Merged revisions 260569 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260569 | lmadsen | 2010-05-03 09:57:39 -0500 (Mon, 03 May 2010) | 1 line Minor typo pointed out by pabelanger on IRC. ........ ................ 2010-04-30 22:47 +0000 [r260440] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 260437 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r260437 | jpeeler | 2010-04-30 17:36:49 -0500 (Fri, 30 Apr 2010) | 18 lines Merged revisions 260434 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260434 | jpeeler | 2010-04-30 17:22:46 -0500 (Fri, 30 Apr 2010) | 11 lines Ensure channel state is not incorrectly set in the case of a very early answer. The needringing bit was being read in dahdi_read after answering thereby setting the state to ringing from up. This clears needringing upon answering so that is no longer possible. (closes issue #17067) Reported by: tzafrir Patches: needringing.diff uploaded by tzafrir (license 46) ........ ................ 2010-04-30 20:18 +0000 [r260354] Mark Michelson * /, res/res_musiconhold.c: Merged revisions 260346 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r260346 | mmichelson | 2010-04-30 15:11:02 -0500 (Fri, 30 Apr 2010) | 24 lines Merged revisions 260345 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260345 | mmichelson | 2010-04-30 15:08:15 -0500 (Fri, 30 Apr 2010) | 18 lines Fix potential crash from race condition due to accessing channel data without the channel locked. In res_musiconhold.c, there are several places where a channel's stream's existence is checked prior to calling ast_closestream on it. The issue here is that in several cases, the channel was not locked while checking the stream. The result was that if two threads checked the state of the channel's stream at approximately the same time, then there could be a situation where both threads attempt to call ast_closestream on the channel's stream. The result here is that the refcount for the stream would go below 0, resulting in a crash. I have added proper channel locking to res_musiconhold.c to ensure that we do not try to check chan->stream without the channel locked. A Digium customer has been using this patch for several weeks and has not had any crashes since applying the patch. ABE-2147 ........ ................ 2010-04-29 23:03 +0000 [r260233] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 260231 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r260231 | rmudgett | 2010-04-29 17:44:14 -0500 (Thu, 29 Apr 2010) | 33 lines Merged revisions 260195 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260195 | rmudgett | 2010-04-29 17:11:47 -0500 (Thu, 29 Apr 2010) | 26 lines DTMF CallerID detection problems. The code handling DTMF CallerID drops digits on long CallerID numbers and may timeout waiting for the first ring with shorter numbers. The DTMF emulation mode was not turned off when processing DTMF CallerID. When the emulation code gets behind in processing the DTMF digits it can skip a digit. For shorter numbers, the timeout may have been too short. I increased it from 2 seconds to 4 seconds. Four seconds is a typical time between rings for many countries. (closes issue #16460) Reported by: sum Patches: issue16460.patch uploaded by rmudgett (license 664) issue16460_v1.6.2.patch uploaded by rmudgett (license 664) Tested by: sum, rmudgett Review: https://reviewboard.asterisk.org/r/634/ JIRA SWP-562 JIRA AST-334 JIRA SWP-901 ........ ................ 2010-04-29 18:18 +0000 [r260154] Tilghman Lesher * configs/extensions.conf.sample, /: Merged revisions 260148 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r260148 | tilghman | 2010-04-29 13:15:57 -0500 (Thu, 29 Apr 2010) | 2 lines Pattern match fail. ........ 2010-04-29 15:37 +0000 [r260052] David Vossel * /, include/asterisk/audiohook.h, main/audiohook.c: Merged revisions 260050 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r260050 | dvossel | 2010-04-29 10:33:27 -0500 (Thu, 29 Apr 2010) | 21 lines Merged revisions 260049 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260049 | dvossel | 2010-04-29 10:31:02 -0500 (Thu, 29 Apr 2010) | 14 lines Fixes crash in audiohook_write_list The middle_frame in the audiohook_write_list function was being freed if a audiohook manipulator returned a failure. This is incorrect logic. This patch resolves this and adds detailed descriptions of how this function should work and why manipulator failures must be ignored. (closes issue #17052) Reported by: dvossel Tested by: dvossel (closes issue #16196) Reported by: atis Review: https://reviewboard.asterisk.org/r/623/ ........ ................ 2010-04-28 22:35 +0000 [r259958] Mark Michelson * /, channels/chan_sip.c: Merged revisions 259957 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r259957 | mmichelson | 2010-04-28 17:34:15 -0500 (Wed, 28 Apr 2010) | 11 lines Don't override peer context with domain context. (closes issue #17040) Reported by: pprindeville Patches: asterisk-1.6-bugid17040.patch uploaded by pprindeville (license 347) Tested by: pprindeville Review: https://reviewboard.asterisk.org/r/565/ ........ 2010-04-28 21:33 +0000 [r259930] David Vossel * main/channel.c, channels/chan_local.c, /: Merged revisions 259870 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259870 | dvossel | 2010-04-28 16:20:03 -0500 (Wed, 28 Apr 2010) | 39 lines Merged revisions 259858 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259858 | dvossel | 2010-04-28 16:16:03 -0500 (Wed, 28 Apr 2010) | 33 lines resolves deadlocks in chan_local Issue_1. In the local_hangup() 3 locks must be held at the same time... pvt, pvt->chan, and pvt->owner. Proper deadlock avoidance is done when the channel to hangup is the outbound chan_local channel, but when it is not the outbound channel we have an issue... We attempt to do deadlock avoidance only on the tech pvt, when both the tech pvt and the pvt->owner are locked coming into that loop. By never giving up the pvt->owner channel deadlock avoidance is not entirely possible. This patch resolves that by doing deadlock avoidance on both the pvt->owner and the pvt when trying to get the pvt->chan lock. Issue_2. ast_prod() is used in ast_activate_generator() to queue a frame on the channel and make the channel's read function get called. This function is used in ast_activate_generator() while the channel is locked, which mean's the channel will have a lock both from the generator code and the frame_queue code by the time it gets to chan_local.c's local_queue_frame code... local_queue_frame contains some of the same crazy deadlock avoidance that local_hangup requires, and this recursive lock prevents that deadlock avoidance from happening correctly. This patch removes ast_prod() from the channel lock so only one lock is held during the local_queue_frame function. (closes issue #17185) Reported by: schmoozecom Patches: issue_17185_v1.diff uploaded by dvossel (license 671) issue_17185_v2.diff uploaded by dvossel (license 671) Tested by: schmoozecom, GameGamer43 Review: https://reviewboard.asterisk.org/r/631/ ........ ................ 2010-04-28 21:09 +0000 [r259855] Leif Madsen * config.guess: Merged revisions 259853 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259853 | lmadsen | 2010-04-28 16:08:34 -0500 (Wed, 28 Apr 2010) | 14 lines Merged revisions 259852 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259852 | lmadsen | 2010-04-28 16:07:48 -0500 (Wed, 28 Apr 2010) | 6 lines Update config.guess. Updating config.guess because after installing Ubuntu Server 9.10 and running all the update scripts, running ./configure would not continue because it was unable to determine what kind of system I had. After updating config.guess things started working again. ........ ................ 2010-04-28 20:33 +0000 [r259776-259850] Jason Parker * /, configure, configure.ac: Merged revisions 259848 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259848 | qwell | 2010-04-28 15:32:14 -0500 (Wed, 28 Apr 2010) | 9 lines Merged revisions 259847 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259847 | qwell | 2010-04-28 15:30:21 -0500 (Wed, 28 Apr 2010) | 1 line Add AC_CONFIG_AUX_DIR to configure script, so systems without install can use install-sh from our source dir. ........ ................ * makeopts.in, /: Merged revisions 259837 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259837 | qwell | 2010-04-28 15:26:35 -0500 (Wed, 28 Apr 2010) | 9 lines Merged revisions 259833 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259833 | qwell | 2010-04-28 15:25:36 -0500 (Wed, 28 Apr 2010) | 1 line Missed this when removing $ID ........ ................ * Makefile, /, configure, configure.ac: Merged revisions 259760 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259760 | qwell | 2010-04-28 14:19:54 -0500 (Wed, 28 Apr 2010) | 14 lines Merged revisions 259748 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259748 | qwell | 2010-04-28 14:17:38 -0500 (Wed, 28 Apr 2010) | 7 lines Remove usage of `id` since it isn't useful and was causing breakge. Solaris `id` doesn't support the -u argument. Instead of figuring out how to fix this to work on Solaris, I decided to check why it was necessary and where else it was used. It was only used in one place, and it hasn't been needed for a very long time (I question whether it was ever needed). ........ ................ 2010-04-28 17:19 +0000 [r259679] Jeff Peeler * apps/app_voicemail.c, /: Merged revisions 259672 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259672 | jpeeler | 2010-04-28 12:18:43 -0500 (Wed, 28 Apr 2010) | 11 lines Merged revisions 259664 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259664 | jpeeler | 2010-04-28 12:13:29 -0500 (Wed, 28 Apr 2010) | 4 lines Do not play goodbye prompt after timeout of message review. ABE-2124 ........ ................ 2010-04-27 22:37 +0000 [r259615] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 259538 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259538 | rmudgett | 2010-04-27 17:18:09 -0500 (Tue, 27 Apr 2010) | 18 lines Merged revisions 259531 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259531 | rmudgett | 2010-04-27 16:53:07 -0500 (Tue, 27 Apr 2010) | 11 lines DAHDI "WARNING" message is confusing and vague "WARNING[28406]: chan_dahdi.c:6873 ss_thread: CallerID feed failed: Success" Changed the warning to "Failed to decode CallerID on channel 'name'". The message before it is likely more specific about why the CallerID decode failed. SWP-501 AST-283 ........ ................ 2010-04-27 21:50 +0000 [r259529] Leif Madsen * sounds/Makefile: Merged revisions 259527 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259527 | lmadsen | 2010-04-27 16:49:36 -0500 (Tue, 27 Apr 2010) | 23 lines Merged revisions 259526 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259526 | lmadsen | 2010-04-27 16:48:47 -0500 (Tue, 27 Apr 2010) | 15 lines Update sounds files. * Add additional sounds prompts for say_enumeration * Update the English conference sounds prompts so they are better quality and all sound more consistent * Clean up the core-sounds-XX.txt and extra-sounds-XX.txt files to include all present sound files Both core (en, fr, es) and extra (en, fr) sounds files have been updated. (closes issue #16200) Reported by: murf (closes issue #17137) Reported by: lmadsen ........ ................ 2010-04-27 21:22 +0000 [r259355-259471] Jason Parker * /, main/editline/configure, main/editline/Makefile.in, main/editline/configure.in: Merged revisions 259439 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r259439 | qwell | 2010-04-27 16:13:01 -0500 (Tue, 27 Apr 2010) | 5 lines Add gar to the check for AR for those silly OSes (Solaris) that don't have ar. autoconf2.13 couldn't handle AC_PROG_GREP, so I removed it. This is fine, since we don't need to use anything that the configure script doesn't. ........ * /, configure, configure.ac: Merged revisions 259353 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259353 | qwell | 2010-04-27 14:31:55 -0500 (Tue, 27 Apr 2010) | 12 lines Merged revisions 259352 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259352 | qwell | 2010-04-27 14:29:26 -0500 (Tue, 27 Apr 2010) | 5 lines Support the silly OSes that don't have ar and strip. Since AC_PATH_TOOL is equiv to AC_CHECK_TOOL when path isn't specified, and AC_PATH_TOOLS doesn't exist, we'll just switch to AC_CHECK_TOOLS. ........ ................ 2010-04-27 18:53 +0000 [r259309] Richard Mudgett * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged revisions 259307 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259307 | rmudgett | 2010-04-27 13:29:33 -0500 (Tue, 27 Apr 2010) | 21 lines Merged revisions 259270 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259270 | rmudgett | 2010-04-27 13:14:54 -0500 (Tue, 27 Apr 2010) | 14 lines hidecalleridname parameter in chan_dahdi.conf Issue #7321 implements a new chan_dahdi configuration option. However, a change mentioned in the issue was never implemented. This is the change that will allow the feature to work. I added a note to chan_dahdi.conf.sample about the feature. (closes issue #17143) Reported by: djensen99 Patches: diff.txt uploaded by djensen99 (license NA) (One line change) Tested by: djensen99 ........ ................ 2010-04-26 21:48 +0000 [r259078-259108] Mark Michelson * main/channel.c, /: Merged revisions 259105 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259105 | mmichelson | 2010-04-26 16:45:13 -0500 (Mon, 26 Apr 2010) | 9 lines Merged revisions 259104 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259104 | mmichelson | 2010-04-26 16:44:43 -0500 (Mon, 26 Apr 2010) | 3 lines Let compilation succeed warning-free when DONT_OPTIMIZE is turned off. ........ ................ * main/channel.c, /: Merged revisions 259023 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r259023 | mmichelson | 2010-04-26 16:13:35 -0500 (Mon, 26 Apr 2010) | 19 lines Merged revisions 259018 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259018 | mmichelson | 2010-04-26 16:03:08 -0500 (Mon, 26 Apr 2010) | 13 lines Prevent Newchannel manager events for dummy channels. No Newchannel manager event will be fired for channels that are allocated to not match a registered technology type. Thus bogus channels allocated solely for variable substitution or CDR operations do not result in a Newchannel event. (closes issue #16957) Reported by: atis Review: https://reviewboard.asterisk.org/r/601 ........ ................ 2010-04-25 18:14 +0000 [r258778] Tilghman Lesher * res/res_monitor.c, /: Merged revisions 258776 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r258776 | tilghman | 2010-04-25 13:12:14 -0500 (Sun, 25 Apr 2010) | 13 lines Merged revisions 258775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r258775 | tilghman | 2010-04-25 13:09:05 -0500 (Sun, 25 Apr 2010) | 6 lines When StopMonitor is called, ensure that it will not be restarted by a channel event. (closes issue #16590) Reported by: kkm Patches: resmonitor-16590-trunk.239289.diff uploaded by kkm (license 888) ........ ................ 2010-04-22 22:24 +0000 [r258705] Matthew Nicholson * main/cdr.c, main/channel.c, /, main/features.c: Merged revisions 258671,258675 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r258671 | mnicholson | 2010-04-22 16:57:59 -0500 (Thu, 22 Apr 2010) | 32 lines Merged revisions 193391,258670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193391 | mnicholson | 2009-05-08 16:01:25 -0500 (Fri, 08 May 2009) | 8 lines Set the proper disposition on originated calls. (closes issue #14167) Reported by: jpt Patches: call-file-missing-cdr2.diff uploaded by mnicholson (license 96) Tested by: dlotina, rmartinez, mnicholson ........ r258670 | mnicholson | 2010-04-22 16:49:07 -0500 (Thu, 22 Apr 2010) | 11 lines Fix broken CDR behavior. This change allows a CDR record previously marked with disposition ANSWERED to be set as BUSY or NO ANSWER. Additionally this change partially reverts r235635 and does not set the AST_CDR_FLAG_ORIGINATED flag on CDRs generated from ast_call(). To preserve proper CDR behavior, the AST_CDR_FLAG_DIALED flag is now cleared from all brige CDRs in ast_bridge_call(). (closes issue #16797) Reported by: VarnishedOtter Tested by: mnicholson ........ (closes issue #16222) Reported by: telles Tested by: mnicholson ................ r258675 | mnicholson | 2010-04-22 17:11:23 -0500 (Thu, 22 Apr 2010) | 2 lines Fix previous commit. ................ 2010-04-21 22:10 +0000 [r258435] Jeff Peeler * apps/app_voicemail.c, /: Merged revisions 258433 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r258433 | jpeeler | 2010-04-21 16:56:09 -0500 (Wed, 21 Apr 2010) | 15 lines Merged revisions 258432 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r258432 | jpeeler | 2010-04-21 16:45:36 -0500 (Wed, 21 Apr 2010) | 8 lines Fix looping forever when no input received in certain voicemail menu scenarios. Specifically, prompting for an extension (when leaving or forwarding a message) or when prompting for a digit (when saving a message or changing folders). ABE-2122 SWP-1268 ........ ................ 2010-04-21 18:23 +0000 [r258334] David Vossel * /, channels/chan_sip.c: Merged revisions 258305 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r258305 | dvossel | 2010-04-21 13:13:36 -0500 (Wed, 21 Apr 2010) | 12 lines fixes issue with double "sip:" in header field This is a clear mistake in logic. Future discussions about how to avoid having to handle uri's like this should take place in the future, but this fix needs to go in for now. (closes issue #15847) Reported by: ebroad Patches: doublesip.patch uploaded by ebroad (license 878) ........ 2010-04-20 17:51 +0000 [r258105] Jeff Peeler * apps/app_voicemail.c, /: Merged revisions 258065 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r258065 | jpeeler | 2010-04-20 12:06:19 -0500 (Tue, 20 Apr 2010) | 17 lines Merged revisions 258029 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r258029 | jpeeler | 2010-04-20 11:16:33 -0500 (Tue, 20 Apr 2010) | 11 lines Play correct prompt when voicemail store failure occurs after attempted forward. If a user's mailbox was full and a message was attempted to be forwarded to said box, warnings on the console would indicate failure. However, the played prompt was that of success (vm-msgsaved). Now storage failure is taken into account and the correct prompt (vm-mailboxfull) is played when appropriate. ABE-2123 SWP-1262 ........ ................ 2010-04-19 18:22 +0000 [r257833] Terry Wilson * /, main/features.c: Merged revisions 257810 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r257810 | twilson | 2010-04-19 12:57:41 -0500 (Mon, 19 Apr 2010) | 5 lines Fix incomplete CDR merge from r195881 Because res/res_features.c was removed and main/cdr.c added, these changes didn't make it to trunk and the 1.6.x branches ........ 2010-04-18 17:28 +0000 [r257770] Tilghman Lesher * configs/cdr_odbc.conf.sample, /: Merged revisions 257768 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r257768 | tilghman | 2010-04-18 12:25:53 -0500 (Sun, 18 Apr 2010) | 2 lines Removing unused configuration parameters ........ 2010-04-16 21:32 +0000 [r257739] Dwayne M. Hubbard * apps/app_mixmonitor.c, /: Merged revisions 257713 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r257713 | dhubbard | 2010-04-16 16:22:30 -0500 (Fri, 16 Apr 2010) | 28 lines Merged revisions 257686 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257686 | dhubbard | 2010-04-16 16:15:43 -0500 (Fri, 16 Apr 2010) | 21 lines Make the mixmonitor thread process audio frames faster Mantis issue 17078 reports MixMonitor recordings have shorter durations than the call duration. This was because the mixmonitor thread was not processing frames from the audiohook fast enough. The mixmonitor thread would slowly fall behind the most recent audio frame and when the channel hangs up, the mixmonitor thread would exit without processing the same number of frames as the channel; leaving the mixmonitor recording shorter than actual call duration. This revision fixes this issue by moving the ast_audiohook_trigger_wait() and the subsequent audiohook.status check into the block where the ast_audiohook_read_frame() function returns NULL. (closes issue #17078) Reported by: geoff2010 Patches: dw-M17078.patch uploaded by dhubbard (license 733) Tested by: dhubbard, geoff2010 Review: https://reviewboard.asterisk.org/r/611/ ........ ................ 2010-04-15 21:33 +0000 [r257508-257594] Tilghman Lesher * include/asterisk/app.h, /, main/app.c: Merged revisions 257560 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r257560 | tilghman | 2010-04-15 16:26:19 -0500 (Thu, 15 Apr 2010) | 13 lines Merged revisions 257544 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257544 | tilghman | 2010-04-15 16:23:24 -0500 (Thu, 15 Apr 2010) | 6 lines Allow application options with arguments to contain parentheses, through a variety of escaping techniques. Fixes SWP-1194 (ABE-2143). Review: https://reviewboard.asterisk.org/r/604/ ........ ................ * /, channels/chan_sip.c: Merged revisions 257493 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r257493 | tilghman | 2010-04-15 15:30:15 -0500 (Thu, 15 Apr 2010) | 20 lines Merged revisions 257467 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257467 | tilghman | 2010-04-15 15:24:50 -0500 (Thu, 15 Apr 2010) | 13 lines Don't recreate peer, when responding to a repeated deregistration attempt. When a reply to a deregistration is lost in transmit, the client retries the deregistration. Previously, this would cause a realtime/autocreate peer to be loaded back into memory, after it had already been correctly purged. Instead, we just want to resend the reply without loading the peer. (closes issue #16908) Reported by: kkm Patches: 20100412__issue16908.diff.txt uploaded by tilghman (license 14) Tested by: kkm ........ ................ 2010-04-15 19:43 +0000 [r257345-257429] Leif Madsen * doc/backtrace.txt: Merged revisions 257427 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r257427 | lmadsen | 2010-04-15 14:41:05 -0500 (Thu, 15 Apr 2010) | 21 lines Merged revisions 257426 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257426 | lmadsen | 2010-04-15 14:40:33 -0500 (Thu, 15 Apr 2010) | 13 lines Update backtrace.txt documentation. Update the backtrace.txt documentation so it conforms to the same layout as other documents we've been working on recently. Additionally, add a bunch of new information about gathering backtraces for crashes and deadlocks, along with ways of verifying your file before uploading it. Create a couple of one line commands for people to generate the files we need. (closes issue #17190) Reported by: lmadsen Patches: backtrace.txt.patch-2 uploaded by lmadsen (license 10) Tested by: lmadsen, pabelanger ........ ................ * doc/backtrace.txt: Merged revisions 257343 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r257343 | lmadsen | 2010-04-15 08:44:38 -0500 (Thu, 15 Apr 2010) | 9 lines Merged revisions 257342 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257342 | lmadsen | 2010-04-15 08:41:45 -0500 (Thu, 15 Apr 2010) | 1 line Update address of the bug tracker. ........ ................ 2010-04-14 23:00 +0000 [r257264] Tilghman Lesher * configs/features.conf.sample, /, main/features.c: Merged revisions 257262 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r257262 | tilghman | 2010-04-14 17:57:35 -0500 (Wed, 14 Apr 2010) | 15 lines Yet another issue where the conversion of the application delimiter to comma caused an issue. Application arguments within the feature map could possibly contain a comma, which conflicts with the syntax of the features.conf configuration file. This patch allows the argument to be wrapped in parentheses or quoted, to allow the application arguments to be interpreted as a single configuration parameter. (closes issue #16646) Reported by: pinga-fogo Patches: 20100414__issue16646.diff.txt uploaded by tilghman (license 14) Tested by: tilghman Review: https://reviewboard.asterisk.org/r/547/ ........ 2010-04-13 19:30 +0000 [r257214] Matthew Nicholson * main/manager.c, /, configs/manager.conf.sample: Merged revisions 257146 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r257146 | mnicholson | 2010-04-13 13:10:30 -0500 (Tue, 13 Apr 2010) | 16 lines Merged revisions 257070 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257070 | mnicholson | 2010-04-13 11:46:30 -0500 (Tue, 13 Apr 2010) | 9 lines Add an option to restore past broken behavor of the Events manager action Before r238915, certain values for the EventMask parameter of the Events action would result in no response being returned. This patch adds an option to restore that broken behavior. Also while fixing this bug I discovered that passing an empty EventMasks parameter would also result in no response being returned, this has been fixed as well while being preserved when the broken behavior is requested. (closes issue #17023) Reported by: nblasgen Review: https://reviewboard.asterisk.org/r/602/ ........ ................ 2010-04-13 19:20 +0000 [r257067-257208] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 257191 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r257191 | tilghman | 2010-04-13 14:17:48 -0500 (Tue, 13 Apr 2010) | 10 lines Also unref the pvt when we delete the provisional keepalive job. (closes issue #16774) Reported by: kowalma Patches: 20100315__issue16774.diff.txt uploaded by tilghman (license 14) Tested by: falves11, jamicque Review: https://reviewboard.asterisk.org/r/591/ ........ * cdr/cdr_sqlite3_custom.c, /: Merged revisions 257065 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r257065 | tilghman | 2010-04-13 11:33:21 -0500 (Tue, 13 Apr 2010) | 8 lines Ensure that we can have commas within cdr values. (closes issue #17001) Reported by: snuffy Patches: 20100412__issue17001.diff.txt uploaded by tilghman (license 14) Tested by: snuffy ........ 2010-04-12 17:31 +0000 [r256903] Leif Madsen * doc/HOWTO_collect_debug_information.txt (added): Merged revisions 256901 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r256901 | lmadsen | 2010-04-12 12:29:53 -0500 (Mon, 12 Apr 2010) | 23 lines Merged revisions 256900 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r256900 | lmadsen | 2010-04-12 12:29:26 -0500 (Mon, 12 Apr 2010) | 15 lines Add How-To document on collecting debugging info for issues.asterisk.org Paul Belanger has been helping a lot with bug tracking recently and created this document that we can now point to when additional debugging information is required. This document will help those filing issues to know how to get the information required when filing their issues. This will make things easier on the developers. Initial text and changes by pabelanger. Tweaks and editing by myself. (closes issue #17159) Reported by: pabelanger Patches: HOWTO_collect_debug_information.txt.patch uploaded by lmadsen (license 10) Tested by: tzafrir, pabelanger, lmadsen ........ ................ 2010-04-06 19:40 +0000 [r256372] Tilghman Lesher * /, configure, include/asterisk/autoconfig.h.in, configure.ac, include/asterisk/lock.h: Merged revisions 256370 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r256370 | tilghman | 2010-04-06 14:28:42 -0500 (Tue, 06 Apr 2010) | 2 lines Mac OS X does not support comparing a mutex to its initializer. Create a test for this. ........ 2010-04-06 18:08 +0000 [r256267-256365] Richard Mudgett * channels/chan_dahdi.c: Fix malformed if test. Regression of issue 15883. Converted if statement to a switch statement for clarity. * channels/chan_dahdi.c, /: Merged revisions 256265 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r256265 | rmudgett | 2010-04-05 19:39:44 -0500 (Mon, 05 Apr 2010) | 12 lines Merged revisions 256225 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r256225 | rmudgett | 2010-04-05 19:10:16 -0500 (Mon, 05 Apr 2010) | 5 lines DAHDI/PRI call to pri_channel_bridge() not protected by PRI lock. SWP-1231 ABE-2163 ........ ................ 2010-05-03 Leif Madsen * Asterisk 1.6.1.19 Released 2010-04-29 Leif Madsen * Asterisk 1.6.1.19-rc3 Released 2010-04-29 10:31 +0000 [r260052] David Vossel * include/asterisk/audiohook.h, main/audiohook.c: Fixes crash in audiohook_write_list. (closes issue 0017052) Reported by: dvossel Tested by: dvossel. (closes issue 0016196) Reported by: atis. Review: https://reviewboard.asterisk.org/r/623/ 2010-04-28 10:31 +0000 [r259930] David Vossel * channels/chan_local.c, main/channel.c: Resolves deadlocks in chan_local. (closes issue 0017185) Reported by: schmoozecom Patches: issue_17185_v1.diff uploaded by dvossel (license 671) issue_17185_v2.diff uploaded by dvossel (license 671) Tested by: schmoozecom, GameGamer43 Review: https://reviewboard.asterisk.org/r/631/ 2010-04-13 Leif Madsen * Asterisk 1.6.1.19-rc2 Released 2010-04-13 [r257208] Tilghman Lesher * Asterisk 1.6.1.19-rc1 Released 2010-04-05 15:15 +0000 [r256163] Leif Madsen * /, doc/tex/localchannel.tex: Merged revisions 256161 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r256161 | lmadsen | 2010-04-05 10:14:53 -0500 (Mon, 05 Apr 2010) | 1 line Fix for localchannel.tex to allow PDFs to be generated again. ........ 2010-04-02 23:47 +0000 [r256012-256017] Russell Bryant * channels/chan_local.c, /: Merged revisions 256015 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r256015 | russell | 2010-04-02 18:46:45 -0500 (Fri, 02 Apr 2010) | 16 lines Merged revisions 256014 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r256014 | russell | 2010-04-02 18:45:56 -0500 (Fri, 02 Apr 2010) | 9 lines Resolve a deadlock that occurs due to a pointless call to ast_bridged_channel() (closes issue #16840) Reported by: bzing2 Patches: patch.txt uploaded by bzing2 (license 902) issue_16840.rev1.diff uploaded by russell (license 2) Tested by: bzing2, russell ........ ................ * main/channel.c, /: Merged revisions 256010 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r256010 | russell | 2010-04-02 18:30:58 -0500 (Fri, 02 Apr 2010) | 9 lines Merged revisions 256009 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r256009 | russell | 2010-04-02 18:30:15 -0500 (Fri, 02 Apr 2010) | 2 lines Remove extremely verbose debug message. ........ ................ 2010-04-02 20:20 +0000 [r255954] Tilghman Lesher * main/asterisk.c, /: Merged revisions 255952 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r255952 | tilghman | 2010-04-02 15:19:01 -0500 (Fri, 02 Apr 2010) | 8 lines Pass the PID of the Asterisk process, not the PID of the canary. (closes issue #17065) Reported by: globalnetinc Patches: astcanary.patch uploaded by makoto (license 38) Tested by: frawd, globalnetinc ........ 2010-04-01 18:21 +0000 [r255675-255815] Tilghman Lesher * /, include/asterisk/lock.h: Merged revisions 255796 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r255796 | tilghman | 2010-04-01 13:16:37 -0500 (Thu, 01 Apr 2010) | 7 lines Fix DEBUG_THREADS build on Darwin. (closes issue #16828) Reported by: oej Patches: 20100331__issue16828.diff.txt uploaded by tilghman (license 14) ........ * apps/app_voicemail.c, /: Recorded merge of revisions 255592 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r255592 | tilghman | 2010-03-31 14:13:02 -0500 (Wed, 31 Mar 2010) | 22 lines Recorded merge of revisions 255591 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r255591 | tilghman | 2010-03-31 14:09:46 -0500 (Wed, 31 Mar 2010) | 15 lines Ensure line terminators in email are consistent. Fixes an issue with certain Mail Transport Agents, where attachments are not interpreted correctly. (closes issue #16557) Reported by: jcovert Patches: 20100308__issue16557__1.4.diff.txt uploaded by tilghman (license 14) 20100308__issue16557__1.6.0.diff.txt uploaded by tilghman (license 14) 20100308__issue16557__trunk.diff.txt uploaded by tilghman (license 14) Tested by: ebroad, zktech Reviewboard: https://reviewboard.asterisk.org/r/544/ ........ ................ 2010-03-31 17:51 +0000 [r255506] Leif Madsen * apps/app_dial.c, configs/sip.conf.sample: Merged revisions 255504 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r255504 | lmadsen | 2010-03-31 12:48:09 -0500 (Wed, 31 Mar 2010) | 5 lines Add documentation clarifying when 't' and 'T' can be used. (closes issue #17021) Reported by: kovzol Tested by: lmadsen, kovzol, davidw, ebroad ........ 2010-03-30 20:57 +0000 [r255325-255412] Russell Bryant * /, channels/chan_h323.c: Merged revisions 255410 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r255410 | russell | 2010-03-30 15:56:26 -0500 (Tue, 30 Mar 2010) | 9 lines Merged revisions 255409 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r255409 | russell | 2010-03-30 15:56:00 -0500 (Tue, 30 Mar 2010) | 2 lines Don't kill Asterisk if the H323 listener does not start. ........ ................ * /, pbx/pbx_dundi.c: Merged revisions 255323 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r255323 | russell | 2010-03-30 11:07:49 -0500 (Tue, 30 Mar 2010) | 9 lines Merged revisions 255322 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r255322 | russell | 2010-03-30 11:06:06 -0500 (Tue, 30 Mar 2010) | 2 lines Don't make Asterisk not start if pbx_dundi fails to initialize. ........ ................ 2010-03-26 19:29 +0000 [r255026-255068] Leif Madsen * configs/sip.conf.sample: Finish syncing documentation for tlsbindaddr. (issue #17054) * /, configs/sip.conf.sample: Merged revisions 255021 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r255021 | lmadsen | 2010-03-26 14:07:38 -0500 (Fri, 26 Mar 2010) | 8 lines Update confusing documentation for tlsbindaddr. Update some confusing documentation for the tlsbindaddr option in sip.conf.sample. Point at a link instead which has better documentation. (closes issue #17054) Reported by: klaus3000 ........ 2010-03-25 20:43 +0000 [r254804] Jason Parker * utils/Makefile, /: Merged revisions 254802 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r254802 | qwell | 2010-03-25 15:41:49 -0500 (Thu, 25 Mar 2010) | 9 lines Merged revisions 254800 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r254800 | qwell | 2010-03-25 15:41:15 -0500 (Thu, 25 Mar 2010) | 1 line Don't remove local copies of utils in uninstall. ........ ................ 2010-03-25 20:09 +0000 [r254720] Russell Bryant * channels/chan_usbradio.c, /: Merged revisions 254718 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r254718 | russell | 2010-03-25 15:08:40 -0500 (Thu, 25 Mar 2010) | 2 lines chan_usbradio depends on alsa. ........ 2010-03-25 20:03 +0000 [r254717] Jason Parker * main/astobj2.c, include/asterisk/astobj2.h: Fix DEBUG_THREADS issue with out-of-tree modules. Take 2, without ABI breakage this time. Review: https://reviewboard.asterisk.org/r/588/ 2010-03-25 17:46 +0000 [r254555] Mark Michelson * include/asterisk/acl.h, /: Merged revisions 254553 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r254553 | mmichelson | 2010-03-25 12:42:36 -0500 (Thu, 25 Mar 2010) | 11 lines Merged revisions 254552 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r254552 | mmichelson | 2010-03-25 12:33:35 -0500 (Thu, 25 Mar 2010) | 5 lines Add doxygen for acl.h Review: https://reviewboard.asterisk.org/r/528 ........ ................ 2010-03-25 17:21 +0000 [r254547] Sean Bright * channels/chan_sip.c: Initialize stream to avoid a compilation error. 2010-03-25 17:10 +0000 [r254541] Mark Michelson * channels/chan_sip.c: Fix potential crashes from trying to reference non-existent RTP streams. 2010-03-25 16:23 +0000 [r254486] Terry Wilson * /, main/file.c: Merged revisions 254453 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r254453 | twilson | 2010-03-25 11:03:51 -0500 (Thu, 25 Mar 2010) | 9 lines Merged revisions 254451 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r254451 | twilson | 2010-03-25 10:57:29 -0500 (Thu, 25 Mar 2010) | 2 lines Handle new SRCCHANGE control message here too ........ ................ 2010-03-25 16:15 +0000 [r254456] Mark Michelson * main/rtp.c, /: Recorded merge of revisions 254454 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r254454 | mmichelson | 2010-03-25 11:04:48 -0500 (Thu, 25 Mar 2010) | 50 lines Recorded merge of revisions 254452 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r254452 | mmichelson | 2010-03-25 10:59:56 -0500 (Thu, 25 Mar 2010) | 44 lines Several fixes regarding RFC2833 DTMF detection. Here is a copy and paste of the details from my request on reviewboard that dealt with these changes: Fix 1. The first change in place is to fix Mantis issue 15811, which deals with a situation where Asterisk will incorrectly interpret out of order RFC2833 frames as duplicate DTMF digits. For instance, we would receive a sequence like: seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1 seqno 4: DTMF 1 seqno 6: DTMF 1 (end) seqno 5: DTMF 1 seqno 7: DTMF 1 (end) seqno 8: DTMF 1 (end) Prior to this patch when we received the frame with seqno 5, we would interpret this as a new DTMF 1. With this patch, we will check the seqno of the incoming digit and not process the frame if the seqno is lower than the last recorded seqno. Note that we do not record the seqno of the dropped DTMF frame for future processing. While the above situation is what was designed to be fixed, the patch is written in such a way that the following would also be fixed too: seqno 9: DTMF 1 seqno 10: DTMF 1 (end) seqno 11: DTMF 1 (end) seqno 13: DTMF 2 seqno 12: DTMF 1 (end) seqno 14: DTMF 2 seqno 15: DTMF 2 (end) seqno 16: DTMF 2 (end) seqno 17: DTMF 2 (end) In this second situation, the beginning of the DTMF 2 arrives before the final end frame of the DTMF 1. With the patch, seqno 12 is no processed and thus we properly interpret the DTMF. Fix 2. The second change in place is to fix an issue like the following: seqno 1: DTMF 1 seqno 2: DTMF 1 seqno 3: DTMF 1 (end) *packet lost* seqno 4: DTMF 1 (end) *packet lost* seqno 5: DTMF 1 (end) *packet lost* seqno 6: DTMF 2 When we receive seqno 6, we had code in place that was supposed to properly end the previously unended DTMF 1. The problem was that the code was essentially a no-op. The code would set up an end frame for the DTMF 1 but would immediately overwrite the frame with the begin for DTMF 2. I changed process_dtmf_rfc2833() so that instead of returning a single frame, it is given as an output parameter a list of frames. Each frame that needs to be returned is appended to this list. Fix 3. The final change is a minor one where an AST_CONTROL_SRCCHANGE frame could get lost. If we process a cisco DTMF or an RFC 3389 frame and no frame was returned, then we would return &ast_null_frame. The problem is that earlier in the function, we may have generated an AST_CONTROL_SRCCHANGE frame and put it in the list of frames we wish to return. This frame would be lost in such a case. The patch fixes this problem ........ ................ 2010-03-25 15:22 +0000 [r254448] Leif Madsen * /, res/res_agi.c: Merged revisions 254446 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r254446 | lmadsen | 2010-03-25 10:21:26 -0500 (Thu, 25 Mar 2010) | 9 lines handle_speechset has 4 arguments. Update code to reflect that handle_speechset has 4 arguments. (closes issue #17093) Reported by: gpatri Patches: res_agi.patch uploaded by gpatri (license 1014) Tested by: pabelanger, mmichelson ........ 2010-03-24 17:18 +0000 [r254283] Jeff Peeler * res/res_monitor.c, /: Merged revisions 254277 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r254277 | jpeeler | 2010-03-24 12:15:05 -0500 (Wed, 24 Mar 2010) | 78 lines Merged revisions 254235 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r254235 | jpeeler | 2010-03-23 19:37:23 -0500 (Tue, 23 Mar 2010) | 72 lines Ensure that monitor recordings are written to the correct location (again) This is an extension to 248860. As such the dialplan test has been extended: ; non absolute path, not combined exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test) exten => 5040, n, dial(sip/5001) ; absolute path, not combined exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2) exten => 5041, n, dial(sip/5001) ; no path, not combined exten => 5042, 1, monitor(wav,monitor_test3) exten => 5042, n, dial(sip/5001) ; combined: changemonitor from non absolute to no path (leaves tmp/jeff) exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test4,m) exten => 5043, n, changemonitor(monitor_test5) exten => 5043, n, dial(sip/5001) ; combined: changemonitor from no path to non absolute path exten => 5044, 1, monitor(wav,monitor_test6,m) exten => 5044, n, changemonitor(tmp/jeff/monitor_test7) ; this wasn't possible before exten => 5044, n, dial(sip/5001) ; non absolute path, combined exten => 5045, 1, monitor(wav,tmp/jeff/monitor_test8,m) exten => 5045, n, dial(sip/5001) ; absolute path, combined exten => 5046, 1, monitor(wav,/tmp/jeff/monitor_test9,m) exten => 5046, n, dial(sip/5001) ; no path, combined exten => 5047, 1, monitor(wav,monitor_test10,m) exten => 5047, n, dial(sip/5001) ; combined: changemonitor from non absolute to absolute (leaves tmp/jeff) exten => 5048, 1, monitor(wav,tmp/jeff/monitor_test11,m) exten => 5048, n, changemonitor(/tmp/jeff/monitor_test12) exten => 5048, n, dial(sip/5001) ; combined: changemonitor from absolute to non absolute (leaves /tmp/jeff) exten => 5049, 1, monitor(wav,/tmp/jeff/monitor_test13,m) exten => 5049, n, changemonitor(tmp/jeff/monitor_test14) exten => 5049, n, dial(sip/5001) ; combined: changemonitor from no path to absolute exten => 5050, 1, monitor(wav,monitor_test15,m) exten => 5050, n, changemonitor(/tmp/jeff/monitor_test16) exten => 5050, n, dial(sip/5001) ; combined: changemonitor from absolute to no path (leaves /tmp/jeff) exten => 5051, 1, monitor(wav,/tmp/jeff/monitor_test17,m) exten => 5051, n, changemonitor(monitor_test18) exten => 5051, n, dial(sip/5001) ; not combined: changemonitor from non absolute to no path (leaves tmp/jeff) exten => 5052, 1, monitor(wav,tmp/jeff/monitor_test19) exten => 5052, n, changemonitor(monitor_test20) exten => 5052, n, dial(sip/5001) ; not combined: changemonitor from no path to non absolute exten => 5053, 1, monitor(wav,monitor_test21) exten => 5053, n, changemonitor(tmp/jeff/monitor_test22) exten => 5053, n, dial(sip/5001) ; not combined: changemonitor from non absolute to absolute (leaves tmp/jeff) exten => 5054, 1, monitor(wav,tmp/jeff/monitor_test23) exten => 5054, n, changemonitor(/tmp/jeff/monitor_test24) exten => 5054, n, dial(sip/5001) ; not combined: changemonitor from absolute to non absolute (leaves /tmp/jeff) exten => 5055, 1, monitor(wav,/tmp/jeff/monitor_test24) exten => 5055, n, changemonitor(tmp/jeff/monitor_test25) exten => 5055, n, dial(sip/5001) ; not combined: changemonitor from no path to absolute exten => 5056, 1, monitor(wav,monitor_test26) exten => 5056, n, changemonitor(/tmp/jeff/monitor_test27) exten => 5056, n, dial(sip/5001) ; not combined: changemonitor from absolute to no path (leaves /tmp/jeff) exten => 5057, 1, monitor(wav,/tmp/jeff/monitor_test28) exten => 5057, n, changemonitor(monitor_test29) exten => 5057, n, dial(sip/5001) ........ ................ 2010-03-23 21:37 +0000 [r254130] Tzafrir Cohen * tests/Makefile, /: Merged revisions 254001 via svnmerge from http://svn.digium.com/svn/asterisk/trunk ........ r254001 | tzafrir | 2010-03-23 21:19:52 +0200 (Tue, 23 Mar 2010) | 2 lines Change the name of the category 'TEST' to match the name of the subdir ........ 2010-03-23 21:20 +0000 [r254065] Jeff Peeler * main/channel.c, /: Merged revisions 254050 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r254050 | jpeeler | 2010-03-23 16:17:23 -0500 (Tue, 23 Mar 2010) | 14 lines Exit native bridging early for greater timing accuracy with warnings This changes native bridging to break one millisecond early so that the more accurate timeval calculations done in the generic bridge can be performed using the bridge config. Currently the time between exiting native bridging slightly late can sometimes cause a large enough discrepancy for warnings to be missed. For the record, 1.4 does not attempt to native bridge at all when warnings are enabled. (closes issue #15815) Reported by: adomjan Review: https://reviewboard.asterisk.org/r/577/ ........ 2010-03-22 19:56 +0000 [r253802] Matthew Nicholson * /, main/features.c: Merged revisions 253800 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r253800 | mnicholson | 2010-03-22 14:52:52 -0500 (Mon, 22 Mar 2010) | 11 lines Merged revisions 253799 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r253799 | mnicholson | 2010-03-22 14:50:00 -0500 (Mon, 22 Mar 2010) | 4 lines Unconditionally copy the caller's account code to the called party. (related to issue #16331) ........ ................ 2010-03-22 19:06 +0000 [r253713-253759] Tilghman Lesher * /, contrib/scripts/dbsep.cgi: Merged revisions 253758 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r253758 | tilghman | 2010-03-22 14:05:27 -0500 (Mon, 22 Mar 2010) | 2 lines Update query should be an UPDATE, not a SELECT. ........ * /, contrib/scripts/dbsep.cgi: Merged revisions 253755 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r253755 | tilghman | 2010-03-22 13:58:48 -0500 (Mon, 22 Mar 2010) | 4 lines Return the list for later manipulation. This fixes an issue with the update procedure. Debugging with mmichelson. ........ * configs/dbsep.conf.sample, /, contrib/scripts/dbsep.cgi: Merged revisions 253712 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r253712 | tilghman | 2010-03-22 11:59:35 -0500 (Mon, 22 Mar 2010) | 2 lines Accomodate equal signs in DSNs and add documentation, based upon mmichelson's feedback. ........ 2010-03-20 17:55 +0000 [r253621-253624] Russell Bryant * main/tcptls.c, /, main/features.c, cdr/cdr_pgsql.c, main/stdtime/localtime.c: Merged revisions 253540 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r253540 | russell | 2010-03-20 07:03:07 -0500 (Sat, 20 Mar 2010) | 2 lines Resolve more compiler warnings on FreeBSD. ........ * apps/app_followme.c, apps/app_dial.c, /: Merged revisions 253538 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r253538 | russell | 2010-03-20 06:43:08 -0500 (Sat, 20 Mar 2010) | 2 lines Resolve compiler warnings on FreeBSD. ........ * /, pbx/pbx_dundi.c: Merged revisions 253537 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r253537 | russell | 2010-03-20 06:39:39 -0500 (Sat, 20 Mar 2010) | 2 lines Resolve a compiler warning on FreeBSD. ........ * channels/chan_dahdi.c, /: Merged revisions 253536 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r253536 | russell | 2010-03-20 06:33:30 -0500 (Sat, 20 Mar 2010) | 4 lines Use SHRT_MAX instead of MAXSHORT. These changes fix build issues I had with this module on FreeBSD. ........ 2010-03-19 07:59 +0000 [r253491] Alec L Davis * main/astobj2.c, /: Merged revisions 253490 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r253490 | alecdavis | 2010-03-19 20:37:00 +1300 (Fri, 19 Mar 2010) | 19 lines prevent segfault if bad magic number is encountered. internal_ao2_ref uses INTERNAL_OBJ which mzy report 'bad magic number', but internal_ao2_ref continues on, causing segfault. Although AO2_MAGIC number is checked by INTERNAL_OBJ before internal_ao2_ref is called, A02_MAGIC is being destroyed (or a wrong pointer) by the time internal_ao2_ref uses INTERNAL_OBJ. internal_ao2_ref now returns -1 if INTERNAL_OBJ encouters a bad magic number. (issue #17037) Reported by: alecdavis Patches: bug17037.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis ........ 2010-03-18 17:56 +0000 [r253258-253347] Leif Madsen * apps/app_userevent.c: Slightly different fix for UserEvent docs update. (issue #16961) * doc/tex/localchannel.tex: Merged revisions 253256 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r253256 | lmadsen | 2010-03-18 10:46:52 -0500 (Thu, 18 Mar 2010) | 9 lines Update to new Local channel documentation. Add same changes as commit to 1.4, but convert to TeX. (issue #16963) Reported by: kobaz Patches: localchannel-2.txt uploaded by kobaz (license 834) ........ 2010-03-17 16:25 +0000 [r253158] Terry Wilson * main/rtp.c, channels/chan_skinny.c, channels/chan_h323.c, channels/chan_mgcp.c, channels/chan_sip.c, include/asterisk/rtp.h: Revert API change in release branches This re-renames ast_rtp_update_source to ast_rtp_new_source 2010-03-17 00:30 +0000 [r253030] Leif Madsen * configs/say.conf.sample: Merged revisions 253028 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r253028 | lmadsen | 2010-03-16 19:29:06 -0500 (Tue, 16 Mar 2010) | 13 lines Merged revisions 253018 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r253018 | lmadsen | 2010-03-16 19:26:19 -0500 (Tue, 16 Mar 2010) | 6 lines Add french snipset to say.conf. Add the french snipset to say.conf. (Closes issue #15799) ........ ................ 2010-03-16 23:53 +0000 [r252977] Tilghman Lesher * apps/app_stack.c, /: Merged revisions 252976 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r252976 | tilghman | 2010-03-16 18:49:35 -0500 (Tue, 16 Mar 2010) | 8 lines Mask out previous arguments on each nested invocation of Gosub. (closes issue #16758) Reported by: wdoekes Patches: 20100316__issue16758.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/561/ ........ 2010-03-16 19:01 +0000 [r252769] Russell Bryant * utils/Makefile, /: Merged revisions 252767 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r252767 | russell | 2010-03-16 14:01:04 -0500 (Tue, 16 Mar 2010) | 13 lines Merged revisions 252766 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r252766 | russell | 2010-03-16 14:00:43 -0500 (Tue, 16 Mar 2010) | 6 lines Don't treat warnings as errors for muted. muted supports OS X, but uses functions marked as deprecated in 10.6. However, the functions are still supported, so just ignore the warnings for now and allow the build to proceed. ........ ................ 2010-03-16 18:49 +0000 [r252764] Leif Madsen * /, configs/extensions.ael.sample: Merged revisions 252762 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r252762 | lmadsen | 2010-03-16 13:48:22 -0500 (Tue, 16 Mar 2010) | 15 lines Merged revisions 252761 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r252761 | lmadsen | 2010-03-16 13:46:20 -0500 (Tue, 16 Mar 2010) | 7 lines Additional extensions.ael global variable fixes. Fixing up a couple more overlapping global variable namespaces shared with extensions.conf.sample. Also noticed a few of the lines that were commented out didn't have the closing semi-colon so I added that as well. (issue #17035) ........ ................ 2010-03-15 21:59 +0000 [r252625] Sean Bright * /, apps/app_meetme.c: Merged revisions 252623 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r252623 | seanbright | 2010-03-15 17:55:44 -0400 (Mon, 15 Mar 2010) | 4 lines Resolve a crash in SLATrunk when the specified trunk doesn't exist. Reported by philipp64 in #asterisk-dev. ........ 2010-03-15 21:54 +0000 [r252621] Tilghman Lesher * contrib/init.d/org.asterisk.asterisk.plist, /: Merged revisions 252619 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r252619 | tilghman | 2010-03-15 16:51:55 -0500 (Mon, 15 Mar 2010) | 9 lines Merged revisions 252617 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r252617 | tilghman | 2010-03-15 16:43:14 -0500 (Mon, 15 Mar 2010) | 2 lines Uh, yeah. Umask. I'm stupid. ........ ................ 2010-03-15 20:53 +0000 [r252536] Leif Madsen * configs/extensions.ael.sample: Merged revisions 252534 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r252534 | lmadsen | 2010-03-15 15:52:32 -0500 (Mon, 15 Mar 2010) | 15 lines Merged revisions 252533 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r252533 | lmadsen | 2010-03-15 15:48:56 -0500 (Mon, 15 Mar 2010) | 7 lines Update extensions.ael file to not overlap extensions.conf. Updated the extensions.ael file so the global variables don't overlap those that we have in extensions.conf (sample files). This way unexpected things won't happed hopefully if both pbx_ael and res_config are loaded. (closes issue #17035) Reported by: pprindeville ........ ................ 2010-03-15 05:03 +0000 [r252364-252443] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 252442 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r252442 | tilghman | 2010-03-14 23:25:35 -0500 (Sun, 14 Mar 2010) | 7 lines THIS IS NOT PYTHON. Indentation doesn't matter, only braces do. (closes issue #17025) Reported by: smurfix Patches: sip.patch uploaded by smurfix (license 547) ........ * main/asterisk.c, Makefile, contrib/init.d/org.asterisk.asterisk.plist (added), /: Merged revisions 252362 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r252362 | tilghman | 2010-03-14 20:37:04 -0500 (Sun, 14 Mar 2010) | 11 lines Merged revisions 252361 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r252361 | tilghman | 2010-03-14 20:33:50 -0500 (Sun, 14 Mar 2010) | 4 lines Launch Asterisk on Mac OS X with launchd. Reviewboard: https://reviewboard.asterisk.org/r/551/ ........ ................ 2010-03-14 17:47 +0000 [r252316] Sean Bright * cdr/cdr_sqlite3_custom.c, /: Merged revisions 252314 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r252314 | seanbright | 2010-03-14 13:43:46 -0400 (Sun, 14 Mar 2010) | 8 lines Fix building CDR and CEL SQLite3 modules. They added a sqlite3_log() function which was conflicting with our function names. (closes issue #17017) Reported by: alephlg ........ 2010-03-13 00:31 +0000 [r252135-252177] Terry Wilson * main/rtp.c: Remove unusued field * include/asterisk/frame.h, main/rtp.c, channels/chan_mgcp.c, main/channel.c, /, channels/chan_sip.c, channels/chan_skinny.c, include/asterisk/rtp.h, channels/chan_h323.c, configs/sip.conf.sample: Merged revisions 252089 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r252089 | twilson | 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) | 20 lines Only change the RTP ssrc when we see that it has changed This change basically reverts the change reviewed in https://reviewboard.asterisk.org/r/374/ and instead limits the updating of the RTP synchronization source to only those times when we detect that the other side of the conversation has changed the ssrc. The problem is that SRCUPDATE control frames are sent many times where we don't want a new ssrc, including whenever Asterisk has to send DTMF in a normal bridge. This is also not the first time that this mistake has been made. The initial implementation of the ast_rtp_new_source function also changed the ssrc--and then it was removed because of this same issue. Then, we put it back in again to fix a different issue. This patch attempts to only change the ssrc when we see that the other side of the conversation has changed the ssrc. It also renames some functions to make their purpose more clear. Review: https://reviewboard.asterisk.org/r/540/ ........ 2010-03-12 19:54 +0000 [r251996] Richard Mudgett * channels/chan_dahdi.c: Forward declaring dahdi_pri was already done. 2010-03-12 19:50 +0000 [r251993] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 251989 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r251989 | tilghman | 2010-03-12 13:43:23 -0600 (Fri, 12 Mar 2010) | 8 lines Don't override a user option with the global option. (closes issue #16849) Reported by: ip-rob Patches: 20100311__issue16849.diff.txt uploaded by tilghman (license 14) Tested by: ip-rob ........ 2010-03-12 19:44 +0000 [r251990] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 251987 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r251987 | rmudgett | 2010-03-12 13:40:16 -0600 (Fri, 12 Mar 2010) | 9 lines Merged revisions 251986 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r251986 | rmudgett | 2010-03-12 13:33:22 -0600 (Fri, 12 Mar 2010) | 1 line Make chan_dahdi wakeup_sub() prototype not conditional. ........ ................ 2010-03-11 21:08 +0000 [r251875-251886] Tilghman Lesher * apps/app_exec.c, /: Merged revisions 251884 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r251884 | tilghman | 2010-03-11 15:07:07 -0600 (Thu, 11 Mar 2010) | 8 lines Because ExecIf needs to reprocess arguments, it's best if we don't remove quotes during parsing. (closes issue #16905) Reported by: ip-rob Patches: 20100303__issue16905.diff.txt uploaded by tilghman (license 14) Tested by: ip-rob ........ * apps/app_system.c, /: Merged revisions 251877 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r251877 | tilghman | 2010-03-11 14:25:02 -0600 (Thu, 11 Mar 2010) | 8 lines If the argument to the system application is quoted, ensure we remove the quotes before trying to execute. (closes issue #16842) Reported by: ip-rob Patches: 20100310__issue16842.diff.txt uploaded by tilghman (license 14) Tested by: ip-rob ........ * funcs/func_odbc.c: Verify whether the created buffer was actually large enough to hold the expanded value. For certain types of queries, where the size of the substituted query was much larger than the template, it was possible for the substitution buffer to be too small. This is only an issue in 1.6.1, as previously we used a static buffer anyway, and we have a substitution routine in 1.6.2 forward that automatically sizes itself appropriately to handle larger expansions. (closes issue #17006) Reported by: viniciusfontes Patches: 20100311__issue17006.diff.txt uploaded by tilghman (license 14) Tested by: tilghman 2010-03-11 Leif Madsen * Asterisk 1.6.1.18 released 2010-03-05 Leif Madsen * Asterisk 1.6.1.18-rc2 released 2010-03-05 Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 250913 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r250913 | tilghman | 2010-03-04 22:37:36 -0600 (Thu, 04 Mar 2010) | 7 lines Missing quote in ODBC query. (closes issue #16953) Reported by: elguero Patches: app_voicemail-odbc-syntax-fix.diff uploaded by elguero (license 37) ........ 2010-03-04 Leif Madsen * Asterisk 1.6.1.18-rc1 released 2010-03-03 21:25 +0000 [r250611] Leif Madsen * /, doc/tex/localchannel.tex: Merged revisions 250609 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r250609 | lmadsen | 2010-03-03 16:22:55 -0500 (Wed, 03 Mar 2010) | 11 lines Update existing Local channel documentation. A complete re-write of the Local channel documentation has been performed, with the existing information from localchannel.txt and localchannel.tex merged in. (closes issue #16637) Reported by: kobaz Patches: localchannel.tex uploaded by lmadsen (license 10) localchannel.txt uploaded by lmadsen (license 10) Tested by: lmadsen, jsmith, mmichelson ........ 2010-03-03 19:09 +0000 [r250483] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 250481 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r250481 | jpeeler | 2010-03-03 13:06:06 -0600 (Wed, 03 Mar 2010) | 22 lines Merged revisions 250480 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r250480 | jpeeler | 2010-03-03 13:04:11 -0600 (Wed, 03 Mar 2010) | 15 lines Make sure to clear red alarm after polarity reversal. From the issue: The automatic overnight line tests (or manual ones) used on UK (BT) lines causes a red alarm on a dahdi / TDM400P connected channel. This is because the line uses voltage tests (battery loss) and polarity reversal. The polarity reversal causes chan_dahdi to initiate v23 CallerID processing but during this the event DAHDI_EVENT_NOALARM is ignored so that the alarm is never cleared. (closes issue #14163) Reported by: jedi98 Patches: chan_dahdi-1.4-inalarm.diff uploaded by jedi98 (license 653) Tested by: mattbrown, Chainsaw, mikeeccleston ........ ................ 2010-03-03 18:05 +0000 [r250260-250397] David Vossel * channels/chan_iax2.c, /: Merged revisions 250395 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r250395 | dvossel | 2010-03-03 12:03:19 -0600 (Wed, 03 Mar 2010) | 22 lines Merged revisions 250394 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r250394 | dvossel | 2010-03-03 12:02:27 -0600 (Wed, 03 Mar 2010) | 16 lines fixes problem with duplicate TXREQ packets When Asterisk receives an IAX2 TXREQ packet, try_transfer() will call store_by_transfercallno() to link the chan_iax2_pvt struct into iax_transfercallno_pvts. If a duplicate TXREQ packet is received for the same call, the pvt struct will be linked into iax_transfercallno_pvts multiple times. This patch fixes this. Thanks rain for debugging this and providing a patch! (closes issue #16904) Reported by: rain Patches: iax2-double-txreq-fix.diff uploaded by rain (license 327) Tested by: rain, dvossel ........ ................ * /, channels/chan_sip.c: Merged revisions 250246 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r250246 | dvossel | 2010-03-02 18:18:28 -0600 (Tue, 02 Mar 2010) | 2 lines fixes signed to unsigned int comparision issue for FaxMaxDatagram value. ........ 2010-03-02 21:11 +0000 [r250039-250053] Leif Madsen * doc/tex/imapstorage.tex, /: Merged revisions 250051 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r250051 | lmadsen | 2010-03-02 16:09:27 -0500 (Tue, 02 Mar 2010) | 8 lines Update IMAP documentation. Update the IMAP documentation to make it clear that storing voicemails in the same folder as a large number of emails could potentially cause significant slow downs when writing or retrieving voicemails. (issue #16704) Reported by: TimeHider Tested by: lmadsen, TimeHider ........ * configs/cdr.conf.sample: Merged revisions 250045 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r250045 | lmadsen | 2010-03-02 15:52:19 -0500 (Tue, 02 Mar 2010) | 15 lines Merged revisions 250043 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r250043 | lmadsen | 2010-03-02 15:51:35 -0500 (Tue, 02 Mar 2010) | 7 lines Update documentation to clarify purpose of unanswered option. (closes issue #16267) Reported by: elsto Patches: cdr.conf.sample.patch.txt uploaded by lmadsen (license 10) Tested by: davidw, elsto ........ ................ * doc/tex/configuration.tex, /: Merged revisions 250037 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r250037 | lmadsen | 2010-03-02 15:36:10 -0500 (Tue, 02 Mar 2010) | 4 lines Update documentation to not imply we support overriding options. (closes issue #16855) Reported by: davidw ........ 2010-03-02 19:47 +0000 [r249949] Alec L Davis * main/editline/makelist.in, apps/app_echo.c, UPGRADE.txt: revert ability to exit echo app caused a regression, as only supported VOICE, not VIDEO etc. (issue #16880) 2010-03-02 19:15 +0000 [r249896] David Vossel * channels/chan_oss.c, channels/misdn_config.c, include/asterisk/abstract_jb.h, configs/alsa.conf.sample, channels/chan_jingle.c, channels/chan_usbradio.c, channels/chan_dahdi.c, channels/chan_skinny.c, configs/mgcp.conf.sample, main/abstract_jb.c, channels/chan_h323.c, channels/chan_alsa.c, configs/sip.conf.sample, channels/chan_mgcp.c, channels/chan_unistim.c, configs/console.conf.sample, configs/chan_dahdi.conf.sample, channels/chan_local.c, configs/oss.conf.sample, channels/chan_sip.c, /, configs/usbradio.conf.sample, configs/misdn.conf.sample, channels/chan_gtalk.c, channels/chan_console.c: Merged revisions 249893 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r249893 | dvossel | 2010-03-02 13:08:38 -0600 (Tue, 02 Mar 2010) | 11 lines fixes adaptive jitterbuffer configuration When configuring the adaptive jitterbuffer, the target_extra value not only could not be set from the configuration, but was not even being set to its proper default. This value is required in order for the adaptive jitterbuffer to work correctly. To resolve this a config option has been added to expose this value to the conf files, and a default value is provided when no config specific value is present. ........ 2010-03-02 09:05 +0000 [r249843] Alec L Davis * apps/app_echo.c: fixes ability to exit echo app when called from a ISDN channel, null frames prevent '#' exit. Now only echo back VOICE and DTMF frames (issue #16880) Reported by: alecdavis Patches: echo_exit_1-6-1.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis 2010-03-01 19:39 +0000 [r249674] Sean Bright * apps/app_voicemail.c, /: Merged revisions 249672 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r249672 | seanbright | 2010-03-01 14:36:30 -0500 (Mon, 01 Mar 2010) | 18 lines Merged revisions 249671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r249671 | seanbright | 2010-03-01 14:35:01 -0500 (Mon, 01 Mar 2010) | 11 lines Fix crash in app_voicemail related to message counting. We were passing a 'struct inprocess **' and treating it like a 'struct inprocess *' causing a segfault. (closes issue #16921) Reported by: whardier Patches: 20100301_issue16921.patch uploaded by seanbright (license 71) Tested by: whardier ........ ................ 2010-03-01 18:39 +0000 [r249624] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 249623 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r249623 | tilghman | 2010-03-01 12:36:06 -0600 (Mon, 01 Mar 2010) | 2 lines Constify a bit of app_voicemail, to make ODBC and IMAP compile once again. ........ 2010-03-01 17:19 +0000 [r249548] Jeff Peeler * channels/chan_local.c, /: Merged revisions 249538 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r249538 | jpeeler | 2010-03-01 11:11:31 -0600 (Mon, 01 Mar 2010) | 18 lines Merged revisions 249536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r249536 | jpeeler | 2010-03-01 11:02:03 -0600 (Mon, 01 Mar 2010) | 11 lines Modify queued frames from local channels to not set the other side to up In this case, attended transfers were broken due to ast_feature_request_and_dial detecting the channel being set to up before the answer frame could be read and therefore failing to mark the channel as ready. This fix is a regression fix for 244785, which should continue to work properly as well. (closes issue #16816) Reported by: jamhed Tested by: jamhed, corruptor ........ ................ 2010-02-28 20:51 +0000 [r249406-249492] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 249491 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r249491 | tilghman | 2010-02-28 14:50:01 -0600 (Sun, 28 Feb 2010) | 5 lines Fix unit test that Alec Davis broke. (closes issue #16927) Reported by: alecdavis ........ * apps/app_voicemail.c, include/asterisk/app.h, /: Merged revisions 249405 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r249405 | tilghman | 2010-02-28 01:10:22 -0600 (Sun, 28 Feb 2010) | 2 lines Properly document voicemail API documents. Also fix a crash reported via the -dev list. ........ 2010-02-27 23:37 +0000 [r249363] Alec L Davis * channels/chan_dahdi.c: overlap receiving: automatically send CALL PROCEEDING when dialplan starts Following Q.931 5.2.4 When the user has determined that sufficient call information has been received the user shall stop T302 and send CALL PROCEEDING to the network. Previously timeouts were possible if the dialplan took a long time to issue any response back to the network. Verified that our local TELCO also does the same. (issue #16789) Reported by: alecdavis Patches: overlap_receiving_trunk.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis 2010-02-27 14:09 +0000 [r249237] Kevin P. Fleming * channels/chan_iax2.c, /: Merged revisions 249235 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r249235 | kpfleming | 2010-02-27 09:08:35 -0500 (Sat, 27 Feb 2010) | 9 lines Merged revisions 249234 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r249234 | kpfleming | 2010-02-27 09:07:59 -0500 (Sat, 27 Feb 2010) | 1 line add a reference to the now-published IAX2 RFC ........ ................ 2010-02-26 18:48 +0000 [r249189] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 249187 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r249187 | tilghman | 2010-02-26 12:41:57 -0600 (Fri, 26 Feb 2010) | 18 lines Cleanups to fix bugs in the VM count API functions. - Urgent voicemails were not attached, because the attachment code looked in the wrong folder. - Urgent voicemails were sometimes counted twice when displaying the count of new messages. - Backends were inconsistent as to which voicemails each API counted. (closes issue #15654) Reported by: tomo1657 Patches: 20100225__issue15654.diff.txt uploaded by tilghman (license 14) Tested by: tilghman (closes issue #16448) Reported by: hevad Review: https://reviewboard.asterisk.org/r/525/ ........ 2010-02-26 17:05 +0000 [r249103] Mark Michelson * /, channels/chan_sip.c: Merged revisions 249101 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r249101 | mmichelson | 2010-02-26 11:04:58 -0600 (Fri, 26 Feb 2010) | 14 lines Merged revisions 249100 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r249100 | mmichelson | 2010-02-26 11:04:29 -0600 (Fri, 26 Feb 2010) | 8 lines For T.38 reINVITEs treat a 606 the same as a 488. (closes issue #16792) Reported by: vrban Patches: t38_606.patch uploaded by vrban (license 756) ........ ................ 2010-02-25 23:11 +0000 [r248954] Jeff Peeler * res/res_monitor.c, /: Merged revisions 248952 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r248952 | jpeeler | 2010-02-25 17:09:54 -0600 (Thu, 25 Feb 2010) | 24 lines Merged revisions 248860 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r248860 | jpeeler | 2010-02-25 15:22:06 -0600 (Thu, 25 Feb 2010) | 18 lines Ensure that monitor recordings are written to the correct location (again) This is an extension to 248757. As such the dialplan test has been extended: exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n, dial(sip/5001) exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n, dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b) exten => 5042, n, dial(sip/5001) exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test3,m) exten => 5043, n, changemonitor(monitor_test4) exten => 5043, n, dial(sip/5001) exten => 5044, 1, monitor(wav,monitor_test4,m) exten => 5044, n, changemonitor(tmp/jeff/monitor_test5) ; this looks to fail by design and emits a warning exten => 5044, n, dial(sip/5001) ........ ................ 2010-02-25 22:42 +0000 [r248948] Mark Michelson * /, main/acl.c: Merged revisions 248946 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r248946 | mmichelson | 2010-02-25 16:41:48 -0600 (Thu, 25 Feb 2010) | 5 lines Fix incorrect ACL behavior when CIDR notation of "/0" is used. AST-2010-003 ........ 2010-02-25 21:24 +0000 [r248863] Tilghman Lesher * main/asterisk.c, /: Merged revisions 248861 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r248861 | tilghman | 2010-02-25 15:22:39 -0600 (Thu, 25 Feb 2010) | 22 lines Merged revisions 248859 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r248859 | tilghman | 2010-02-25 15:21:05 -0600 (Thu, 25 Feb 2010) | 15 lines Some platforms clear /var/run at boot, which makes connecting a remote console... difficult. Previously, we only created the default /var/run/asterisk directory at install time. While we could create it in the init script, that would not work for those who start asterisk manually from the command line. So the safest thing to do is to create it as part of the Asterisk boot process. This also changes the ownership of the directory, because the pid and ctl files are created after we setuid/setgid. (closes issue #16802) Reported by: Brian Patches: 20100224__issue16802.diff.txt uploaded by tilghman (license 14) Tested by: tzafrir ........ ................ 2010-02-25 18:52 +0000 [r248796] Jeff Peeler * res/res_monitor.c, /: Merged revisions 248793 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r248793 | jpeeler | 2010-02-25 12:37:56 -0600 (Thu, 25 Feb 2010) | 22 lines Merged revisions 248757 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r248757 | jpeeler | 2010-02-25 12:06:54 -0600 (Thu, 25 Feb 2010) | 15 lines Ensure that monitor recordings are written to the correct location. Recordings should be placed in the monitor directory when a non-absolute path is used. Exact dialplan used for testing: exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test,b) exten => 5040, n, dial(sip/5001) exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2,b) exten => 5041, n, dial(sip/5001) exten => 5042, 1, monitor(wav,monitor_test3,b) exten => 5042, n, dial(sip/5001) ABE-2101 ........ ................ 2010-02-24 21:29 +0000 [r248641] Tilghman Lesher * /, main/logger.c: Merged revisions 248584 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r248584 | tilghman | 2010-02-24 15:17:26 -0600 (Wed, 24 Feb 2010) | 14 lines Merged revisions 248582 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r248582 | tilghman | 2010-02-24 15:02:18 -0600 (Wed, 24 Feb 2010) | 7 lines Remove color code sequences from verbose messages that go to logfiles. (closes issue #16786) Reported by: dodo Patches: logger2.patch uploaded by dodo (license 989) Tested by: tilghman ........ ................ 2010-02-23 16:48 +0000 [r248399] David Vossel * /, channels/chan_sip.c: Merged revisions 248397 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r248397 | dvossel | 2010-02-23 10:34:39 -0600 (Tue, 23 Feb 2010) | 15 lines Merged revisions 248396 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r248396 | dvossel | 2010-02-23 10:26:05 -0600 (Tue, 23 Feb 2010) | 9 lines fixes invite with replaces deadlock (closes issue #16862) Reported by: pwalker Patches: replaces_deadlock_1.4 uploaded by dvossel (license 671) Tested by: pwalker, dvossel ........ ................ 2010-02-19 19:05 +0000 [r248009] Tilghman Lesher * main/loader.c, /, channels/chan_console.c: Merged revisions 228798 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk (closes issue #16470) Reported by: kjotte ........ r228798 | tilghman | 2009-11-09 01:37:52 -0600 (Mon, 09 Nov 2009) | 14 lines Fix various problems detected with Valgrind. * chan_console accessed pvts after deallocation. * The module loader did not check usecount on shutdown, which led to chan_iax2 reading a timer that was already unloaded. (closes issue #16062) Reported by: alexanderheinz Patches: 20091109__issue16062.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ 2010-02-19 18:22 +0000 [r247946] Richard Mudgett * channels/chan_misdn.c, /: Merged revisions 247914 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r247914 | rmudgett | 2010-02-19 11:33:33 -0600 (Fri, 19 Feb 2010) | 62 lines Merged revisions 247910 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r247910 | rmudgett | 2010-02-19 11:18:49 -0600 (Fri, 19 Feb 2010) | 55 lines Merged revision 247904 from https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... .......... r247904 | rmudgett | 2010-02-19 10:49:44 -0600 (Fri, 19 Feb 2010) | 49 lines Make chan_misdn DTMF processing consistent with other channel technologies. The processing of DTMF tones on the receiving side of an ISDN channel is inconsistent with the way it is handled in other channels, especially DAHDI analog. This causes DTMF tones sent from an ISDN phone to be doubled at the connected party. We are using the following 2 options of misdn.conf 1) astdtmf=yes 2) senddtmf=yes Option one is necessary because the asterisk DSP DTMF detection is better than mISDN's internal DSP. Not as many false positives. Option two is necessary to transmit DTMF tones end to end when mISDN channels are connected to SIP channels with out of band DTMF for example. The symptom is that DTMF tones sent by an ISDN phone are doubled on the way through asterisk when two mISDN channels are connected with a Local channel in between or if it is bridged to an analog channel. The doubling of DTMF tones is because DTMF is passed inband to asterisk by the mISDN channel and passed out of band once again after the release of the DTMF tone. Passing it inband is wrong. Neither an analog channel nor SIP channel passes DTMF inband if configured to inband DTMF. Analog and SIP channels filter out the DTMF tones because they use the voice frames returned by ast_dsp_process. But chan_misdn passes the unfiltered input voice frames instead. To overcome one aspect of the problem, the doubling of DTMF tones when two mISDN channels are directly bridged, someone made an 'optimization', where in that case the DTMF tone passed out-of-band to the peer channel is not translated to an inband tone at the transmit side. This optimization is bad because it does not work in general. For example, analog channels or mISDN channels when bridged through an intermediary local channel will generate DTMF tones from out-of-band information. Also, of course, it must not be done when there is no inband DTMF available. This patch fixes the issue. Now chan_misdn will filter the received inband DTMF signal the same as other channel types. Another change included: No need to build an extra translation path because ast_process_dsp does it if required. Patches: misdn-dtmf.patch JIRA ABE-2080 ................ ................ 2010-02-18 23:15 +0000 [r247790-247843] Tilghman Lesher * res/res_speech.c, /: Merged revisions 247841 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r247841 | tilghman | 2010-02-18 17:13:46 -0600 (Thu, 18 Feb 2010) | 7 lines Revert an errant part of a previous cleanup, to fix a memory corruption issue. (closes issue #16368) Reported by: thirionjwf Patches: res_speech.c.patch uploaded by thirionjwf (license 955) ........ * /, channels/chan_sip.c: Merged revisions 247787 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r247787 | tilghman | 2010-02-18 15:42:53 -0600 (Thu, 18 Feb 2010) | 17 lines If the peer record is from realtime, it could be set to 0, due to MySQL not representing NULL well in integer columns. NULL means the value is not specified for the column, which normally means the driver uses whatever is the default value. However, on MySQL, placing a NULL in either a float or integer column results in a retrieval of the 0 value. Hence, users get an errant error on load. This patch suppresses that error and makes the value as if it was not there. Note that this cannot be done in the realtime driver, because the lack of difference between NULL and 0 can only be intepreted correctly by the driver itself. If we did it in the realtime driver, then it would be effectively impossible to set any realtime field to 0, because it would act as if the field were unspecified and possibly take on a different value. (closes issue #16683) Reported by: wdoekes ........ 2010-02-18 19:45 +0000 [r247654] Matthew Nicholson * /, main/features.c: Merged revisions 247652 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r247652 | mnicholson | 2010-02-18 13:39:37 -0600 (Thu, 18 Feb 2010) | 13 lines Merged revisions 247651 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247651 | mnicholson | 2010-02-18 13:38:09 -0600 (Thu, 18 Feb 2010) | 6 lines Copy the calling party's account code to the called party if they don't already have one. (closes issue #16331) Reported by: bluefox Tested by: mnicholson ........ ................ 2010-02-18 16:57 +0000 [r247505-247511] Leif Madsen * README-SERIOUSLY.bestpractices.txt: Merged revisions 247509 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r247509 | lmadsen | 2010-02-18 11:54:43 -0500 (Thu, 18 Feb 2010) | 9 lines Merged revisions 247508 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247508 | lmadsen | 2010-02-18 11:53:44 -0500 (Thu, 18 Feb 2010) | 1 line Add additional link to best practices document per jsmith. ........ ................ * README-SERIOUSLY.bestpractices.txt (added): Merged revisions 247503 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r247503 | lmadsen | 2010-02-18 11:41:04 -0500 (Thu, 18 Feb 2010) | 18 lines Merged revisions 247502 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247502 | lmadsen | 2010-02-18 11:38:17 -0500 (Thu, 18 Feb 2010) | 10 lines Add best practices documentation. (issue #16808) Reported by: lmadsen (issue #16810) Reported by: Nick_Lewis Tested by: lmadsen Review: https://reviewboard.asterisk.org/r/507/ ........ ................ 2010-02-18 04:21 +0000 [r247425] Russell Bryant * sounds/Makefile, Makefile, /: Merged revisions 247423 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r247423 | russell | 2010-02-17 22:20:11 -0600 (Wed, 17 Feb 2010) | 17 lines Merged revisions 247422 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247422 | russell | 2010-02-17 22:19:01 -0600 (Wed, 17 Feb 2010) | 10 lines Tweak argument handling for wget in the sounds Makefile. 1) Fix the check to see if we are using wget to not be full of fail. The configure script populates this variable with the absolute path to wget if it is found, so it didn't work. 2) Allow some extra arguments to be passed in for wget. This is just a simple change to allow our Bamboo build script to tell wget to be quiet and not fill up our logs with download status output. ........ ................ 2010-02-17 21:28 +0000 [r246987-247336] Mark Michelson * /, main/utils.c, include/asterisk/strings.h: Merged revisions 247335 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r247335 | mmichelson | 2010-02-17 15:22:40 -0600 (Wed, 17 Feb 2010) | 20 lines Fix two problems in ast_str functions found while writing a unit test. 1. The documentation for ast_str_set and ast_str_append state that the max_len parameter may be -1 in order to limit the size of the ast_str to its current allocated size. The problem was that the max_len parameter in all cases was a size_t, which is unsigned. Thus a -1 was interpreted as UINT_MAX instead of -1. Changing the max_len parameter to be ssize_t fixed this issue. 2. Once issue 1 was fixed, there was an off-by-one error in the case where we attempted to write a string larger than the current allotted size to a string when -1 was passed as the max_len parameter. When trying to write more than the allotted size, the ast_str's __AST_STR_USED was set to 1 higher than it should have been. Thanks to Tilghman for quickly spotting the offending line of code. Oh, and the unit test that I referenced in the top line of this commit will be added to reviewboard shortly. Sit tight... ........ * apps/app_queue.c, /: Merged revisions 247169 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r247169 | mmichelson | 2010-02-17 10:24:54 -0600 (Wed, 17 Feb 2010) | 9 lines Merged revisions 247168 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247168 | mmichelson | 2010-02-17 10:24:17 -0600 (Wed, 17 Feb 2010) | 3 lines Make sure that when autofill is disabled that callers not in the front of the queue cannot place calls. ........ ................ * /, main/utils.c: Merged revisions 247076 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r247076 | mmichelson | 2010-02-16 17:44:33 -0600 (Tue, 16 Feb 2010) | 12 lines Add va_end calls to __ast_str_helper. According to the man page for stdarg(3), "Each invocation of va_copy() must be matched by a corresponding invocation of va_end() in the same function." There were several cases in __ast_str_helper where va_copy was not matched with a corresponding call to va_end. ........ * /, include/asterisk/strings.h: Merged revisions 246985 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r246985 | mmichelson | 2010-02-16 15:15:38 -0600 (Tue, 16 Feb 2010) | 3 lines Add some clarifying documentation to the ast_str_set and ast_str_append functions. ........ 2010-02-16 21:05 +0000 [r246902-246983] David Vossel * main/tcptls.c, /: Merged revisions 246980 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r246980 | dvossel | 2010-02-16 14:54:48 -0600 (Tue, 16 Feb 2010) | 8 lines warning message if openssl support is missing while attempting tls connection (closes issue #16673) Reported by: michaesc Patches: tls_error_msg.diff uploaded by dvossel (license 671) ........ * main/channel.c, /: Merged revisions 246899 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r246899 | dvossel | 2010-02-16 11:07:41 -0600 (Tue, 16 Feb 2010) | 16 lines fixes sample rate conversion issue with Monitor application When using ast_seekstream with the read/write streams of a monitor, the number of samples we are seeking must be of the same rate as the stream or the jump calculation will be incorrect. This patch adds logic to correctly convert the number of samples to jump to the sample rate the read/write stream is using. For example, if the call is G722 (16khz) and the read/write stream is recording a 8khz wav, seeking 320 samples of 16khz audio is not the same as seeking 320 samples of 8khz audio when performing the ast_seekstream on the stream. ABE-2044 ........ 2010-02-15 23:44 +0000 [r246712] Tilghman Lesher * Makefile, /: Merged revisions 246710 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r246710 | tilghman | 2010-02-15 17:43:28 -0600 (Mon, 15 Feb 2010) | 12 lines Merged revisions 246709 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r246709 | tilghman | 2010-02-15 17:42:33 -0600 (Mon, 15 Feb 2010) | 5 lines Make the menuselect instructions correct by allowing 'make menuselect' to actually solve dependency problems. (Previously, it would fail out again with the same message about running 'make menuselect', which was NOT at all helpful.) ........ ................ 2010-02-12 23:34 +0000 [r246548] David Vossel * main/channel.c, /: Merged revisions 246546 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r246546 | dvossel | 2010-02-12 17:32:33 -0600 (Fri, 12 Feb 2010) | 21 lines Merged revisions 246545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r246545 | dvossel | 2010-02-12 17:30:17 -0600 (Fri, 12 Feb 2010) | 16 lines lock channel during datastore removal On channel destruction the channel's datastores are removed and destroyed. Since there are public API calls to find and remove datastores on a channel, a lock should be held whenever datastores are removed and destroyed. This resolves a crash caused by a race condition in app_chanspy.c. (closes issue #16678) Reported by: tim_ringenbach Patches: datastore_destroy_race.diff uploaded by tim ringenbach (license 540) Tested by: dvossel ........ ................ 2010-02-12 19:08 +0000 [r246463] Jason Parker * main/channel.c: Fix some silly formatting that made my head hurt. 2010-02-10 21:28 +0000 [r246202-246206] Tilghman Lesher * /, funcs/func_strings.c: Merged revisions 246204 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r246204 | tilghman | 2010-02-10 15:24:10 -0600 (Wed, 10 Feb 2010) | 2 lines Fussy compiler on another machine... ........ * /, funcs/func_strings.c: Merged revisions 246200 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r246200 | tilghman | 2010-02-10 15:19:35 -0600 (Wed, 10 Feb 2010) | 2 lines Fix weird issue with unit tests on optimized build - turned out to be a signing issue. ........ 2010-02-10 17:52 +0000 [r246119] David Vossel * apps/app_queue.c, /: Merged revisions 246116 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r246116 | dvossel | 2010-02-10 11:49:34 -0600 (Wed, 10 Feb 2010) | 14 lines Merged revisions 246115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r246115 | dvossel | 2010-02-10 11:44:20 -0600 (Wed, 10 Feb 2010) | 8 lines fixes random deadlock in app_queue with use_weight during reload (closes issue #16677) Reported by: tim_ringenbach Patches: app_queue_use_weight_deadlock.diff uploaded by tim ringenbach (license 540) ........ ................ 2010-02-10 16:55 +0000 [r246072] Jeff Peeler * channels/chan_local.c, /: Merged revisions 246070 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r246070 | jpeeler | 2010-02-10 10:47:37 -0600 (Wed, 10 Feb 2010) | 22 lines Change channel state on local channels for busy,answer,ring. Previously local channels channel state never changed. This became problematic when the state of the other side of the local channel was lost, for example during a masquerade. Changing the state of the local channel allows for the scenario to be detected when the channel state is set to ringing, but the peer isn't ringing. The specific problem scenario is described in 164201. Although this was noted on one of the issues, here is the tested dialplan verified to work: exten => 9700,1,Dial(Local/*9700@default&Local/0009700@default) exten => *9700,1,Set(GLOBAL(TESTCHAN)=${CHANNEL:0:${MATH(${LEN(${CHANNEL})}-1):0:2}}1) exten => *9700,n,wait(3) ;3 works, 1 did not exten => *9700,n,Dial(SIP/5001) exten => 0009700,1,Wait(1) ;1 works, 3 did not exten => 0009700,n,ChannelRedirect(${TESTCHAN},parkedcalls,701,1) (closes issue #14992) Reported by: davidw ........ 2010-02-10 15:38 +0000 [r245947-246024] Tilghman Lesher * /, funcs/func_strings.c: Merged revisions 246022 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r246022 | tilghman | 2010-02-10 09:36:57 -0600 (Wed, 10 Feb 2010) | 2 lines Enable warnings on atypical conditions for the FILTER function (suggested by mmichelson on the -dev list). ........ * configs/extensions.conf.sample, /, funcs/func_strings.c: Merged revisions 245945 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r245945 | tilghman | 2010-02-10 08:06:12 -0600 (Wed, 10 Feb 2010) | 9 lines Merged revisions 245944 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r245944 | tilghman | 2010-02-10 07:37:13 -0600 (Wed, 10 Feb 2010) | 2 lines Include examples of FILTER usage in extension patterns where a "." may be a risk. ........ ................ 2010-02-09 23:13 +0000 [r245795] David Vossel * channels/chan_iax2.c, /: Merged revisions 245793 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r245793 | dvossel | 2010-02-09 17:07:17 -0600 (Tue, 09 Feb 2010) | 18 lines Merged revisions 245792 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r245792 | dvossel | 2010-02-09 16:55:38 -0600 (Tue, 09 Feb 2010) | 12 lines Fixes iaxs and iaxsl size off by one issue. 2^15 = 32768 which is the maximum allowed iax2 callnumber. Creating the iaxs and iaxsl array of size 32768 means the maximum callnumber is actually out of bounds. This causes a nasty crash. (closes issue #15997) Reported by: exarv Patches: iax_fix.diff uploaded by dvossel (license 671) ........ ................ 2010-02-09 18:09 +0000 [r245731] Tilghman Lesher * /, apps/app_fax.c: Merged revisions 245729 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r245729 | tilghman | 2010-02-09 12:06:30 -0600 (Tue, 09 Feb 2010) | 8 lines Ensure frames are only freed once. (closes issue #16361) Reported by: vlad Patches: 20100208__issue16361.diff.txt uploaded by tilghman (license 14) Tested by: kenny, bloodoff, misaksen ........ 2010-02-09 16:26 +0000 [r245682] Kevin P. Fleming * /, apps/app_fax.c: Merged revisions 245680 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r245680 | kpfleming | 2010-02-09 10:24:52 -0600 (Tue, 09 Feb 2010) | 8 lines Don't offer MMR or JBIG transcoding during T.38 negotiation. After further discussion with Steve Underwood, we should not (yet) be offering to receive MMR or JBIG transcoded streams from T.38 endpoints. A future spandsp release will support those features, and then they can be enabled during negotiation ........ 2010-02-08 23:45 +0000 [r245625] Russell Bryant * /, main/event.c: Merged revisions 245624 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r245624 | russell | 2010-02-08 17:43:00 -0600 (Mon, 08 Feb 2010) | 5 lines Fix return value of get_ie_str() and get_ie_str_hash() for non-existent IE. I found this bug while developing a unit test for event allocation. Testing is awesome. ........ 2010-02-08 22:46 +0000 [r245580] Tilghman Lesher * channels/Makefile, /, main/Makefile: Merged revisions 245578 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r245578 | tilghman | 2010-02-08 16:31:40 -0600 (Mon, 08 Feb 2010) | 12 lines Actually use _ASTLDFLAGS in the main/ and channels/ Makefiles. They were previously passed correctly, but they simply weren't used. This caused issues with various platforms whose builds needed to pass special linker flags via the configure script. (closes issue #16596) Reported by: pprindeville Patches: asterisk-1.6-astldflags.patch uploaded by pprindeville (license 347) Tested by: tilghman ........ 2010-02-08 20:42 +0000 [r245499] Jason Parker * main/ast_expr2.fl, /, main/ast_expr2f.c: Merged revisions 245497 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r245497 | qwell | 2010-02-08 14:41:05 -0600 (Mon, 08 Feb 2010) | 11 lines Merged revisions 245496 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r245496 | qwell | 2010-02-08 14:39:50 -0600 (Mon, 08 Feb 2010) | 4 lines Remove reference of documentation in source directory. People don't always build Asterisk from source (distro packages, anybody?). ........ ................ 2010-02-05 19:26 +0000 [r245095] Jeff Peeler * contrib/firmware (removed), /, LICENSE: Merged revisions 245090 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r245090 | jpeeler | 2010-02-05 13:26:22 -0600 (Fri, 05 Feb 2010) | 11 lines Merged revisions 245044 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r245044 | kpfleming | 2010-02-05 12:32:29 -0600 (Fri, 05 Feb 2010) | 5 lines Remove contrib/firmware directory as it is empty Remove explicit license for IAXy firmware as it is no longer included in the tree ........ ................ 2010-02-05 17:10 +0000 [r244929] Sean Bright * main/asterisk.c, /: Merged revisions 244927 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r244927 | seanbright | 2010-02-05 12:05:32 -0500 (Fri, 05 Feb 2010) | 9 lines Merged revisions 244926 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r244926 | seanbright | 2010-02-05 12:03:35 -0500 (Fri, 05 Feb 2010) | 1 line Update main copyright date. ........ ................ 2010-02-03 19:27 +0000 [r244553] Mark Michelson * main/sched.c, /: Merged revisions 244547 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r244547 | mmichelson | 2010-02-03 13:26:53 -0600 (Wed, 03 Feb 2010) | 3 lines Initialize counters in ast_sched_report so that resulting data is not bogus. ........ 2010-02-03 18:43 +0000 [r244507] Tilghman Lesher * channels/chan_dahdi.c, /: Merged revisions 244505 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r244505 | tilghman | 2010-02-03 12:34:29 -0600 (Wed, 03 Feb 2010) | 8 lines The chanvar= setting should inherit the entire list of variables, not just the first one. (closes issue #16359) Reported by: raarts Patches: dahdi-setvars.diff uploaded by raarts (license 937) Tested by: raarts ........ 2010-02-02 22:31 +0000 [r244446] David Vossel * main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h: Merged revisions 244443 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r244443 | dvossel | 2010-02-02 16:27:23 -0600 (Tue, 02 Feb 2010) | 18 lines fixes crash during T.38 negotiation caused by invalid or missing FaxMaxDatagram field AST-2010-001 (closes issue #16634) Reported by: krn (closes issue #16724) Reported by: barthpbx (closes issue #16517) Reported by: bklang (closes issue #16485) Reported by: elsto ........ 2010-02-02 20:35 +0000 [r244394] Tilghman Lesher * apps/app_dial.c, /: Merged revisions 244393 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r244393 | tilghman | 2010-02-02 14:32:29 -0600 (Tue, 02 Feb 2010) | 18 lines Properly respect GOSUB_RESULT as to what to do with the master channel. Previously, we would parse GOSUB_RESULT, but not actually do anything with it. (closes issue #16686) Reported by: bklang Patches: app_dial-respect-gosub_result.patch uploaded by bklang (license 919) (with modifications) ........ 2010-02-02 Leif Madsen * Release Asterisk 1.6.1.14 * AST-2010-001: An attacker attempting to negotiate T.38 over SIP can remotely crash Asterisk by modifying the FaxMaxDatagram field of the SDP to contain either a negative or exceptionally large value. The same crash occurs when the FaxMaxDatagram field is omitted from the SDP as well. 2010-01-14 Leif Madsen * Release Asterisk 1.6.1.13 2010-01-08 Leif Madsen * Release Asterisk 1.6.1.13-rc1 2010-01-07 21:17 +0000 [r238497] Tilghman Lesher * channels/chan_oss.c, main/poll.c, channels/chan_usbradio.c, include/asterisk/utils.h, /, channels/chan_sip.c, channels/chan_alsa.c, channels/chan_console.c: Merged revisions 209400 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209400 | kpfleming | 2009-07-28 08:49:46 -0500 (Tue, 28 Jul 2009) | 3 lines Define side-effect-safe MIN and MAX macros and remove duplicate definitions from various files. (closes issue #16251) Reported by: asgaroth ........ 2010-01-07 20:20 +0000 [r238363-238430] David Vossel * channels/chan_iax2.c, /: Merged revisions 238412 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r238412 | dvossel | 2010-01-07 14:15:27 -0600 (Thu, 07 Jan 2010) | 16 lines Merged revisions 238411 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r238411 | dvossel | 2010-01-07 14:14:25 -0600 (Thu, 07 Jan 2010) | 10 lines fixes crash in "scheduled_destroy" in chan_iax A signed short was used to represent a callnumber. This is makes it possible to attempt to access the iaxs array with a negative index. (closes issue #16565) Reported by: jensvb ........ ................ * /, channels/chan_sip.c: Merged revisions 238405 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r238405 | dvossel | 2010-01-07 14:00:31 -0600 (Thu, 07 Jan 2010) | 8 lines Change in sip show channels display format allowing more digits for CID (closes issue #16459) Reported by: Rzadzins Patches: chan_sip_longer_cid.patch uploaded by Rzadzins (license 953) ........ * apps/app_queue.c, /: Merged revisions 238361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r238361 | dvossel | 2010-01-07 12:58:23 -0600 (Thu, 07 Jan 2010) | 8 lines cli 'queue show' formatting fix. queue name was truncated over 12 characters (closes issue #16078) Reported by: RoadKill Patches: quequename_limit.patch uploaded by ppyy (license 906) Tested by: dvossel ........ 2010-01-07 12:09 +0000 [r238351] Tzafrir Cohen * /, configs/sip.conf.sample: Merged revisions 238313 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r238313 | tzafrir | 2010-01-07 11:14:57 +0200 (ה', 07 ינו 2010) | 2 lines Document the usefulness of explicit udp:// in the register string ........ 2010-01-06 21:48 +0000 [r238233] Tilghman Lesher * /, funcs/func_cdr.c: Merged revisions 238231 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r238231 | tilghman | 2010-01-06 15:45:17 -0600 (Wed, 06 Jan 2010) | 11 lines Merged revisions 238230 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r238230 | tilghman | 2010-01-06 15:41:55 -0600 (Wed, 06 Jan 2010) | 4 lines Revise documentation on disposition values to the actual values used. (closes issue #16289) Reported by: wdoekes ........ ................ 2010-01-06 20:39 +0000 [r238136-238183] Jeff Peeler * apps/app_meetme.c: Merged revisions 238181 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r238181 | jpeeler | 2010-01-06 14:37:18 -0600 (Wed, 06 Jan 2010) | 8 lines Fix misreverting from 177158. (closes issue #15725) Reported by: shanermn Patches: v1-15725.patch uploaded by dimas (license 88) Tested by: shanermn ........ * /, main/features.c: Merged revisions 238134 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r238134 | jpeeler | 2010-01-06 13:05:06 -0600 (Wed, 06 Jan 2010) | 10 lines Fix channel name comparison for bridge application. The channel name comparison was not comparing the whole string and therefore if one channel name was a substring of the other, the bridge would fail. (closes issue #16528) Reported by: telecos82 Patches: res_features_r236843.diff uploaded by telecos82 (license 687) ........ 2010-01-06 15:21 +0000 [r238012] Russell Bryant * /, apps/app_mp3.c: Merged revisions 238010 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r238010 | russell | 2010-01-06 09:19:10 -0600 (Wed, 06 Jan 2010) | 14 lines Merged revisions 238009 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r238009 | russell | 2010-01-06 09:18:22 -0600 (Wed, 06 Jan 2010) | 7 lines Resolve a crash due to an ast_frame not being fully initialized. (closes issue #16531) Reported by: john8675309 (closes SWP-615) ........ ................ 2010-01-06 06:52 +0000 [r237967] Tilghman Lesher * channels/chan_sip.c: One duplicate setting here (dead code). 2010-01-05 23:09 +0000 [r237841-237922] David Vossel * apps/app_queue.c, /: Merged revisions 237920 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r237920 | dvossel | 2010-01-05 17:08:50 -0600 (Tue, 05 Jan 2010) | 16 lines fixes holdtime playback issue in app_queue When reporting hold time, the number of seconds should be mod 60. Otherwise audio playback could be something like "2 minutes 123 seconds" rather than "2 minutes 3 seconds". Also, the "minute" sound file is missing, so for the moment until that file can be created the "minutes" file is used instead. (closes issue #16168) Reported by: nickilo Patches: patch-unified-trunk-rev-222176 uploaded by nickilo (license ) Tested by: nickilo, wonderg ........ * main/pbx.c, /: Merged revisions 237839 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r237839 | dvossel | 2010-01-05 13:29:47 -0600 (Tue, 05 Jan 2010) | 19 lines fixes subscriptions being lost after 'module reload' During a module reload if multiple extension configs are present, such as both extensions.conf and extensions.ael, watchers for one config's hints will be lost during the merging of the other config. This happens because hint watchers are only preserved for the current config being merged. The old context list is destroyed after the merging takes place, meaning any watchers that were not perserved will be removed. Now all hints are preserved during merging regardless of what config file is being merged. These hints are only restored if they are present within the new context list. (closes issue #16093) Reported by: jlaroff ........ 2010-01-05 17:22 +0000 [r237724] Russell Bryant * /, main/utils.c: Merged revisions 237699 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r237699 | russell | 2010-01-05 11:16:01 -0600 (Tue, 05 Jan 2010) | 14 lines Merged revisions 237697 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r237697 | russell | 2010-01-05 11:13:28 -0600 (Tue, 05 Jan 2010) | 7 lines Change a NOTICE log message to DEBUG where it belongs. (closes issue #16479) Reported by: alexrecarey (closes SWP-577) ........ ................ 2010-01-04 21:52 +0000 [r237408-237576] Tilghman Lesher * /, main/say.c: Merged revisions 237574 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r237574 | tilghman | 2010-01-04 15:48:20 -0600 (Mon, 04 Jan 2010) | 13 lines Merged revisions 237573 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r237573 | tilghman | 2010-01-04 15:45:46 -0600 (Mon, 04 Jan 2010) | 6 lines Bounds checking for input string (closes issue #16407) Reported by: qwell Patches: 20100104__issue16407.diff.txt uploaded by tilghman (license 14) ........ ................ * main/pbx.c, /: Merged revisions 237494 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r237494 | tilghman | 2010-01-04 14:59:01 -0600 (Mon, 04 Jan 2010) | 15 lines Merged revisions 237493 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r237493 | tilghman | 2010-01-04 14:57:35 -0600 (Mon, 04 Jan 2010) | 8 lines Regression in issue #15421 - Pattern matching (closes issue #16482) Reported by: wdoekes Patches: astsvn-16482-betterfix.diff uploaded by wdoekes (license 717) 20091223__issue16482.diff.txt uploaded by tilghman (license 14) Tested by: wdoekes, tilghman ........ ................ * main/config.c, /: Merged revisions 237414 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r237414 | tilghman | 2010-01-04 13:03:20 -0600 (Mon, 04 Jan 2010) | 2 lines Oops, didn't compile (thanks, kpfleming) ........ * main/config.c, /: Merged revisions 237410 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r237410 | tilghman | 2010-01-04 12:42:10 -0600 (Mon, 04 Jan 2010) | 7 lines Further reduce the encoded blank values back to blank in the realtime API. (closes issue #16533) Reported by: sergee Patches: 200100104__issue16533.diff.txt uploaded by tilghman (license 14) Tested by: sergee ........ * main/pbx.c, /, res/res_agi.c, include/asterisk/channel.h: Merged revisions 237406 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r237406 | tilghman | 2010-01-04 12:28:28 -0600 (Mon, 04 Jan 2010) | 23 lines Merged revisions 237405 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r237405 | tilghman | 2010-01-04 12:19:00 -0600 (Mon, 04 Jan 2010) | 16 lines Add a flag to disable the Background behavior, for AGI users. This is in a section of code that relates to two other issues, namely issue #14011 and issue #14940), one of which was the behavior of Background when called with a context argument that matched the current context. This fix broke FreePBX, however, in a post-Dial situation. Needless to say, this is an extremely difficult collision of several different issues. While the use of an exception flag is ugly, fixing all of the issues linked is rather difficult (although if someone would like to propose a better solution, we're happy to entertain that suggestion). (closes issue #16434) Reported by: rickead2000 Patches: 20091217__issue16434.diff.txt uploaded by tilghman (license 14) 20091222__issue16434__1.6.1.diff.txt uploaded by tilghman (license 14) Tested by: rickead2000 ........ ................ 2010-01-04 16:51 +0000 [r237329] David Vossel * apps/app_queue.c, /: Merged revisions 237327 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r237327 | dvossel | 2010-01-04 10:39:11 -0600 (Mon, 04 Jan 2010) | 10 lines app_queue segfaults if realtime field uniqueid is NULL (closes issue #16385) Reported by: haakon Patches: app_queue.c.patch uploaded by haakon (license 880) app_queue.c.patch_v2 uploaded by dvossel (license 671) Tested by: haakon ........ 2010-01-04 16:26 +0000 [r237325] Jeff Peeler * /, res/res_agi.c: Merged revisions 237323 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r237323 | jpeeler | 2010-01-04 10:24:51 -0600 (Mon, 04 Jan 2010) | 5 lines Fix timeout for AGI command speech recognize. (closes issue #16297) Reported by: semond ........ 2010-01-04 16:21 +0000 [r237321] Tilghman Lesher * channels/chan_local.c, /: Merged revisions 237319 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r237319 | tilghman | 2010-01-04 10:20:03 -0600 (Mon, 04 Jan 2010) | 10 lines Merged revisions 237318 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r237318 | tilghman | 2010-01-04 10:18:59 -0600 (Mon, 04 Jan 2010) | 3 lines It's also possible for the Local channel to directly execute an Application. Reviewboard: https://reviewboard.asterisk.org/r/452/ ........ ................ 2010-01-02 10:01 +0000 [r237138] Olle Johansson * /, channels/chan_sip.c: Merged revisions 237136 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r237136 | oej | 2010-01-02 10:54:22 +0100 (Lör, 02 Jan 2010) | 10 lines Merged revisions 237135 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r237135 | oej | 2010-01-02 10:52:30 +0100 (Lör, 02 Jan 2010) | 2 lines Release memory of the contact acl before unloading module ........ ................ 2009-12-30 22:00 +0000 [r236984] Tilghman Lesher * channels/chan_local.c, /: Merged revisions 236982 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r236982 | tilghman | 2009-12-30 15:59:18 -0600 (Wed, 30 Dec 2009) | 16 lines Merged revisions 236981 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236981 | tilghman | 2009-12-30 15:57:10 -0600 (Wed, 30 Dec 2009) | 9 lines Don't queue frames to channels that have no means to process them. (closes issue #15609) Reported by: aragon Patches: 20091230__issue16521__1.4__chan_local_only.diff.txt uploaded by tilghman (license 14) Tested by: aragon Review: https://reviewboard.asterisk.org/r/452/ ........ ................ 2009-12-30 21:13 +0000 [r236904] Jeff Peeler * /, utils/ael_main.c: Merged revisions 236902 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r236902 | jpeeler | 2009-12-30 15:09:28 -0600 (Wed, 30 Dec 2009) | 2 lines One more LOW_MEMORY compile fix. ........ 2009-12-30 17:57 +0000 [r236803-236851] Tilghman Lesher * /, cdr/cdr_adaptive_odbc.c: Merged revisions 236847 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r236847 | tilghman | 2009-12-30 11:53:29 -0600 (Wed, 30 Dec 2009) | 4 lines When the field is blank, don't warn about the field being unable to be coerced, just skip the column. (closes http://lists.digium.com/pipermail/asterisk-dev/2009-December/041362.html) Reported by Nic Colledge on the -dev list, fixed by me. ........ * /, channels/chan_sip.c: Merged revisions 236802 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r236802 | tilghman | 2009-12-29 17:05:45 -0600 (Tue, 29 Dec 2009) | 7 lines Shut down the SIP session timers more gracefully, in order to prevent a possible crash. (closes issue #16452) Reported by: corruptor Patches: 20091221__issue16452.diff.txt uploaded by tilghman (license 14) Tested by: corruptor ........ 2009-12-28 22:11 +0000 [r236715] Jason Parker * main/ast_expr2.c, /, main/ast_expr2.y: Merged revisions 236713 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r236713 | qwell | 2009-12-28 16:09:40 -0600 (Mon, 28 Dec 2009) | 8 lines Allow "REMAINDER" to function properly in expressions. (closes issue #16427) Reported by: wdoekes Patches: ast16-reminder-remainder.patch uploaded by wdoekes (license 717) Tested by: wdoekes ........ 2009-12-28 17:40 +0000 [r236669] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 236667 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r236667 | tilghman | 2009-12-28 11:37:46 -0600 (Mon, 28 Dec 2009) | 4 lines Use recommended option, not deprecated option. (closes issue #16515) Reported by: ManChicken ........ 2009-12-28 15:31 +0000 [r236512-236634] Sean Bright * include/asterisk/threadstorage.h, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 236613 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r236613 | seanbright | 2009-12-28 10:22:54 -0500 (Mon, 28 Dec 2009) | 14 lines Merged revisions 236585 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236585 | seanbright | 2009-12-28 10:12:08 -0500 (Mon, 28 Dec 2009) | 7 lines Try a test compile to see if PTHREAD_ONCE_INIT requires extra braces. There was conditional code (based on build platform) to optioinally wrap PTHREAD_ONCE_INIT in braces that was removed since it is fixed in newer versions of Solaris/OpenSolaris, but I am still running into it on Solaris 10 x86 so add a configure-time check for it. ........ ................ * /, apps/app_meetme.c: Merged revisions 236510 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r236510 | seanbright | 2009-12-28 07:44:58 -0500 (Mon, 28 Dec 2009) | 19 lines Merged revisions 236509 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236509 | seanbright | 2009-12-28 07:43:36 -0500 (Mon, 28 Dec 2009) | 12 lines Avoid a crash with large numbers of MeetMe conferences. Similar to changes made to Queue(), when we have large numbers of conferences in meetme.conf (1000s) and we use alloca()/strdupa(), we can blow out the stack and crash, so instead just use a single fixed buffer. (closes issue #16509) Reported by: Kashif Raza Patches: 20091223_16509.patch uploaded by seanbright (license 71) Tested by: seanbright ........ ................ 2009-12-27 18:22 +0000 [r236436] Tilghman Lesher * contrib/init.d/rc.debian.asterisk, /: Merged revisions 236434 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r236434 | tilghman | 2009-12-27 12:20:53 -0600 (Sun, 27 Dec 2009) | 9 lines Merged revisions 236433 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236433 | tilghman | 2009-12-27 12:19:38 -0600 (Sun, 27 Dec 2009) | 2 lines Turn on colors in the daemon, since there's many requests for it on Ubuntu. ........ ................ 2009-12-26 15:31 +0000 [r236360] Kevin P. Fleming * sounds/Makefile, /: Merged revisions 236358 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r236358 | kpfleming | 2009-12-26 09:27:44 -0600 (Sat, 26 Dec 2009) | 9 lines Merged revisions 236357 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236357 | kpfleming | 2009-12-26 09:26:17 -0600 (Sat, 26 Dec 2009) | 1 line update to latest releases with zero uid/gid ........ ................ 2009-12-23 18:26 +0000 [r236188-236302] Tilghman Lesher * apps/app_stack.c, /: Merged revisions 236300 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r236300 | tilghman | 2009-12-23 12:25:27 -0600 (Wed, 23 Dec 2009) | 7 lines AGI may be invoked from outside the dialplan (closes issue #16510) Reported by: atis Patches: 20091223__issue16510.diff.txt uploaded by tilghman (license 14) Tested by: atis ........ * /, res/res_agi.c: Merged revisions 236186 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r236186 | tilghman | 2009-12-22 21:07:48 -0600 (Tue, 22 Dec 2009) | 11 lines Merged revisions 236184 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236184 | tilghman | 2009-12-22 20:55:24 -0600 (Tue, 22 Dec 2009) | 4 lines If EXEC only gets a single argument, don't crash when the second is used. (closes issue #16504) Reported by: bklang ........ ................ 2009-12-22 17:06 +0000 [r236065] David Vossel * /, channels/chan_sip.c: Merged revisions 236063 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r236063 | dvossel | 2009-12-22 11:00:08 -0600 (Tue, 22 Dec 2009) | 18 lines Merged revisions 236062 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236062 | dvossel | 2009-12-22 10:58:19 -0600 (Tue, 22 Dec 2009) | 11 lines fixes issue with p->method incorrectly set to ACK It is possible for a second ACK to come in for a retransmitted message. If an ack does not match an unacked message in our queue, restore the previous p->method as this ACK is completely ignored. (closes issue #16295) Reported by: omolenkamp Patches: issue16295_v2.diff uploaded by dvossel (license 671) ........ ................ 2009-12-21 19:55 +0000 [r235943] Jeff Peeler * res/res_monitor.c, /: Merged revisions 235941 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r235941 | jpeeler | 2009-12-21 13:54:20 -0600 (Mon, 21 Dec 2009) | 20 lines Merged revisions 235940 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r235940 | jpeeler | 2009-12-21 13:43:41 -0600 (Mon, 21 Dec 2009) | 13 lines Change Monitor to not assume file to write to does not contain pathing. 227944 changed the fname_base argument to always append the configured monitor path. This change was necessary to properly compare files for uniqueness. If a full path is given though, nothing needs to be appended and that is handled correctly now. (closes issue #16377) (closes issue #16376) Reported by: bcnit Patches: res_monitor.c-issue16376-1.patch uploaded by dant (license 670) ........ ................ 2009-12-21 17:11 +0000 [r235825] Tilghman Lesher * /, main/features.c: Merged revisions 235822 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r235822 | tilghman | 2009-12-21 11:00:46 -0600 (Mon, 21 Dec 2009) | 15 lines Merged revisions 235821 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r235821 | tilghman | 2009-12-21 10:45:03 -0600 (Mon, 21 Dec 2009) | 8 lines Send parking lot announcement to the channel which parked the call, not the park-ee. (closes issue #16234) Reported by: yeshuawatso Patches: 20091210__issue16234.diff.txt uploaded by tilghman (license 14) 20091221__issue16234__1.4.diff.txt uploaded by tilghman (license 14) Tested by: yeshuawatso ........ ................ 2009-12-20 09:07 +0000 [r235776] Alec L Davis * main/dsp.c: restarts busydetector (if enabled) when DTMF is received after call is bridged. (closes issue #16389) Reported by: alecdavis Tested by: alecdavis Patch dtmf_busydetector.diff2.txt uploaded by alecdavis (license 585) 2009-12-18 23:03 +0000 [r235663] Jeff Peeler * main/channel.c, /, include/asterisk/cdr.h: Merged revisions 235660 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r235660 | jpeeler | 2009-12-18 16:51:37 -0600 (Fri, 18 Dec 2009) | 55 lines Merged revisions 235635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r235635 | jpeeler | 2009-12-18 16:29:51 -0600 (Fri, 18 Dec 2009) | 48 lines Correct CDR dispositions for BUSY/FAILED This patch is simple in that it reorders the disposition defines so that the fix for issue 12946 works properly (the default CDR disposition was changed to AST_CDR_NOANSWER). Also, the AST_CDR_FLAG_ORIGINATED flag was set in ast_call to ensure all CDR records are written. The side effects of CDR changes are scary, so I'm documenting the test cases performed to attempt to catch any regressions. The following tests were all performed using 1.4 rev 195881 vs head (235571) + patch: A calls B C calls B (busy) Hangup C Hangup A (Both SIP and features) A calls B A blind transfers to C Hangup C (Both SIP and features) A calls B A attended transfers to C Hangup C A calls B A attended transfers to C (SIP) C blind transfers to A (features) Hangup A All of the test scenario CDRs matched. The following tests were performed just with the patch to ensure proper operation (with unanswered=yes): exten =>s,1,Answer exten =>s,n,ResetCDR(w) exten =>s,n,ResetCDR(w) exten =>s,1,ResetCDR(w) exten =>s,n,ResetCDR(w) (closes issue #16180) Reported by: aatef Patches: bug16180.patch uploaded by jpeeler (license 325) ........ ................ 2009-12-18 22:42 +0000 [r235575-235658] Tilghman Lesher * /, configure, configure.ac: Merged revisions 235656 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r235656 | tilghman | 2009-12-18 16:40:46 -0600 (Fri, 18 Dec 2009) | 9 lines Merged revisions 235652 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r235652 | tilghman | 2009-12-18 16:39:30 -0600 (Fri, 18 Dec 2009) | 2 lines Revise verbiage, per #asterisk-dev discussion ........ ................ * /, configure, configure.ac: Merged revisions 235573 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r235573 | tilghman | 2009-12-18 15:19:43 -0600 (Fri, 18 Dec 2009) | 9 lines Merged revisions 235572 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r235572 | tilghman | 2009-12-18 15:18:16 -0600 (Fri, 18 Dec 2009) | 2 lines Point to the typical missing package, not the cryptic "termcap support". ........ ................ 2009-12-17 Leif Madsen * Release Asterisk 1.6.1.12 2009-12-09 Leif Madsen * Release Asterisk 1.6.1.12-rc1 2009-12-08 18:31 +0000 [r233730] Tilghman Lesher * /, res/res_musiconhold.c: Merged revisions 233718 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r233718 | tilghman | 2009-12-08 12:22:44 -0600 (Tue, 08 Dec 2009) | 8 lines Find another ref leak and change how we manage module references. (closes issue #16388) Reported by: parisioa Patches: 20091208__issue16388.diff.txt uploaded by tilghman (license 14) Tested by: parisioa, tilghman Review: https://reviewboard.asterisk.org/r/442/ ........ 2009-12-08 18:02 +0000 [r233693] Russell Bryant * formats/format_ilbc.c, formats/format_vox.c, formats/format_pcm.c, formats/format_g723.c, formats/format_h263.c, formats/format_h264.c, formats/format_g726.c, formats/format_jpeg.c, formats/format_gsm.c, formats/format_g729.c, /, formats/format_sln.c, formats/format_wav.c, formats/format_ogg_vorbis.c, formats/format_sln16.c, formats/format_wav_gsm.c: Merged revisions 233692 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r233692 | russell | 2009-12-08 12:00:16 -0600 (Tue, 08 Dec 2009) | 16 lines Set a module load priority for format modules. A recent change to app_voicemail made it such that the module now assumes that all format modules are available while processing voicemail configuration. However, when autoloading modules, it was possible that app_voicemail was loaded before the format modules. Since format modules don't depend on anything, set a module load priority on them to ensure that they get loaded first when autoloading. This fix applies to trunk, 1.6.1, and 1.6.2. The fix for 1.4 and 1.6.0 will require a different approach since the module load priority functionality is not present in the module API. (issue #16412) Reported by: jiddings ........ 2009-12-08 07:40 +0000 [r233688] TransNexus OSP Development * apps/app_osplookup.c: Fixed compile error with OSP Toolkit 3.6. 2009-12-07 23:56 +0000 [r233616] Atis Lezdins * contrib/valgrind.supp, /: Merged revisions 233577 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r233577 | atis | 2009-12-08 01:10:13 +0200 (Tue, 08 Dec 2009) | 8 lines Fix compatibility with valgrind 3.3 and older. (noticed in issue #16388) Reported by: parisioa Patches: valgrind.supp uloaded by atis (license 242) Tested by: atis, parisioa ........ 2009-12-07 23:29 +0000 [r233474-233613] David Vossel * /, main/utils.c: Merged revisions 233611 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r233611 | dvossel | 2009-12-07 17:28:51 -0600 (Mon, 07 Dec 2009) | 4 lines fixes incorrect logic in ast_uri_encode issue #16299 ........ * /, channels/chan_sip.c: Merged revisions 233472 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r233472 | dvossel | 2009-12-07 12:08:46 -0600 (Mon, 07 Dec 2009) | 15 lines Merged revisions 233471 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r233471 | dvossel | 2009-12-07 12:07:38 -0600 (Mon, 07 Dec 2009) | 9 lines fixes missing Contact header angle brackets (closes issue #16298) Reported by: mgernoth Patches: reg_parse_issue_1.4.diff uploaded by dvossel (license 671) Tested by: dvossel ........ ................ 2009-12-07 16:16 +0000 [r233395] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 233394 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r233394 | mnicholson | 2009-12-07 10:14:42 -0600 (Mon, 07 Dec 2009) | 8 lines Do not reject SDP packets describing only non audio streams. (closes issue #16387) Reported by: zalex1953 Patches: media-level-c-fix1.diff uploaded by mnicholson (license 96) Tested by: mnicholson, zalex1953 ........ 2009-12-04 21:55 +0000 [r233283] David Vossel * configs/iax.conf.sample, /: Merged revisions 233280 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r233280 | dvossel | 2009-12-04 15:54:44 -0600 (Fri, 04 Dec 2009) | 14 lines Merged revisions 233279 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r233279 | dvossel | 2009-12-04 15:54:01 -0600 (Fri, 04 Dec 2009) | 7 lines clarify requirecalltoken option in iax.sample.conf (closes issue #16223) Reported by: bklang Patches: clarify-iax-requirecalltoken.patch uploaded by bklang (license 919) ........ ................ 2009-12-04 21:00 +0000 [r233238] Matthias Nick * pbx/pbx_config.c, /: Merged revisions 233093 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r233093 | mnick | 2009-12-04 11:15:47 -0600 (Fri, 04 Dec 2009) | 8 lines Parse global variables or expressions in hint extensions Parse global variables or expressions in hint extensions. Like: exten => 400,hint,DAHDI/i2/${GLOBAL(var)} (closes issue #16166) Reported by: rmudgett Tested by: mnick, rmudgett ........ 2009-12-04 17:37 +0000 [r233166] David Vossel * apps/app_voicemail.c, /: Merged revisions 233121 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r233121 | dvossel | 2009-12-04 11:22:31 -0600 (Fri, 04 Dec 2009) | 12 lines Merged revisions 233116 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r233116 | dvossel | 2009-12-04 11:21:34 -0600 (Fri, 04 Dec 2009) | 6 lines document and rename strip_control() in app_voicemail (closes issue #16291) Reported by: wdoekes ........ ................ 2009-12-04 17:22 +0000 [r233122] Russell Bryant * main/channel.c, /: Merged revisions 233100 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r233100 | russell | 2009-12-04 11:18:22 -0600 (Fri, 04 Dec 2009) | 14 lines Merged revisions 233092 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r233092 | russell | 2009-12-04 11:12:47 -0600 (Fri, 04 Dec 2009) | 7 lines Only do frame payload check for HOLD frames. This code was added for helping to debug the source of invalid HOLD frames. However, a side effect of this is that it will incorrectly report errors for frames that have an integer payload. Make the check for this block specific to the HOLD frame case. ........ ................ 2009-12-04 15:51 +0000 [r233048] Matthias Nick * main/dsp.c, /: Merged revisions 233046 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r233046 | mnick | 2009-12-04 09:38:33 -0600 (Fri, 04 Dec 2009) | 17 lines Merged revisions 233014 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r233014 | mnick | 2009-12-04 09:17:03 -0600 (Fri, 04 Dec 2009) | 11 lines Warning message gets displayed only once Added additional field 'int display_inband_dtmf_warning', which when set to '1' displays the warning ('Inband DTMF is not supported on codec %s. Use RFC2833'), and when set to '0' doesn't display the warning. Otherwise you would get hundreds of warnings every second. (closes issue #15769) Reported by: falves11 Patches: patch_15769_14.txt uploaded by mnick (license 874) Tested by: mnick, falves11 ........ ................ 2009-12-03 21:03 +0000 [r232865] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 232854 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r232854 | tilghman | 2009-12-03 14:47:07 -0600 (Thu, 03 Dec 2009) | 15 lines Merged revisions 232820 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232820 | tilghman | 2009-12-03 14:10:19 -0600 (Thu, 03 Dec 2009) | 8 lines Deprecate "cz" in favor of "cs". Also, change the use of language codes so that language registers as a prefix, rather than an exact match. (closes issue #16272) Reported by: patrol-cz Patches: 20091203__issue16272.diff.txt uploaded by tilghman (license 14) ........ ................ 2009-12-03 15:03 +0000 [r232812] David Ruggles * apps/app_externalivr.c: Merged revisions 232587 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r232587 | diruggles | 2009-12-02 17:17:22 -0500 (Wed, 02 Dec 2009) | 12 lines Prevent double closing of FDs by EIVR This caused a problem when asterisk was under heavy load and running both AGI and EIVR applications. EIVR would close an FD at which point it would be considered freed and be used by a new AGI instance the second close would then close the FD now in use by AGI. (closes issue #16305) Reported by: diLLec Tested by: thedavidfactor, diLLec Review: https://reviewboard.asterisk.org/r/436/ ........ 2009-12-03 00:18 +0000 [r232666] Tilghman Lesher * /, res/res_musiconhold.c: Recorded merge of revisions 232660-232661 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r232660 | tilghman | 2009-12-02 18:08:55 -0600 (Wed, 02 Dec 2009) | 19 lines Fix multiple issues with musiconhold, which led to classes not getting destroyed properly. * Classes are now tracked past removal from the core container, and module removal is actively prevented until all references are freed. * A hanging reference stored in the channel has been removed. This could have caused a mismatch and the music state not properly cleared, if two or more reloads occurred between MOH being stopped and MOH being restarted. * In certain circumstances, duplicate classes were possible. * A race existed at reload time between a process being killed and the thread responsible for reading from the related pipe respawning that process. * Several reference counts have also been corrected. At least one could have caused deleted classes to stick around forever, consuming resources. This originally manifested as MOH external processes that were not killed at reload time. (closes issue #16279, closes issue #16207) Reported by: parisioa, dcabot Patches: 20091202__issue16279__2.diff.txt uploaded by tilghman (license 14) Tested by: parisioa, tilghman ........ r232661 | tilghman | 2009-12-02 18:09:36 -0600 (Wed, 02 Dec 2009) | 2 lines Remove debugging line ........ 2009-12-02 22:04 +0000 [r232578-232584] Jeff Peeler * main/manager.c, /: Merged revisions 232582 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r232582 | jpeeler | 2009-12-02 16:02:43 -0600 (Wed, 02 Dec 2009) | 14 lines Merged revisions 232581 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232581 | jpeeler | 2009-12-02 15:57:42 -0600 (Wed, 02 Dec 2009) | 7 lines Send ack (response/message) after receiving manager action userevent (closes issue #16264) Reported by: dimas Patches: event-ack.patch uploaded by dimas (license 88) ........ ................ * main/manager.c, /: Merged revisions 232576 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r232576 | jpeeler | 2009-12-02 15:32:50 -0600 (Wed, 02 Dec 2009) | 8 lines Make manager response to "Action: events" finish with empty line (closes issue #16275) Reported by: vnovy Patches: manager.c.diff uploaded by vnovy (license 922) ........ 2009-12-02 17:10 +0000 [r232358] Joshua Colp * /, apps/app_amd.c: Merged revisions 232356 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r232356 | file | 2009-12-02 13:06:54 -0400 (Wed, 02 Dec 2009) | 12 lines Merged revisions 232355 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232355 | file | 2009-12-02 13:04:52 -0400 (Wed, 02 Dec 2009) | 5 lines Fix a bug where if you hung up very quickly after calling AMD it would overwrite the AMDSTATUS of HANGUP with TOOLONG. (closes issue #16239) Reported by: CGMChris ........ ................ 2009-12-02 17:02 +0000 [r232353] David Vossel * /, main/acl.c: Merged revisions 232351 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r232351 | dvossel | 2009-12-02 11:00:15 -0600 (Wed, 02 Dec 2009) | 12 lines Merged revisions 232350 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232350 | dvossel | 2009-12-02 10:59:18 -0600 (Wed, 02 Dec 2009) | 6 lines ast_outaddrfor doesn't do htons() on port, looks odd in strace. (closes issue #16290) Reported by: wdoekes ........ ................ 2009-12-02 16:42 +0000 [r232347] Joshua Colp * /, channels/chan_sip.c: Merged revisions 232345 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r232345 | file | 2009-12-02 12:40:14 -0400 (Wed, 02 Dec 2009) | 7 lines Add support for handling the 415 Unsupported media type response like we do for a 488 Not acceptable here response. (closes issue #16186) Reported by: atis Patches: sip_t38_response_415.patch uploaded by atis (license 242) ........ 2009-12-02 15:43 +0000 [r232271] David Vossel * funcs/func_groupcount.c, /: Merged revisions 232269 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r232269 | dvossel | 2009-12-02 09:42:54 -0600 (Wed, 02 Dec 2009) | 15 lines Merged revisions 232268 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232268 | dvossel | 2009-12-02 09:41:36 -0600 (Wed, 02 Dec 2009) | 9 lines fixes segfault in func_groupcount closes issue #16337) Reported by: Parantido Patches: issue_16337.diff uploaded by dvossel (license 671) Tested by: Parantido, dvossel ........ ................ 2009-12-02 14:55 +0000 [r232231] Joshua Colp * /, channels/chan_sip.c: Merged revisions 232230 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r232230 | file | 2009-12-02 10:54:28 -0400 (Wed, 02 Dec 2009) | 5 lines Fix a bug where a scheduled item ID would get retained on registrations in a certain scenario causing code to execute during reload that should not. (issue AST-263) ........ 2009-12-02 00:51 +0000 [r232093] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 232091 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r232091 | jpeeler | 2009-12-01 18:45:18 -0600 (Tue, 01 Dec 2009) | 17 lines Merged revisions 232090 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232090 | jpeeler | 2009-12-01 18:42:58 -0600 (Tue, 01 Dec 2009) | 10 lines Do not modify the gain settings on data calls. (The digital flag actually represents a data call.) (closes issue #15972) Reported by: udosw Patches: transcap_digital_fix.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis ........ ................ 2009-12-01 23:39 +0000 [r232010-232014] Russell Bryant * /, funcs/func_lock.c: Merged revisions 232012 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r232012 | russell | 2009-12-01 17:38:34 -0600 (Tue, 01 Dec 2009) | 2 lines Fix a build error on FreeBSD. ........ * /, main/file.c: Merged revisions 232008 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r232008 | russell | 2009-12-01 17:27:53 -0600 (Tue, 01 Dec 2009) | 9 lines Merged revisions 232007 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232007 | russell | 2009-12-01 17:25:36 -0600 (Tue, 01 Dec 2009) | 2 lines Fix a warning pointed out by buildbot. ........ ................ 2009-12-01 22:00 +0000 [r231929] Jeff Peeler * main/channel.c, /: Merged revisions 231927 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r231927 | jpeeler | 2009-12-01 15:54:21 -0600 (Tue, 01 Dec 2009) | 19 lines Merged revisions 231911 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231911 | jpeeler | 2009-12-01 15:29:31 -0600 (Tue, 01 Dec 2009) | 12 lines Fix crash with invalid frame data The crash was happening as a result of a frame containing an invalid data pointer, but was set with data length of zero. The few times the issue was reproduced it _seemed_ that the frame was queued properly, that is the data pointer was set to NULL. I never could reproduce the crash so as a last resort the crash has been fixed, but a check in __ast_read has been added to give as much information about the source of problematic frames in the future. (closes issue #16058) Reported by: atis ........ ................ 2009-12-01 21:21 +0000 [r231876] David Vossel * main/pbx.c, /: Merged revisions 231867 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r231867 | dvossel | 2009-12-01 15:20:19 -0600 (Tue, 01 Dec 2009) | 9 lines Merged revisions 231853 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231853 | dvossel | 2009-12-01 15:14:31 -0600 (Tue, 01 Dec 2009) | 3 lines WaitExten m option with no parameters generates frame with zero datalen but non-null data ptr ........ ................ 2009-12-01 15:48 +0000 [r231742] Matthew Nicholson * /, main/file.c: Merged revisions 231741 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r231741 | mnicholson | 2009-12-01 09:47:36 -0600 (Tue, 01 Dec 2009) | 9 lines Merged revisions 231740 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231740 | mnicholson | 2009-12-01 09:34:57 -0600 (Tue, 01 Dec 2009) | 2 lines Ignore unknown formats in ast_format_str_reduce() and return an error if no know formats are found. ........ ................ 2009-11-30 21:55 +0000 [r231694] Kevin P. Fleming * main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h: Merged revisions 231692 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r231692 | kpfleming | 2009-11-30 15:47:42 -0600 (Mon, 30 Nov 2009) | 22 lines Another round of UDPTL stack fixes/improvements: 1) Allow users of UDPTL stack to associate a character-string tag with a UDPTL session, so that log/error/debug messages generated by the UDPTL stack can be 'connected' to the endpoint that caused them to be generated. 2) Improve comments (and process) of calculating the far end's maximum IFP size when redundancy mode is in use for error correction. 3) When an IFP larger than the calculated 'far max IFP' size is presented for writing, truncate it rather than putting in the buffer and allowing the buffer to overflow; this will cause the ends to retrain to a lower bit rate that produces IFPs of an appropriate size if possible, and if not possible, the FAX transfer will fail completely. In these cases, it is due to the one endpoint supplying a T38FaxMaxDatagram value that is improperly calculated and is too low to be of use; we have configuration options available to override this behavior. 4) Eliminate use of T38FaxMaxDatagram value in udptl.conf; it is no longer needed. ........ 2009-11-30 21:36 +0000 [r231690] Matthew Nicholson * apps/app_voicemail.c, include/asterisk/file.h, /, main/file.c, main/app.c: Merged revisions 231688 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r231688 | mnicholson | 2009-11-30 15:31:55 -0600 (Mon, 30 Nov 2009) | 15 lines Merged revisions 231614 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov 2009) | 8 lines Remove duplicate entries from voicemail format lists. This prevents app_voicemail from entering an infinite loop when the same format is specified twice in the format list. (closes issue #15625) Reported by: Shagg63 Tested by: mnicholson Review: https://reviewboard.asterisk.org/r/429/ ........ ................ 2009-11-30 20:58 +0000 [r231608] Tilghman Lesher * apps/app_queue.c: Turn off debug mode in 1.6.1; fix such that debug mode and non-debug mode functions return the same types. (Fixes an issue brought up in chat by twilson) 2009-11-30 20:47 +0000 [r231604] Joshua Colp * /, channels/chan_sip.c: Merged revisions 231602 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r231602 | file | 2009-11-30 16:44:30 -0400 (Mon, 30 Nov 2009) | 5 lines When receiving SDP that matches the version of the last one do not treat it as a fatal error. (closes issue #16238) Reported by: seandarcy ........ 2009-11-30 18:57 +0000 [r231512-231559] David Vossel * apps/app_queue.c, /: Merged revisions 231556 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r231556 | dvossel | 2009-11-30 12:55:07 -0600 (Mon, 30 Nov 2009) | 11 lines app_queue crashes randomly, often during call-transfers This patch adds a ref to the queue_ent object's parent call_queue in queue_exec() so the call_queue won't be destroyed while the the queue_ent still holds a pointer to it. (closes issue 0015686) Tested by: dvossel, aragon ........ * main/rtp.c, /: Merged revisions 231491 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r231491 | dvossel | 2009-11-30 11:28:28 -0600 (Mon, 30 Nov 2009) | 17 lines Merged revisions 231441 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231441 | dvossel | 2009-11-30 11:14:08 -0600 (Mon, 30 Nov 2009) | 11 lines fixes crash caused by RTP comfort noise payload greater than 24 bytes AST-2009-010 (closes issue #16242) Reported by: amorsen Patches: issue16242.diff uploaded by oej (license 306) Tested by: amorsen, oej, dvossel ........ ................ 2009-11-25 22:34 +0000 [r231301] Tilghman Lesher * main/channel.c, /: Merged revisions 231299 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r231299 | tilghman | 2009-11-25 16:33:02 -0600 (Wed, 25 Nov 2009) | 9 lines Merged revisions 231298 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231298 | tilghman | 2009-11-25 16:31:57 -0600 (Wed, 25 Nov 2009) | 2 lines After a frame duplication failure, unlock the channel before returning. ........ ................ 2009-11-25 15:44 +0000 [r231190] Matthew Nicholson * /, pbx/pbx_lua.c: Merged revisions 231189 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r231189 | mnicholson | 2009-11-25 09:42:48 -0600 (Wed, 25 Nov 2009) | 4 lines Load pbx_lua with global symbols to allow linking with other lua libraries. Found by Maxim Litnitskiy. ........ 2009-11-24 20:35 +0000 [r231135] Tilghman Lesher * apps/app_queue.c, /: Merged revisions 231134 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r231134 | tilghman | 2009-11-24 14:31:28 -0600 (Tue, 24 Nov 2009) | 7 lines Found a few places where queue refcounts were counted incorrectly. Also add debug statements. (closes issue #15982, closes issue #15984) Reported by: atis Patches: 20091111__issue15982.diff.txt uploaded by tilghman (license 14) Tested by: atis ........ 2009-11-24 18:54 +0000 [r231097] Jeff Peeler * /, main/features.c: Merged revisions 231095 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r231095 | jpeeler | 2009-11-24 12:50:36 -0600 (Tue, 24 Nov 2009) | 11 lines Fix erroneous hangup extension execution ast_spawn_extension behaves differently from 1.4 in that hangups and extensions that do not exist do not return an error, whereas in 1.6 it does. This is now taken into account so that the AST_FLAG_BRIDGE_HANGUP_RUN flag gets set properly. (closes issue #16106) Reported by: ajohnson Tested by: ajohnson ........ 2009-11-23 15:47 +0000 [r230883] Joshua Colp * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 230881 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r230881 | file | 2009-11-23 09:45:45 -0600 (Mon, 23 Nov 2009) | 7 lines Change fax detection in chan_sip so it behaves as one would expect. Internally the way T.38 is negotiated has changed and the option no longer reflects a behavior that is valid. It will now look for a CNG tone on received calls and if present send the call to the 'fax' extension. It is then up to the application or channel to request the switch over to T.38. ........ 2009-11-23 15:36 +0000 [r230790-230879] Kevin P. Fleming * /, channels/chan_sip.c: Merged revisions 230877 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r230877 | kpfleming | 2009-11-23 09:34:16 -0600 (Mon, 23 Nov 2009) | 9 lines Merged revisions 230839 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r230839 | kpfleming | 2009-11-23 09:09:24 -0600 (Mon, 23 Nov 2009) | 1 line Correct fix for issue #16268... the reporter's original patch was very close to correct. ........ ................ * /, channels/chan_sip.c: Merged revisions 230773 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r230773 | kpfleming | 2009-11-23 08:15:48 -0600 (Mon, 23 Nov 2009) | 12 lines Merged revisions 230772 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r230772 | kpfleming | 2009-11-23 08:13:56 -0600 (Mon, 23 Nov 2009) | 5 lines Ensure that SDP parsing does not ignore the last line of the SDP. (closes issue #16268) Reported by: sgimeno ........ ................ 2009-11-20 22:37 +0000 [r230728] David Vossel * channels/chan_iax2.c, /: Merged revisions 230726 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r230726 | dvossel | 2009-11-20 16:35:54 -0600 (Fri, 20 Nov 2009) | 7 lines fixes iax2 show cache locking error, thanks alecdavis! (closes issue #16094) Reported by: alecdavis Patches: bug16094.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis, dvossel ........ 2009-11-20 21:08 +0000 [r230630] Matthew Nicholson * /, main/features.c: Merged revisions 230628 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r230628 | mnicholson | 2009-11-20 15:01:10 -0600 (Fri, 20 Nov 2009) | 15 lines Merged revisions 230627 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r230627 | mnicholson | 2009-11-20 14:53:06 -0600 (Fri, 20 Nov 2009) | 8 lines Copy the peer CDR's userfield to the bridge CDR if it exists. This is necessary for the recordagentcalls option in chan_agent to store the recorded file name in the bridge CDR. (closes issue #14590) Reported by: msetim Patches: queue_agent_userfield.patch uploaded by Laureano (license 265) Tested by: Laureano, mnicholson ........ ................ 2009-11-20 17:32 +0000 [r230511-230586] David Vossel * /, include/asterisk/audiohook.h, main/audiohook.c: Merged revisions 230583 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r230583 | dvossel | 2009-11-20 11:26:20 -0600 (Fri, 20 Nov 2009) | 6 lines audiohook signal trigger on every status change (issue #14618) Review: https://reviewboard.asterisk.org/r/434/ ........ * apps/app_mixmonitor.c, /: Merged revisions 230509 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r230509 | dvossel | 2009-11-19 15:26:21 -0600 (Thu, 19 Nov 2009) | 17 lines Merged revisions 230508 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r230508 | dvossel | 2009-11-19 15:22:46 -0600 (Thu, 19 Nov 2009) | 10 lines fixes MixMonitor thread not exiting when StopMixMonitor is used (closes issue #16152) Reported by: AlexMS Patches: stopmixmonitor_1.4.diff uploaded by dvossel (license 671) Tested by: dvossel, AlexMS Review: https://reviewboard.asterisk.org/r/424/ ........ ................ 2009-11-30 Leif Madsen * Release Asterisk 1.6.1.11 * AST-2009-010 * SDP parser regression fix (issue #16268, issue #16238) 2009-11-18 Leif Madsen * Release Asterisk 1.6.1.10 2009-11-13 Leif Madsen * Release Asterisk 1.6.1.10-rc3 2009-11-13 15:57 +0000 [r229914] Joshua Colp * /, channels/chan_sip.c: Merged revisions 229912 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r229912 | file | 2009-11-13 09:56:16 -0600 (Fri, 13 Nov 2009) | 2 lines Fix T.38 negotiation regression introduced with the SDP parser changes. ........ 2009-11-12 23:31 +0000 [r229751] Jason Parker * channels/chan_oss.c, /: Merged revisions 229750 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r229750 | qwell | 2009-11-12 17:30:10 -0600 (Thu, 12 Nov 2009) | 1 line Fix mute toggling on OSS channels. ........ 2009-11-12 16:48 +0000 [r229672] David Vossel * funcs/func_audiohookinherit.c, /: Merged revisions 229670 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r229670 | dvossel | 2009-11-12 10:44:39 -0600 (Thu, 12 Nov 2009) | 12 lines Merged revisions 229669 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229669 | dvossel | 2009-11-12 10:41:49 -0600 (Thu, 12 Nov 2009) | 6 lines fixes merging error, datastore was being freed in the wrong function. (closes issue #16219) Reported by: aragon ........ ................ 2009-11-11 20:48 +0000 [r229569] David Ruggles * doc/externalivr.txt: Merged revisions 229568 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r229568 | diruggles | 2009-11-11 15:47:06 -0500 (Wed, 11 Nov 2009) | 9 lines Remove non-functional feature from ExternalIVR documentation Remove non-functional socket implementation of ExternalIVR from documentation (closes issue #16225) Reported by: thedavidfactor Patches: externalivr.txt.20091111.1542.patch uploaded by thedavidfactor (license 903) ........ 2009-11-11 19:54 +0000 [r229491-229501] David Brooks * main/pbx.c, /: Merged revisions 229499 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r229499 | dbrooks | 2009-11-11 13:48:18 -0600 (Wed, 11 Nov 2009) | 15 lines Merged revisions 229498 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229498 | dbrooks | 2009-11-11 13:46:19 -0600 (Wed, 11 Nov 2009) | 8 lines Solaris doesn't like NULL going to ast_log Solaris will crash if NULL is passed to ast_log. This simple patch simply uses S_OR to get around this. (closes issue #15392) Reported by: yrashk ........ ................ * /, apps/app_softhangup.c: Merged revisions 229460 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r229460 | dbrooks | 2009-11-11 12:13:56 -0600 (Wed, 11 Nov 2009) | 7 lines Flags not initialized in app_softhangup.c, causing undefined behavior Trivial patch [kobaz] to initialize an ast_flags = {0} (closes issue #16129) Reported by: kobaz ........ 2009-11-10 22:17 +0000 [r229364] Tilghman Lesher * main/pbx.c, /: Merged revisions 229361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r229361 | tilghman | 2009-11-10 16:14:22 -0600 (Tue, 10 Nov 2009) | 19 lines Merged revisions 229360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229360 | tilghman | 2009-11-10 16:09:16 -0600 (Tue, 10 Nov 2009) | 12 lines If two pattern classes start with the same digit and have the same number of characters, they will compare equal. The example given in the issue report is that of [234] and [246], which have these characteristics, yet they are clearly not equivalent. The code still uses these two characteristics, yet when the two scores compare equal, an additional check will be done to compare all characters within the class to verify equality. (closes issue #15421) Reported by: jsmith Patches: 20091109__issue15421__2.diff.txt uploaded by tilghman (license 14) Tested by: jsmith, thedavidfactor ........ ................ 2009-11-10 22:04 +0000 [r229358] David Ruggles * doc/externalivr.txt: Merged revisions 229356 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r229356 | diruggles | 2009-11-10 17:01:50 -0500 (Tue, 10 Nov 2009) | 16 lines Merged revisions 229355 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229355 | diruggles | 2009-11-10 16:45:15 -0500 (Tue, 10 Nov 2009) | 9 lines Fix ExternalIVR Documentation Remove documentation for event that doesn't function (closes issue #16220) Reported by: thedavidfactor Patches: externalivr.txt.20091110.1622.patch uploaded by thedavidfactor (license 903) ........ ................ 2009-11-10 21:31 +0000 [r229353] Tilghman Lesher * apps/app_stack.c, /: Merged revisions 229351 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r229351 | tilghman | 2009-11-10 15:22:50 -0600 (Tue, 10 Nov 2009) | 7 lines When GOSUB is invoked within an AGI, it may not exit correctly. (closes issue #16216) Reported by: atis Patches: 20091110__atis_work.diff.txt uploaded by tilghman (license 14) Tested by: atis ........ 2009-11-10 20:09 +0000 [r229284] Joshua Colp * /, codecs/codec_g726.c: Merged revisions 229282 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r229282 | file | 2009-11-10 16:06:13 -0400 (Tue, 10 Nov 2009) | 15 lines Merged revisions 229281 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229281 | file | 2009-11-10 16:03:14 -0400 (Tue, 10 Nov 2009) | 8 lines Remove broken support for direct transcoding between G.726 RFC3551 and G.726 AAL2. On some systems the translation core would actually consider g726aal2 -> g726 -> signed linear to be a quicker path then g726aal2 -> signed linear which exposed this problem. (closes issue #15504) Reported by: globalnetinc ........ ................ 2009-11-10 17:53 +0000 [r229233] David Vossel * channels/chan_iax2.c, /: Merged revisions 229168 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r229168 | dvossel | 2009-11-10 11:16:49 -0600 (Tue, 10 Nov 2009) | 15 lines Merged revisions 229167 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229167 | dvossel | 2009-11-10 11:15:57 -0600 (Tue, 10 Nov 2009) | 9 lines don't crash on log message in solaris AST-2009-006 (closes issue #16206) Reported by: bklang Tested by: bklang ........ ................ 2009-11-10 17:38 +0000 [r229230] David Ruggles * doc/externalivr.txt: Merged revisions 229228 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r229228 | diruggles | 2009-11-10 12:33:47 -0500 (Tue, 10 Nov 2009) | 18 lines Merged revisions 229191 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229191 | diruggles | 2009-11-10 12:23:59 -0500 (Tue, 10 Nov 2009) | 11 lines Document ExternalIVR event tag collision ExternalIVR uses the D tag for two different event types. This documents that behavior and how to differentiate between the two cases. Also includes a minor spelling fix and clarification (closes issue #16211) Reported by: thedavidfactor Patches: externalivr.txt.20091109.1507.patch uploaded by thedavidfactor (license 903) ........ ................ 2009-11-10 15:38 +0000 [r229099] Matthew Nicholson * channels/chan_sip.c: Reverted revision 202008. (closes issue #16175) Reported by: paul-tg 2009-11-10 15:36 +0000 [r229095-229098] David Vossel * res/res_config_pgsql.c: reverting changes made by r229095 as they are not applicable to 1.6.1 * res/res_config_pgsql.c, /: Merged revisions 229093 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r229093 | dvossel | 2009-11-10 09:27:45 -0600 (Tue, 10 Nov 2009) | 11 lines fixes pgsql double free of threadstorage A thread storage variable was being freed incorrectly, which resulted in a double free if two queries were made in the same thread. (closes issue #16011) Reported by: cristiandimache Patches: issue16011.diff uploaded by dvossel (license 671) ........ 2009-11-10 11:22 +0000 [r229067] Gavin Henry * contrib/scripts/asterisk.ldap-schema, /: Merged revisions 229050 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r229050 | ghenry | 2009-11-10 11:16:10 +0000 (Tue, 10 Nov 2009) | 20 lines Schema file additions * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses to allow standalone dialplan, account and mailbox entries (STRUCTURAL) * Added new Fields: - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir, - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap, - AstAccountVideoSupport, AstAccountIgnoreSDPVersion * Removed redundant IPaddr (there's already IPAddress) - Gives more configuration Flags for SIP-Users available (tested) - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses without extensibleObject (which really should be the last resort); gives also additional possibilities for LDAP-filter (closes issue #15874) Reported by: Medozas Patches: asterisk.ldap-schema.patch uploaded by Medozas (license 41) Tested by: Medozas, suretec ........ 2009-11-09 22:52 +0000 [r229016] Terry Wilson * channels/chan_local.c, /: Merged revisions 229015 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r229015 | twilson | 2009-11-09 16:50:22 -0600 (Mon, 09 Nov 2009) | 8 lines Don't crash when bridge->tech_pvt == NULL This is a similar solution to what is in place for chan_agent (closes issue #16003) Reported by: atis Tested by: twilson ........ 2009-11-09 22:18 +0000 [r229014] David Vossel * channels/chan_sip.c: fixes segfault when transferring a queue caller In sip_hangup we attempted to lock p->owner after we set it to NULL. Thanks to fhackenberger for reporting the issue and submitting a patch. (closes issue 0015848) Reported by: fhackenberger Patches: digium_bug_0015848 uploaded by fhackenberger (license 592) Tested by: fhackenberger, lmadsen, TomS, shin-shoryuken, dvossel 2009-11-09 Leif Madsen * Release Asterisk 1.6.1.10-rc2 2009-11-09 15:39 +0000 [r228899] Leif Madsen * main/channel.c: Merged revisions 228897 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228897 | lmadsen | 2009-11-09 09:38:38 -0600 (Mon, 09 Nov 2009) | 14 lines Merged revisions 228896 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228896 | lmadsen | 2009-11-09 09:37:43 -0600 (Mon, 09 Nov 2009) | 6 lines Update WARNING message. Update a WARNING message to give a suggested fix when encountered. (closes issue #16198) Reported by: atis Tested by: atis ........ ................ 2009-11-09 14:54 +0000 [r228860] Matthew Nicholson * /, include/asterisk/lock.h: Merged revisions 228858 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228858 | mnicholson | 2009-11-09 08:37:07 -0600 (Mon, 09 Nov 2009) | 15 lines Merged revisions 228827 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228827 | mnicholson | 2009-11-09 08:16:03 -0600 (Mon, 09 Nov 2009) | 8 lines Perform limited bounds checking when destroying ast_mutex_t structures to make sure we don't try to use negative indices. (closes issue #15588) Reported by: zerohalo Patches: 20090820__issue15588.diff.txt uploaded by tilghman (license 14) Tested by: zerohalo ........ ................ 2009-11-06 22:37 +0000 [r228695] David Vossel * main/channel.c, /: Merged revisions 228693 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228693 | dvossel | 2009-11-06 16:35:44 -0600 (Fri, 06 Nov 2009) | 16 lines Merged revisions 228692 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228692 | dvossel | 2009-11-06 16:33:27 -0600 (Fri, 06 Nov 2009) | 9 lines fixes audiohook write crash occuring in chan_spy whisper mode. After writing to the audiohook list in ast_write(), frames were being freed incorrectly. Under certain conditions this resulted in a double free crash. (closes issue #16133) Reported by: wetwired ........ ................ 2009-11-06 20:37 +0000 [r228650] Matthew Nicholson * funcs/func_base64.c, /, main/utils.c: Merged revisions 228620 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228620 | mnicholson | 2009-11-06 13:47:11 -0600 (Fri, 06 Nov 2009) | 15 lines Merged revisions 228378 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228378 | mnicholson | 2009-11-06 10:26:59 -0600 (Fri, 06 Nov 2009) | 8 lines Properly handle '=' while decoding base64 messages and null terminate strings returned from BASE64_DECODE. (closes issue #15271) Reported by: chappell Patches: base64_fix.patch uploaded by chappell (license 8) Tested by: kobaz ........ ................ 2009-11-06 18:41 +0000 [r228550] Joshua Colp * /, channels/chan_sip.c: Merged revisions 228548 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228548 | file | 2009-11-06 14:37:59 -0400 (Fri, 06 Nov 2009) | 11 lines Merged revisions 228547 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228547 | file | 2009-11-06 14:32:58 -0400 (Fri, 06 Nov 2009) | 4 lines Don't overwrite caller ID name on a trunk with the configured fullname when using users.conf (issue ABE-1989) ........ ................ 2009-11-06 Leif Madsen * Release Asterisk 1.6.1.10-rc1 2009-11-06 17:53 +0000 [r228502] Joshua Colp * /, doc/tex/localchannel.tex: Merged revisions 228499 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r228499 | file | 2009-11-06 13:52:00 -0400 (Fri, 06 Nov 2009) | 2 lines Fix the localchannel.tex file. ........ 2009-11-06 17:24 +0000 [r228422-228451] David Vossel * /, codecs/codec_ilbc.c: Merged revisions 228441 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r228441 | dvossel | 2009-11-06 11:22:31 -0600 (Fri, 06 Nov 2009) | 3 lines Fixes merging issue from 1.4, frame data is held in data.ptr in trunk ........ * /, codecs/codec_ilbc.c: Merged revisions 228420 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228420 | dvossel | 2009-11-06 11:09:01 -0600 (Fri, 06 Nov 2009) | 19 lines Merged revisions 228418 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228418 | dvossel | 2009-11-06 11:07:13 -0600 (Fri, 06 Nov 2009) | 13 lines fixes segfault in iLBC For reasons not yet known, it appears possible for an ast_frame to have a datalen greater than zero while the actual data is NULL during Packet Loss Concealment. Most codecs don't support PLC so this doesn't affect them. This patch catches the malformed frame and prevents the crash from occuring. Additional efforts to determine why it is possible for a frame to look like this are still being investigated. (issue #16979) ........ ................ 2009-11-06 16:44 +0000 [r228412] Joshua Colp * /, main/abstract_jb.c: Merged revisions 228410 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228410 | file | 2009-11-06 12:42:23 -0400 (Fri, 06 Nov 2009) | 14 lines Merged revisions 228409 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228409 | file | 2009-11-06 12:41:20 -0400 (Fri, 06 Nov 2009) | 7 lines Fix a bug caused by a partially invalid frame (from the jitterbuffer) passing through the Asterisk core. (closes issue #15560) Reported by: jvandal (closes issue #15709) Reported by: covici ........ ................ 2009-11-06 15:44 +0000 [r228267-228341] David Vossel * /, main/astfd.c: Merged revisions 228339 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228339 | dvossel | 2009-11-06 09:42:46 -0600 (Fri, 06 Nov 2009) | 12 lines Merged revisions 228338 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228338 | dvossel | 2009-11-06 09:41:41 -0600 (Fri, 06 Nov 2009) | 5 lines fixes crash in astfd.c (closes issue #15981) Reported by: slavon ........ ................ * funcs/func_audiohookinherit.c, /: Merged revisions 228268 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r228268 | dvossel | 2009-11-06 09:04:24 -0600 (Fri, 06 Nov 2009) | 9 lines fixes memory leak in func_audiohookinherit.c (closes issue #15394) Reported by: boroda Patches: bug15394_memoryleak_diff2.txt uploaded by dbrooks (license 790) Tested by: dbrooks, boroda ........ * /, channels/chan_sip.c: Merged revisions 227238 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r227238 | dvossel | 2009-11-03 11:12:52 -0600 (Tue, 03 Nov 2009) | 5 lines user.conf entries in SIP were not having their peer type set. (closes issue #16120) Reported by: jsmith ........ 2009-11-05 22:13 +0000 [r228193-228197] Tilghman Lesher * /, apps/app_meetme.c: Merged revisions 228196 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r228196 | tilghman | 2009-11-05 16:12:45 -0600 (Thu, 05 Nov 2009) | 2 lines Yet another error message in the dialplan (thanks, rmudgett/russellb) ........ * /, apps/app_meetme.c: Merged revisions 228191 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r228191 | tilghman | 2009-11-05 15:24:21 -0600 (Thu, 05 Nov 2009) | 7 lines MEETME_INFO should not return a literal error message to the dialplan. (closes issue #15450) Reported by: JimVanM Patches: meetmeinfopatch.diff.txt uploaded by dbrooks (license 790) Tested by: JimVanM ........ 2009-11-05 21:24 +0000 [r228192] Jeff Peeler * apps/app_chanspy.c, /: Merged revisions 228189 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r228189 | jpeeler | 2009-11-05 15:23:06 -0600 (Thu, 05 Nov 2009) | 11 lines Fix the fix for chanspy option o In 224178, I assumed the uploaded patch was correct as it had received positive feedback. The flags were being checked in the incorrect location. Upon testing the fix this time it was also found that the flags from the dialplan weren't being copied to the chanspy_translation_helper. (closes issue #16167) Reported by: marhbere ........ 2009-11-05 19:41 +0000 [r228147] David Brooks * channels/chan_misdn.c, /: Merged revisions 228145 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228145 | dbrooks | 2009-11-05 13:34:50 -0600 (Thu, 05 Nov 2009) | 16 lines Merged revisions 228078 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228078 | dbrooks | 2009-11-05 12:59:41 -0600 (Thu, 05 Nov 2009) | 9 lines chan_misdn Asterisk 1.4.27-rc2 crash Crash related to chan_misdn connection. Patch submitted by gknispel_proformatique, tested by francesco_r. "I have many crash since i have upgraded to Asterisk 1.4.27-rc2. Attached a full bt." This patch zeros out an ast_frame. (closes issue #16041) Reported by: francesco_r ........ ................ 2009-11-05 19:19 +0000 [r228090] Jason Parker * channels/chan_vpb.cc, /: Merged revisions 228080 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r228080 | qwell | 2009-11-05 13:16:29 -0600 (Thu, 05 Nov 2009) | 15 lines Merged revisions 228079 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228079 | qwell | 2009-11-05 13:14:25 -0600 (Thu, 05 Nov 2009) | 8 lines Fix crash on VPB exception when no hardware is present. (closes issue #14970) Reported by: tzafrir Patches: vpb_exception.diff uploaded by tzafrir (license 46) Tested by: markwaters ........ ................ 2009-11-05 17:10 +0000 [r228016] Tilghman Lesher * /, apps/app_externalivr.c: Merged revisions 228015 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r228015 | tilghman | 2009-11-05 11:08:02 -0600 (Thu, 05 Nov 2009) | 4 lines Don't crash if no arguments are passed. (closes issue #16119) Reported by: thedavidfactor ........ 2009-11-04 23:56 +0000 [r227948] Jeff Peeler * res/res_monitor.c, /: Merged revisions 227945 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r227945 | jpeeler | 2009-11-04 17:50:59 -0600 (Wed, 04 Nov 2009) | 21 lines Merged revisions 227944 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227944 | jpeeler | 2009-11-04 17:47:08 -0600 (Wed, 04 Nov 2009) | 14 lines Fix incorrect filename comparsion after monitor file change The logic to detect if a requested file is indeed a different file from the current file was incorrect. The main issue being confusion of the use of filename_base which was previously set without pathing information and then compared to another full path. Robust file comparison logic has been added to properly check if two files are the same even if symlinks are used. (closes issue #15313) Reported by: caspy Patches: 20091103__issue15313__1.4.diff.txt uploaded by jpeeler (license 325) but mostly tilghman's work ........ ................ 2009-11-04 21:15 +0000 [r227761-227832] Matthew Nicholson * apps/app_dial.c, /: Merged revisions 227829 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r227829 | mnicholson | 2009-11-04 15:03:33 -0600 (Wed, 04 Nov 2009) | 17 lines Merged revisions 227827 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227827 | mnicholson | 2009-11-04 14:52:27 -0600 (Wed, 04 Nov 2009) | 10 lines This patch modifies the Dial application to monitor the calling channel for hangups while playing back announcements. (closes issue #16005) Reported by: falves11 Patches: dial-announce-hangup-fix1.diff uploaded by mnicholson (license 96) Tested by: mnicholson, falves11 Review: https://reviewboard.asterisk.org/r/407/ ........ ................ * channels/chan_sip.c: Modify the SDP parsing code to parse session and media level items separately. With the new code, media level proprieties should no longer be confused with session level proprieties. This change also reorganizes some of the SDP parsing code which should make it easier to manage in the future. (closes issue #14994) Reported by: frawd 2009-11-04 19:27 +0000 [r227723-227745] Joshua Colp * /, static-http/prototype.js: Merged revisions 227739 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r227739 | file | 2009-11-04 15:26:19 -0400 (Wed, 04 Nov 2009) | 12 lines Merged revisions 227735 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227735 | file | 2009-11-04 15:25:37 -0400 (Wed, 04 Nov 2009) | 5 lines Fix a security issue where it may be possible for someone to execute a cross-site AJAX request exploit. (AST-2009-009) ........ ................ * /, channels/chan_sip.c: Merged revisions 227712 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r227712 | file | 2009-11-04 15:20:46 -0400 (Wed, 04 Nov 2009) | 12 lines Merged revisions 227700 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227700 | file | 2009-11-04 15:17:39 -0400 (Wed, 04 Nov 2009) | 5 lines Fix a security issue where sending a REGISTER with a differing username in the From URI and Authorization header would reveal whether it was valid or not. (AST-2009-008) ........ ................ 2009-11-03 20:01 +0000 [r227374] Jason Parker * Makefile, /, main/Makefile: Merged revisions 227372 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r227372 | qwell | 2009-11-03 13:59:46 -0600 (Tue, 03 Nov 2009) | 9 lines Fix some build issues on Solaris. (closes issue #14517) (SWP-109) Reported by: asgaroth Patches: bug_14517.diff uploaded by snuffy (license 35) Tested by: asgaroth, snuffy, dougm, qwell ........ 2009-11-03 19:49 +0000 [r227363-227370] Leif Madsen * apps/app_controlplayback.c, /: Merged revisions 227368 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r227368 | lmadsen | 2009-11-03 13:48:53 -0600 (Tue, 03 Nov 2009) | 8 lines Change warning message to debug message. app_controlplayback outputs a warning, when in fact it is normal. (closes issue #16071) Reported by: atis Patches: controlplayback_warning.patch uploaded by atis (license 242) ........ * configs/extensions.conf.sample, /: Merged revisions 227361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r227361 | lmadsen | 2009-11-03 13:25:18 -0600 (Tue, 03 Nov 2009) | 11 lines Additional fixes to the extensions.conf.sample file. Update the extensions.conf.sample [stdexten] context so that we use the variable instead of requiring it to be passed explicitly. Also updated uses of the [stdexten] context throughout. (closes issue #15858) Reported by: pprindeville Patches: stdexten-context-update.txt uploaded by lmadsen (license 10) Tested by: pprindeville ........ 2009-11-03 18:11 +0000 [r227279] Richard Mudgett * channels/chan_dahdi.c: Merged revisions 227275 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227275 | rmudgett | 2009-11-03 11:55:47 -0600 (Tue, 03 Nov 2009) | 4 lines Make sure the outgoing flag is cleared if a new channel fails to get created for outgoing calls. This is the relevant portion of asterisk/trunk -r226648 ........ 2009-11-03 15:38 +0000 [r227169] Joshua Colp * /, channels/chan_sip.c: Merged revisions 227167 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r227167 | file | 2009-11-03 11:37:08 -0400 (Tue, 03 Nov 2009) | 12 lines Merged revisions 227166 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227166 | file | 2009-11-03 11:36:16 -0400 (Tue, 03 Nov 2009) | 5 lines Fix a bug where an RPID header could be generated with a blank username in the URI. (closes issue #15909) Reported by: kobaz ........ ................ 2009-11-03 15:24 +0000 [r227164] Leif Madsen * configs/extensions.conf.sample, /: Merged revisions 227162 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r227162 | lmadsen | 2009-11-03 09:19:47 -0600 (Tue, 03 Nov 2009) | 7 lines Update extensions.conf.sample file to fix incorrect extensions. (closes issue #15857) Reported by: pprindeville Patches: stdexten.patch#2 uploaded by pprindeville (license 347) Tested by: pprindeville ........ 2009-11-03 13:32 +0000 [r227155] Olle Johansson * /, channels/chan_sip.c: Merged revisions 227091 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r227091 | oej | 2009-11-03 12:11:15 +0100 (Tis, 03 Nov 2009) | 15 lines Merged revisions 227088 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227088 | oej | 2009-11-03 11:29:59 +0100 (Tis, 03 Nov 2009) | 7 lines Use proper response code when violating Contact ACL's. https://reviewboard.asterisk.org/r/415/ Thanks kpfleming for a quick review. (EDVX-003) ........ ................ 2009-11-02 21:05 +0000 [r226977] David Brooks * channels/chan_sip.c: SIP channel name uniqueness SIP channel names were supposed to be unique by way of a name suffix derived from the pointer to the channel's private data. Uniqueness was preserved on 32-bit systems, but not on 64-bit systems. This patch, as suggested by kpfleming, replaces this suffix with a simple incremented unsigned int. (closes issue #15152) Reported by: palbrecht Review: https://reviewboard.asterisk.org/r/420/ 2009-11-02 18:11 +0000 [r226892] Joshua Colp * apps/app_dial.c, /: Merged revisions 226890 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r226890 | file | 2009-11-02 14:08:54 -0400 (Mon, 02 Nov 2009) | 18 lines Merged revisions 226889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226889 | file | 2009-11-02 14:08:11 -0400 (Mon, 02 Nov 2009) | 11 lines Fix a bug where the recorded privacy introduction file would not get removed if the caller hung up while the called party had not yet answered. This was fixed by introducing an argument to the 'n' option which, when enabled, removes the introduction file under all scenarios. This was done to preserve the behavior that has existed for quite some time. (closes issue #14674) Reported by: ulogic Patches: bug14674.patch uploaded by jpeeler (license 325) ........ ................ 2009-11-02 17:17 +0000 [r226814] Tilghman Lesher * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 226812 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r226812 | tilghman | 2009-11-02 11:15:31 -0600 (Mon, 02 Nov 2009) | 15 lines Merged revisions 226811 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226811 | tilghman | 2009-11-02 11:14:20 -0600 (Mon, 02 Nov 2009) | 8 lines Don't allow two separate instances of safe_asterisk when restarting from the init script. (closes issue #14562) Reported by: davidw Patches: Initially 20091022__issue14562.diff.txt uploaded by tilghman (license 14) Modified to 20091030__Issue14562_diff.txt uploaded by davidw (license 780) Tested by: davidw ........ ................ 2009-10-29 18:15 +0000 [r226534] Joshua Colp * channels/chan_local.c, /, doc/tex/localchannel.tex: Merged revisions 226532 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r226532 | file | 2009-10-29 15:13:42 -0300 (Thu, 29 Oct 2009) | 13 lines Merged revisions 226531 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226531 | file | 2009-10-29 15:11:26 -0300 (Thu, 29 Oct 2009) | 6 lines Add an option to enabling passing music on hold start and stop requests through instead of acting on them in chan_local. (closes issue #14709) Reported by: dimas ........ ................ 2009-10-28 20:15 +0000 [r226380-226386] Leif Madsen * configs/sip.conf.sample: Merged revisions 226384 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r226384 | lmadsen | 2009-10-28 15:11:07 -0500 (Wed, 28 Oct 2009) | 17 lines Merged revisions 226382 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226382 | lmadsen | 2009-10-28 15:06:13 -0500 (Wed, 28 Oct 2009) | 9 lines Update documentation in sip.conf.sample. Update the documentation in sip.conf.sample in order to make it more clear that directmedia/canreinvite do not cause Asterisk to ignore reINVITEs. It is only used to stop Asterisk from generating a reINVITE, but does not stop it from accepting them if necessary. (closes issue #15644) Reported by: lmadsen ........ ................ * /, doc/tex/channelvariables.tex: Merged revisions 226378 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r226378 | lmadsen | 2009-10-28 14:50:00 -0500 (Wed, 28 Oct 2009) | 15 lines Merged revisions 226377 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226377 | lmadsen | 2009-10-28 14:48:29 -0500 (Wed, 28 Oct 2009) | 7 lines Update CALLINGSUBADDR channel variable documentation. (closes issue #15734) Reported by: alecdavis Patches: channelvariables.tex.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis ........ ................ 2009-10-28 18:05 +0000 [r226169-226307] Tilghman Lesher * /, include/asterisk/linkedlists.h: Merged revisions 226305 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r226305 | tilghman | 2009-10-28 13:04:05 -0500 (Wed, 28 Oct 2009) | 9 lines Merged revisions 226304 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226304 | tilghman | 2009-10-28 13:02:25 -0500 (Wed, 28 Oct 2009) | 2 lines Fix documentation (pointed out by TheDavidFactor on #-dev) ........ ................ * main/manager.c, /: Merged revisions 226159 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r226159 | tilghman | 2009-10-27 15:22:07 -0500 (Tue, 27 Oct 2009) | 14 lines Merged revisions 226138 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226138 | tilghman | 2009-10-27 15:16:49 -0500 (Tue, 27 Oct 2009) | 7 lines Manager output is not always NULL-terminated, so force a NULL at the end of the filestream. (closes issue #15495) Reported by: pdf Patches: 20090916__issue15495.diff.txt uploaded by tilghman (license 14) Tested by: pdf ........ ................ 2009-10-27 17:04 +0000 [r226100] Terry Wilson * /, res/res_http_post.c: Merged revisions 226099 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r226099 | twilson | 2009-10-27 11:48:54 -0500 (Tue, 27 Oct 2009) | 2 lines Don't prepend the URI prefix to the post directory ........ 2009-10-26 23:48 +0000 [r226053] Tzafrir Cohen * /, configure, configure.ac: detect ARM Linux EABI OSARCH as linux-gnu instead of linux-gnueabi * Set OSARCH to linux-gnu even if host_os is linux-gnueabi * When checking if we are Linux, check OSARCH rather than host_os The newer ARM ABI ("EABI") shows the OS name 'linux-gnueabi' rather than 'linux-gnu' . This patch sets OSARCH to be 'linux-gnu' even in such a case. OSARCH is tested for the value of 'linux-gnu' in one or two places in the tree. This patch also fixes the check libcap to check for $OSARCH rather than $host_os . See also: http://wiki.debian.org/ArmEabiPort Merged revisions 225957 via svnmerge from http://svn.digium.com/svn/asterisk/branches/1.4 Merged revisions 226018 via svnmerge from http://svn.digium.com/svn/asterisk/trunk 2009-10-26 19:41 +0000 [r225913] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 225912 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r225912 | jpeeler | 2009-10-26 14:40:26 -0500 (Mon, 26 Oct 2009) | 12 lines ACL check not present for verifying SIP INVITEs The ACL check in check_peer_ok was missing and has now been restored. The missing check allowed for calls to be made on prohibited networks where an ACL was defined in sip.conf and the allowguest option was set to off. See the AST security advisory below for more information. Merge code associated with AST-2009-007. (closes issue #16091) Reported by: thom4fun ........ 2009-10-26 15:51 +0000 [r225870] Kevin P. Fleming * apps/app_fax.c: Backport audio handling loop fixes from trunk version of app_fax. This backport resolves some issues handling audio frames during FAX processing, and ensures that the FAX application doesn't accidentally get notified of a T.38 switchover at the end of a successful FAX. (issue #16127) 2009-10-23 14:50 +0000 [r225652] David Vossel * /, channels/chan_sip.c: Merged revisions 225650 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r225650 | dvossel | 2009-10-23 09:41:50 -0500 (Fri, 23 Oct 2009) | 3 lines Fixes an iterator memory leak and uninitialized memory ........ 2009-10-23 14:07 +0000 [r225584] Kevin P. Fleming * Makefile, /: Merged revisions 225582 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r225582 | kpfleming | 2009-10-23 09:02:42 -0500 (Fri, 23 Oct 2009) | 17 lines Merged revisions 225581 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225581 | kpfleming | 2009-10-23 09:00:01 -0500 (Fri, 23 Oct 2009) | 10 lines Don't force menuselect.makeopts to be rebuilt on every build. For some reason the menuselect.makeopts file was listed as PHONY in the Makefile, resulting in 'make' needing to rebuild it for every build. This then resulted in the embedded module rules being rebuilt on every build, which can be slow and is unnecessary. This patch fixes the problem by properly allowing 'make' to know when the menuselect.makeopts file needs to be rebuilt (defining the proper dependencies). ........ ................ 2009-10-22 22:07 +0000 [r225490] David Vossel * main/tcptls.c, /, channels/chan_sip.c, apps/app_externalivr.c, include/asterisk/tcptls.h: Merged revisions 225445 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r225445 | dvossel | 2009-10-22 14:55:51 -0500 (Thu, 22 Oct 2009) | 50 lines SIP TCP/TLS: move client connection setup/write into tcp helper thread, various related locking/memory fixes. What this patch fixes 1.Moves sip TCP/TLS connection setup into the TCP helper thread: Connection setup takes awhile and before this it was being done while holding the monitor lock. 2.Moves TCP/TLS writing to the TCP helper thread: Through the use of a packet queue and an alert pipe, the TCP helper thread can now be woken up to write data as well as read data. 3.Locking error: sip_xmit returned an XMIT_ERROR without giving up the tcptls_session lock. This lock has been completely removed from sip_xmit and placed in the new sip_tcptls_write() function. 4.Memory leak: When creating a tcptls_client the tls_cfg was alloced but never freed unless the tcptls_session failed to start. Now the session_args for a sip client are an ao2 object which frees the tls_cfg on destruction. 5.Pointer to stack variable: During sip_prepare_socket the creation of a client's ast_tcptls_session_args was done on the stack and stored as a pointer in the newly created tcptls_session. Depending on the events that followed, there was a slight possibility that pointer could have been accessed after the stack returned. Given the new changes, it is always accessed after the stack returns which is why I found it. Notable code changes 1.I broke tcptls.c's ast_tcptls_client_start() function into two functions. One for creating and allocating the new tcptls_session, and a separate one for starting and handling the new connection. This allowed me to create the tcptls_session, launch the helper thread, and then establish the connection within the helper thread. 2.Writes to a tcptls_session are now done within the helper thread. This is done by using an alert pipe to wake up the thread if new data needs to be sent. The thread's sip_threadinfo object contains the alert pipe as well as the packet queue. 3.Since the threadinfo object contains the alert pipe, it must now be accessed outside of the helper thread for every write (queuing of a packet). For easy lookup, I moved the threadinfo objects from a linked list to an ao2_container. (closes issue #13136) Reported by: pabelanger Tested by: dvossel, whys (closes issue #15894) Reported by: dvossel Tested by: dvossel Review: https://reviewboard.asterisk.org/r/380/ ........ 2009-10-22 21:54 +0000 [r225487] Leif Madsen * doc/valgrind.txt, contrib/valgrind.supp (added): Merged revisions 225485 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r225485 | lmadsen | 2009-10-22 16:52:30 -0500 (Thu, 22 Oct 2009) | 19 lines Merged revisions 225484 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225484 | lmadsen | 2009-10-22 16:51:52 -0500 (Thu, 22 Oct 2009) | 11 lines Clean valgrind output by suppressing false errors. Update valgrind.txt documentation and add valgrind.supp file in order to allow those who are creating valgrind output to have less false errors in the logfile. (closes issue #16007) Reported by: atis Patches: valgrind.txt.diff uploaded by atis (license 242) asterisk2.supp uploaded by atis (license 242) Tested by: atis, amorsen ........ ................ 2009-10-22 17:14 +0000 [r225362] Tilghman Lesher * main/pbx.c, /, apps/app_meetme.c, include/asterisk/channel.h: Merged revisions 225360 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r225360 | tilghman | 2009-10-22 12:11:23 -0500 (Thu, 22 Oct 2009) | 11 lines Merged revisions 225105 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009) | 4 lines Fix documentation for ast_softhangup() and correct the misuse thereof. (closes issue #16103) Reported by: majorbloodnok ........ ................ 2009-10-21 22:02 +0000 [r225062-225309] David Vossel * channels/chan_iax2.c, /: Merged revisions 225307 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r225307 | dvossel | 2009-10-21 16:58:46 -0500 (Wed, 21 Oct 2009) | 20 lines Merged revisions 225243 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225243 | dvossel | 2009-10-21 15:58:08 -0500 (Wed, 21 Oct 2009) | 13 lines IAX2: VNAK loop caused by signaling frames with no destination call number It is possible for the PBX thread to queue up signaling frames before a destination call number is received. This can result in signaling frames being sent out with no destination call number. Since recent versions of Asterisk require accurate destination callnumbers for all Full Frames, this can cause a VNAK loop to occur. To resolve this no signaling frames are sent until a destination callnumber is received, and destination call numbers are now only required for iax_pvt matching when the frame is an ACK. Review: https://reviewboard.asterisk.org/r/413/ ........ ................ * channels/chan_iax2.c, configs/iax.conf.sample, /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 225033 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r225033 | dvossel | 2009-10-21 09:39:10 -0500 (Wed, 21 Oct 2009) | 27 lines Merged revisions 225032 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009) | 20 lines IAX/SIP shrinkcallerid option The shrinking of caller id removes '(', ' ', ')', non-trailing '.', and '-' from the string. This means values such as 555.5555 and test-test result in 555555 and testtest. There are instances, such as Skype integration, where a specific value is passed via caller id that must be preserved unmodified. This patch makes the shrinking of caller id optional in chan_sip and chan_iax in order to support such cases. By default this option is on to preserve previous expected behavior. (closes issue #15940) Reported by: dimas Patches: v2-15940.patch uploaded by dimas (license 88) 15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671) Tested by: dvossel Review: https://reviewboard.asterisk.org/r/408/ ........ ................ 2009-10-21 03:17 +0000 [r224935] Russell Bryant * include/asterisk/frame.h, include/asterisk/translate.h, main/dsp.c, main/frame.c, /, main/translate.c, include/asterisk/dsp.h, codecs/codec_dahdi.c: Merged revisions 224932 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r224932 | russell | 2009-10-20 22:09:04 -0500 (Tue, 20 Oct 2009) | 12 lines Merged revisions 224931 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224931 | russell | 2009-10-20 21:59:54 -0500 (Tue, 20 Oct 2009) | 5 lines Isolate frames returned from a DSP instance or codec translator. The reasoning for these changes are the same as what I wrote in the commit message for rev 222878. ........ ................ 2009-10-20 22:11 +0000 [r224858] Tilghman Lesher * funcs/func_speex.c, /, main/audiohook.c: Merged revisions 224856 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r224856 | tilghman | 2009-10-20 17:09:07 -0500 (Tue, 20 Oct 2009) | 12 lines Merged revisions 224855 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224855 | tilghman | 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009) | 5 lines Pay attention to the return value of the manipulate function. While this looks like an optimization, it prevents a crash from occurring when used with certain audiohook callbacks (diagnosed with SVN trunk, backported to 1.4 to keep the source consistent across versions). ........ ................ 2009-10-20 17:49 +0000 [r224776] Joshua Colp * /, main/features.c: Merged revisions 224774 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r224774 | file | 2009-10-20 14:47:34 -0300 (Tue, 20 Oct 2009) | 12 lines Merged revisions 224773 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224773 | file | 2009-10-20 14:46:37 -0300 (Tue, 20 Oct 2009) | 5 lines Add support for relaying early media in the features attended transfer option. (closes issue #14828) Reported by: licedey ........ ................ 2009-10-19 23:56 +0000 [r224673] Kevin P. Fleming * main/rtp.c: Merged revisions 224671 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r224671 | kpfleming | 2009-10-19 18:47:39 -0500 (Mon, 19 Oct 2009) | 14 lines Merged revisions 224670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224670 | kpfleming | 2009-10-19 18:44:07 -0500 (Mon, 19 Oct 2009) | 7 lines Correct timestamp calculations when RTP sample rates over 8kHz are used. While testing some endpoints that support 16kHz and 32kHz sample rates, some log messages were generated due to calc_rxstamp() computing timestamps in a way that produced odd results, so this patch sanitizes the result of the computations. ........ ................ 2009-10-19 19:51 +0000 [r224570] Joshua Colp * apps/app_dial.c, /: Merged revisions 224567 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r224567 | file | 2009-10-19 16:49:09 -0300 (Mon, 19 Oct 2009) | 12 lines Merged revisions 224565 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224565 | file | 2009-10-19 16:47:50 -0300 (Mon, 19 Oct 2009) | 5 lines Do not attempt early media bridging (ie: direct RTP setup) if options are enabled that should prevent it. (closes issue #14763) Reported by: cupotka ........ ................ 2009-10-17 02:02 +0000 [r224333-224336] Jeff Peeler * channels/chan_dahdi.c: fix typo, sorry * channels/chan_dahdi.c, /: Merged revisions 224331 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r224331 | jpeeler | 2009-10-16 20:36:08 -0500 (Fri, 16 Oct 2009) | 20 lines Merged revisions 224330 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224330 | jpeeler | 2009-10-16 20:32:47 -0500 (Fri, 16 Oct 2009) | 13 lines Fix stale caller id data from being reported in AMI NewChannel event The problem here is that chan_dahdi is designed in such a way to set certain values in the dahdi_pvt only once. One of those such values is the configured caller id data in chan_dahdi.conf. For PRI, the configured caller id data could be overwritten during a call. Instead of saving the data and restoring, it was decided that for all non-analog channels it was simply best to not set the configured caller id in the first place and also clear it at the end of the call. (closes issue #15883) Reported by: jsmith ........ ................ 2009-10-16 20:53 +0000 [r224263] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 224261 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r224261 | rmudgett | 2009-10-16 15:40:57 -0500 (Fri, 16 Oct 2009) | 25 lines Merged revisions 224260 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224260 | rmudgett | 2009-10-16 15:25:23 -0500 (Fri, 16 Oct 2009) | 18 lines Never released PRI channels when using Busy() or Congestion() dialplan apps. When the Busy() or Congestion() application is used towards ISDN (an ISDN progress is sent), the responding ISDN Disconnect or Release may contain the ISDN cause user busy or one of the congestion causes. In chan_dahdi.c these causes will only set the needbusy or needcongestion flags and not activate the softhangup procedure. Unfortunately only the latter can interrupt the endless wait loop of Busy()/Congestion(). Result: PRI channels staying in state busy for the rest of asterisk life or until the other end times out and forces the call to clear. (in issue 0014292) Reported by: tomaso Patches: disc_rel_userbusy.patch uploaded by tomaso (license 564) (This patch is unrelated to the issue.) ........ ................ 2009-10-15 15:58 +0000 [r224180] Jeff Peeler * apps/app_chanspy.c, /: Merged revisions 224178 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r224178 | jpeeler | 2009-10-15 10:57:14 -0500 (Thu, 15 Oct 2009) | 11 lines Readd removed ability to allow listening to one side of the call in app_chanspy (Option o) (closes issue #15675) Reported by: john8675309 Patches: issue15675patchtrunk.txt uploaded by dbrooks (license 790) Tested by: jgutierrez on users list: http://lists.digium.com/pipermail/asterisk-users/2009-October/239155.html ........ 2009-10-12 23:55 +0000 [r223834] Jeff Peeler * apps/app_dial.c, /: Merged revisions 223832 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r223832 | jpeeler | 2009-10-12 18:48:09 -0500 (Mon, 12 Oct 2009) | 15 lines Merged revisions 223804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223804 | jpeeler | 2009-10-12 18:12:50 -0500 (Mon, 12 Oct 2009) | 8 lines Ensure ringing continues for branched calls after progress is received While waiting for an answer, don't send progress for branched calls for which ringing was sent. (closes issue #15028) Reported by: fnordian ........ ................ 2009-10-12 21:03 +0000 [r223758] David Vossel * configs/iax.conf.sample, /: Merged revisions 223756 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r223756 | dvossel | 2009-10-12 15:58:27 -0500 (Mon, 12 Oct 2009) | 5 lines Clarifies trunkmaxsize, trunkfreq, and trunkmtu iax2 options SWP-151 ........ 2009-10-12 14:32 +0000 [r223654] Kevin P. Fleming * /, channels/chan_sip.c, apps/app_fax.c: Merged revisions 223652 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r223652 | kpfleming | 2009-10-12 09:25:29 -0500 (Mon, 12 Oct 2009) | 13 lines Remove automatic switching from T.38 to voice mode in chan_sip. chan_sip has some code to automatically switch from T.38 mode to voice mode when a voice frame is written to the channel while it is in T.38 mode; this was intended to handle the situation when a FAX transmission has ended and the channel is not yet hung up, but is causing problems at the beginning of FAX sessions as well when there are still voice frames 'in flight' at the time the T.38 negotiation completes. This patch removes the automatic switchover, and changes app_fax to explicitly switch off T.38 mode when the FAX transmission process ends. (closes issue #16025) Reported by: jamicque ........ 2009-10-11 17:31 +0000 [r223489] Russell Bryant * main/autoservice.c, /: Merged revisions 223487 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r223487 | russell | 2009-10-11 12:25:42 -0500 (Sun, 11 Oct 2009) | 17 lines Merged revisions 223485-223486 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223485 | russell | 2009-10-11 12:22:52 -0500 (Sun, 11 Oct 2009) | 6 lines Don't use data outside of its scope. The purpose of this code was to have a hangup frame put on the list of deferred frames. However, the code that read the hangup frame was outside of the scope of where the hangup frame was declared. ........ r223486 | russell | 2009-10-11 12:25:06 -0500 (Sun, 11 Oct 2009) | 2 lines Remove some unnecessary code. ........ ................ 2009-10-09 23:11 +0000 [r223405] Jeff Peeler * channels/chan_dahdi.c, channels/chan_h323.c: Fix interpretation of PRIREDIRECTIONREASON set by chan_sip. This commit is the simplest way to solve a problem that has already been solved in trunk with the "COLP/CONP and Redirecting party information into Asterisk" commit. In trunk the redirection reason is translated into a generic redirect reason. I would have had to do the same fix except chan_sip never reads PRIREDIRECTREASON. So both chan_dahdi and chan_h323 have been modified to interpret the one different redirect reason of "no-answer" properly and set the ISDN reason code 2 of "no reply". (closes issue #15033) Reported by: steinwej 2009-10-09 21:00 +0000 [r223332] Kevin P. Fleming * /, apps/app_fax.c: Merged revisions 223330 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r223330 | kpfleming | 2009-10-09 15:58:44 -0500 (Fri, 09 Oct 2009) | 10 lines Initiate T.38 switchover when acting as called party, regardless of FAX direction. SendFAX() and ReceiveFAX() can be given options to indicate whether they should act as the calling or called party; this mode should be used to decide whether to initiate a switchover to T.38, not the direction that the FAX transfer will take place. (closes issue #16039) Reported by: jamicque ........ 2009-10-09 18:36 +0000 [r223277] Matthew Nicholson * main/channel.c, /: Merged revisions 223273 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r223273 | mnicholson | 2009-10-09 13:34:08 -0500 (Fri, 09 Oct 2009) | 14 lines Merged revisions 223225 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223225 | mnicholson | 2009-10-09 13:20:11 -0500 (Fri, 09 Oct 2009) | 8 lines Signal timeouts by returning AST_CONTROL_RINGING when originating calls. (closes issue #15104) Reported by: nblasgen Patches: manager-timeout1.diff uploaded by mnicholson (license 96) Tested by: nblasgen, mnicholson ........ ................ 2009-10-09 18:25 +0000 [r223241] Mark Michelson * apps/app_dial.c, /: Merged revisions 223215 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r223215 | mmichelson | 2009-10-09 13:17:34 -0500 (Fri, 09 Oct 2009) | 9 lines Recorded merge of revisions 223213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223213 | mmichelson | 2009-10-09 13:17:12 -0500 (Fri, 09 Oct 2009) | 3 lines Fix potential memory leak in app_dial.c ........ ................ 2009-10-09 17:56 +0000 [r223209] David Vossel * /, channels/chan_sip.c: Merged revisions 223206 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r223206 | dvossel | 2009-10-09 12:53:37 -0500 (Fri, 09 Oct 2009) | 16 lines Merged revisions 223205 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223205 | dvossel | 2009-10-09 12:52:35 -0500 (Fri, 09 Oct 2009) | 10 lines fixes sip registration using authuser in user.conf (closes issue #14954) Reported by: tornblad Tested by: mmichelson, tornblad, dvossel ........ ................ 2009-10-09 17:27 +0000 [r223171] Matthew Nicholson * cdr/cdr_sqlite3_custom.c, /: Merged revisions 223136 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r223136 | mnicholson | 2009-10-09 12:14:38 -0500 (Fri, 09 Oct 2009) | 8 lines Don't close the sqlite database when reloading. Only close the database when unloading. (closes issue #15953) Reported by: frawd Patches: sqlite3_rev220097.diff uploaded by frawd (license 610) Tested by: frawd ........ 2009-10-09 17:10 +0000 [r223090-223134] David Vossel * /, channels/chan_sip.c: Merged revisions 223132 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r223132 | dvossel | 2009-10-09 11:54:02 -0500 (Fri, 09 Oct 2009) | 9 lines 'auth=' did not parse md5 secret correctly (closes issue #15949) Reported by: ebroad Patches: authparsefix.patch uploaded by ebroad (license 878) 15949_trunk.diff uploaded by dvossel (license 671) Tested by: ebroad ........ * /, channels/chan_sip.c: Merged revisions 223088 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r223088 | dvossel | 2009-10-09 10:49:30 -0500 (Fri, 09 Oct 2009) | 14 lines p->peerauth is always empty in transmit_register() When using callbackextension or specifing the peer name in a registration string, the peer's specific auth settings set by the "auth=" strings within the peer definition are not used by the registration. Thanks to ebroad for reporting the issue and providing the patch. (closes issue #15955) Reported by: ebroad Patches: regauthfix.patch uploaded by ebroad (license 878) ........ 2009-10-08 19:57 +0000 [r222882] Russell Bryant * include/asterisk/frame.h, include/asterisk/file.h, main/frame.c, /, main/file.c: Merged revisions 222880 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r222880 | russell | 2009-10-08 14:52:03 -0500 (Thu, 08 Oct 2009) | 51 lines Merged revisions 222878 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222878 | russell | 2009-10-08 14:45:47 -0500 (Thu, 08 Oct 2009) | 44 lines Make filestream frame handling safer by isolating frames before returning them. This patch is related to a number of issues on the bug tracker that show crashes related to freeing frames that came from a filestream. A number of fixes have been made over time while trying to figure out these problems, but there re still people seeing the crash. (Note that some of these bug reports include information about other problems. I am specifically addressing the filestream frame crash here.) I'm still not clear on what the exact problem is. However, what is _very_ clear is that we have seen quite a few problems over time related to unexpected behavior when we try to use embedded frames as an optimization. In some cases, this optimization doesn't really provide much due to improvements made in other areas. In this case, the patch modifies filestream handling such that the embedded frame will not be returned. ast_frisolate() is used to ensure that we end up with a completely mallocd frame. In reality, though, we will not actually have to malloc every time. For filestreams, the frame will almost always be allocated and freed in the same thread. That means that the thread local frame cache will be used. So, going this route doesn't hurt. With this patch in place, some people have reported success in not seeing the crash anymore. (SWP-150) (AST-208) (ABE-1834) (issue #15609) Reported by: aragon Patches: filestream_frisolate-1.4.diff2.txt uploaded by russell (license 2) Tested by: aragon, russell (closes issue #15817) Reported by: zerohalo Tested by: zerohalo (closes issue #15845) Reported by: marhbere Review: https://reviewboard.asterisk.org/r/386/ ........ ................ 2009-10-08 19:42 +0000 [r222875] David Vossel * main/netsock.c, /, include/asterisk/netsock.h: Merged revisions 222873 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r222873 | dvossel | 2009-10-08 14:35:30 -0500 (Thu, 08 Oct 2009) | 6 lines fixes an ast_netsock_list memory leak. ABE-1998 Review: https://reviewboard.asterisk.org/r/395/ ........ 2009-10-08 16:49 +0000 [r222694-222801] Richard Mudgett * channels/misdn_config.c, /: Merged revisions 222799 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r222799 | rmudgett | 2009-10-08 11:44:33 -0500 (Thu, 08 Oct 2009) | 19 lines Merged revisions 222797 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222797 | rmudgett | 2009-10-08 11:33:06 -0500 (Thu, 08 Oct 2009) | 12 lines Fix memory leak if chan_misdn config parameter is repeated. Memory leak when the same config option is set more than once in an misdn.conf section. Why must this be considered? Templates! Defining a template with default port options and later adding to or overriding some of them. Patches: memleak-misdn.patch JIRA ABE-1998 ........ ................ * channels/chan_misdn.c, /: Merged revisions 222692 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r222692 | rmudgett | 2009-10-07 16:56:36 -0500 (Wed, 07 Oct 2009) | 21 lines Merged revisions 222691 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222691 | rmudgett | 2009-10-07 16:51:24 -0500 (Wed, 07 Oct 2009) | 14 lines chan_misdn.c:process_ast_dsp() memory leak misdn.conf: astdtmf must be set to "yes". With "no", buffer loss does not occur. The translated frame "f2" when passing through ast_dsp_process() is not freed whenever it is not used further in process_ast_dsp(). Then in the end it is never ever freed. Patches: translate.patch JIRA ABE-1993 ........ ................ 2009-10-07 17:46 +0000 [r222545] David Vossel * /, channels/chan_sip.c: Merged revisions 222543 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r222543 | dvossel | 2009-10-07 12:44:52 -0500 (Wed, 07 Oct 2009) | 14 lines Merged revisions 222542 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222542 | dvossel | 2009-10-07 12:41:21 -0500 (Wed, 07 Oct 2009) | 8 lines crash on transfer handle_invite_replaces() attempts to uplock a pvt's owner channel without first verifing that it exists. (issue #16027) ........ ................ 2009-10-06 23:58 +0000 [r222353-222465] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 222463 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r222463 | jpeeler | 2009-10-06 18:56:01 -0500 (Tue, 06 Oct 2009) | 14 lines Merged revisions 222462 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222462 | jpeeler | 2009-10-06 18:51:19 -0500 (Tue, 06 Oct 2009) | 8 lines Add missing unlock(s) in dahdi_read (two cases in trunk) (closes issue #15683) Reported by: alecdavis ........ ................ * channels/chan_dahdi.c: Fix potential crash when entire span request is received. The variable index used in this scenario for accessing the dahdi_pvts was wrong and was most likely copied from the several other places it is used correctly. (closes issue #15998) Reported by: tsearle Patches: dahdi_reset_crash.patch uploaded by tsearle (license 373) Modified: branches/1.4/channels/chan_dahdi.c * channels/chan_dahdi.c, /: Merged revisions 222351 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r222351 | jpeeler | 2009-10-06 15:35:19 -0500 (Tue, 06 Oct 2009) | 9 lines Fix 222298 (crash during destruction of second channel when variable set with setvar). I mistakenly reasoned that setvar would be used on all channels. Since it can be set per channel, give each dahdi channel a copy of the variable. (related to #15899) ........ 2009-10-06 19:34 +0000 [r222310] Tilghman Lesher * res/res_config_pgsql.c, /, cdr/cdr_pgsql.c: Recorded merge of revisions 222309 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r222309 | tilghman | 2009-10-06 14:31:39 -0500 (Tue, 06 Oct 2009) | 10 lines Change schema query to involve the use of an optional schema parameter. This change is done in such a way as to allow the driver to continue to function with older databases which don't have these features. (closes issue #16000) Reported by: jamicque Patches: 20091002__issue16000.diff.txt uploaded by tilghman (license 14) 20091002__issue16000__1.6.1.diff.txt uploaded by tilghman (license 14) Tested by: jamicque ........ 2009-10-06 19:26 +0000 [r222303] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 222298 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r222298 | jpeeler | 2009-10-06 14:24:59 -0500 (Tue, 06 Oct 2009) | 9 lines Fix crash during destruction of second channel when variable set with setvar. The setvar line in chan_dahdi.conf is shared among all the channels, so make sure to only free the resources only when the last channel is destroyed. (closes issue #15899) Reported by: tzafrir ........ 2009-10-06 19:20 +0000 [r222282] Tilghman Lesher * res/ael/pval.c, /: Merged revisions 222273 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r222273 | tilghman | 2009-10-06 14:17:11 -0500 (Tue, 06 Oct 2009) | 5 lines When we call a gosub routine, the variables should be scoped to avoid contaminating the caller. This affected the ~~EXTEN~~ hack, where a subroutine might have changed the value before it was used in the caller. Patch by myself, tested by ebroad on #asterisk ........ 2009-11-04 Leif Madsen * Release Asterisk 1.6.1.9 * AST-2009-008 and AST-2009-009 2009-10-26 Leif Madsen * Release Asterisk 1.6.1.8 * AST-2009-007 2009-10-06 Leif Madsen * Release Asterisk 1.6.1.7-rc2 2009-10-06 01:36 +0000 [r222112-222186] Kevin P. Fleming * apps/app_queue.c, channels/chan_iax2.c, main/astobj2.c, res/res_odbc.c, /, channels/chan_sip.c, funcs/func_dialgroup.c, include/asterisk/astobj2.h, res/res_phoneprov.c, channels/chan_console.c, res/res_musiconhold.c: Merged revisions 222176 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r222176 | kpfleming | 2009-10-05 20:24:24 -0500 (Mon, 05 Oct 2009) | 27 lines Recorded merge of revisions 222152 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct 2009) | 20 lines Fix ao2_iterator API to hold references to containers being iterated. See Mantis issue for details of what prompted this change. Additional notes: This patch changes the ao2_iterator API in two ways: F_AO2I_DONTLOCK has become an enum instead of a macro, with a name that fits our naming policy; also, it is now necessary to call ao2_iterator_destroy() on any iterator that has been created. Currently this only releases the reference to the container being iterated, but in the future this could also release other resources used by the iterator, if the iterator implementation changes to use additional resources. (closes issue #15987) Reported by: kpfleming Review: https://reviewboard.asterisk.org/r/383/ ........ ................ * main/udptl.c, /, channels/chan_sip.c, configs/udptl.conf.sample, UPGRADE.txt, configs/sip.conf.sample: Merged revisions 222110 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r222110 | kpfleming | 2009-10-05 14:45:00 -0500 (Mon, 05 Oct 2009) | 25 lines Allow non-compliant T.38 endpoints to be supportable via configuration option. Many T.38 endpoints incorrectly send the maximum IFP frame size they can accept as the T38FaxMaxDatagram value in their SDP, when in fact this value is supposed to be the maximum UDPTL payload size (datagram size) they can accept. If the value they supply is small enough (a commonly supplied value is '72'), T.38 UDPTL transmissions will likely fail completely because the UDPTL packets will not have enough room for a primary IFP frame and the redundancy used for error correction. If this occurs, the Asterisk UDPTL stack will emit log messages warning that data loss may occur, and that the value may need to be overridden. This patch extends the 't38pt_udptl' configuration option in sip.conf to allow the administrator to override the value supplied by the remote endpoint and supply a value that allows T.38 FAX transmissions to be successful with that endpoint. In addition, in any SIP call where the override takes effect, a debug message will be printed to that effect. This patch also removes the T38FaxMaxDatagram configuration option from udptl.conf.sample, since it has not actually had any effect for a number of releases. In addition, this patch cleans up the T.38 documentation in sip.conf.sample (which incorrectly documented that T.38 support was passthrough only). (issue #15586) Reported by: globalnetinc ........ 2009-10-02 17:36 +0000 [r222035] David Vossel * channels/chan_iax2.c, /: Merged revisions 222030 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r222030 | dvossel | 2009-10-02 12:34:07 -0500 (Fri, 02 Oct 2009) | 9 lines Merged revisions 222026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222026 | dvossel | 2009-10-02 12:32:13 -0500 (Fri, 02 Oct 2009) | 3 lines Removes unnecessary unlock, clarifies a memcpy. ........ ................ 2009-10-02 17:01 +0000 [r221969-221973] Tilghman Lesher * main/astobj2.c, /, funcs/func_lock.c: Merged revisions 221971 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r221971 | tilghman | 2009-10-02 11:59:57 -0500 (Fri, 02 Oct 2009) | 9 lines Merged revisions 221970 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221970 | tilghman | 2009-10-02 11:58:03 -0500 (Fri, 02 Oct 2009) | 2 lines Ensure the result of the hash function is positive. Negative array offsets suck. ........ ................ * funcs/func_lock.c: Hash needs to return a positive integer 2009-10-02 13:04 +0000 [r221964] Sean Bright * funcs/func_strings.c: Revert XML docs that ended up in the 1.6.0 and 1.6.1 branches during a merge. 2009-10-02 03:06 +0000 [r221922] Tilghman Lesher * /, main/logger.c: Merged revisions 221920 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r221920 | tilghman | 2009-10-01 22:04:34 -0500 (Thu, 01 Oct 2009) | 4 lines Initialize a variable that we check immediately upon startup. (closes issue #15973) Reported by: atis ........ 2009-10-02 01:26 +0000 [r221871] Richard Mudgett * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h, /: Merged revisions 221844 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r221844 | rmudgett | 2009-10-01 20:09:31 -0500 (Thu, 01 Oct 2009) | 33 lines Merged revisions 221769 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221769 | rmudgett | 2009-10-01 18:18:28 -0500 (Thu, 01 Oct 2009) | 26 lines Occasionally losing use of B channels in chan_misdn. I have not been able to reproduce the problem of losing channels. However, I have seen in the code a reentrancy problem that might give these symptoms. The reentrancy patch does several things: 1) Guards B channel and B channel structure allocation. 2) Makes the B channel structure find routines more precise in locating records. 3) Never leave a B channel allocated if we received cause 44. The last item may cause temporary outgoing call problems, but they should clear when the line becomes idle. (closes issue #15490) Reported by: slutec18 Patches: issue15490_channel_alloc_reentrancy.patch uploaded by rmudgett (license 664) Tested by: rmudgett, slutec18 (closes issue #15458) Reported by: FabienToune Patches: issue15458_channel_alloc_reentrancy.patch uploaded by rmudgett (license 664) Tested by: FabienToune, rmudgett, slutec18 ........ ................ 2009-10-02 00:06 +0000 [r221743-221779] Tilghman Lesher * main/asterisk.c, main/rtp.c, /, main/say.c: Merged revisions 221777 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r221777 | tilghman | 2009-10-01 18:59:15 -0500 (Thu, 01 Oct 2009) | 9 lines Merged revisions 221776 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221776 | tilghman | 2009-10-01 18:53:12 -0500 (Thu, 01 Oct 2009) | 2 lines Fix a bunch of off-by-one errors ........ ................ * /, channels/chan_sip.c: Merged revisions 221705 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r221705 | tilghman | 2009-10-01 15:09:46 -0500 (Thu, 01 Oct 2009) | 2 lines Revision 220906 (a merge from 1.4) was not merged correctly, causing a problem with non-dynamic peers. ........ 2009-10-01 19:52 +0000 [r221702] David Vossel * /, channels/chan_sip.c: Merged revisions 221697 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r221697 | dvossel | 2009-10-01 14:33:33 -0500 (Thu, 01 Oct 2009) | 9 lines outbound tls connections were not defaulting to port 5061 (closes issue #15854) Reported by: dvossel Patches: sip_port_config_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel ........ 2009-10-01 17:01 +0000 [r221661] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 221554,221589 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r221554 | oej | 2009-10-01 02:00:04 -0500 (Thu, 01 Oct 2009) | 3 lines Simplify code for porturi, use TRUE/FALSE constructs when it's just TRUE or FALSE. ................ r221589 | mnicholson | 2009-10-01 10:26:20 -0500 (Thu, 01 Oct 2009) | 9 lines Merged revisions 221588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221588 | mnicholson | 2009-10-01 10:24:00 -0500 (Thu, 01 Oct 2009) | 2 lines Use unsigned ints for portinuri flags. ........ ................ 2009-10-01 16:19 +0000 [r221602] Kevin P. Fleming * main/udptl.c, /, configs/udptl.conf.sample, UPGRADE.txt: Merged revisions 221592 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r221592 | kpfleming | 2009-10-01 11:16:09 -0500 (Thu, 01 Oct 2009) | 12 lines Remove ability to control T.38 FAX error correction from udptl.conf. chan_sip has had the ability to control T.38 FAX error correction mode on a per-peer (or global) basis for a couple of releases now, which is where it should have been all along. This patch removes the ability to configure it in udptl.conf, but issues a warning if the user tries to do, telling them to look at sip.conf.sample for how to configure it now. For any SIP peers that are T.38 enabled in sip.conf, there is already a default for FEC error correction even if the user does not specify any mode, so this change will not turn off error correction by default, it will have the same default value that has been in the udptl.conf sample file. ........ 2009-09-30 23:10 +0000 [r221478-221487] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 221484 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r221484 | mnicholson | 2009-09-30 18:04:03 -0500 (Wed, 30 Sep 2009) | 2 lines Cleaned up merge from r221432 ........ * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 221432 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r221432 | mnicholson | 2009-09-30 15:40:20 -0500 (Wed, 30 Sep 2009) | 17 lines Merged revisions 221360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221360 | mnicholson | 2009-09-30 14:36:06 -0500 (Wed, 30 Sep 2009) | 10 lines Fix SRV lookup and Request-URI generation in chan_sip. This patch adds a new field "portinuri" to the sip dialog struct and the sip peer struct. That field is used during RURI generation to determine if the port should be included in the RURI. It is also used in some places to determine if an SRV lookup should occur. (closes issue #14418) Reported by: klaus3000 Tested by: klaus3000, mnicholson Review: https://reviewboard.asterisk.org/r/369/ ........ ................ 2009-09-30 21:41 +0000 [r221370-221470] Matthias Nick * apps/app_queue.c, /: Merged revisions 221436 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r221436 | mnick | 2009-09-30 16:15:01 -0500 (Wed, 30 Sep 2009) | 2 lines Prevents from division by zero ........ * configs/cdr_custom.conf.sample, /, funcs/func_strings.c: Merged revisions 221368 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r221368 | mnick | 2009-09-30 14:42:36 -0500 (Wed, 30 Sep 2009) | 23 lines Merged revisions 221153,221157,221303 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221153 | mnick | 2009-09-30 10:37:39 -0500 (Wed, 30 Sep 2009) | 2 lines check bounds - prevents for buffer overflow ........ r221157 | mnick | 2009-09-30 10:41:46 -0500 (Wed, 30 Sep 2009) | 8 lines added a new dialplan function 'CSV_QUOTE' and changed the cdr_custom.sample.conf (closes issue #15471) Reported by: dkerr Patches: csv_quote_14.txt uploaded by mnick (license ) Tested by: mnick ........ r221303 | mnick | 2009-09-30 14:02:00 -0500 (Wed, 30 Sep 2009) | 2 lines changed the prototype definition of csv_quote ........ ................ 2009-09-30 18:58 +0000 [r221302] Terry Wilson * main/rtp.c, /, channels/chan_sip.c, include/asterisk/rtp.h, configs/sip.conf.sample: Merged revisions 221266 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r221266 | twilson | 2009-09-30 12:52:30 -0500 (Wed, 30 Sep 2009) | 32 lines Merged revisions 221086 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) | 25 lines Change the SSRC by default when our media stream changes Be default, change SSRC when doing an audio stream changes Asterisk doesn't honor marker bit when reinvited to already-bridged RTP streams,resulting in far-end stack discarding packets with "old" timestamps that areactually part of a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is a reinvite, unless the 'constantssrc' is set to true in sip.conf. The original issue reported to Digium support detailed the following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in fromITSP, Asterisk dials the app server which sends a re-invite back toAsterisk--not to negotiate to send media directly to the ITSP, but to indicatethat it's changing the stream it's sending to Asterisk. The app servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker bit on the new stream. Asterisk passes through the teimstamp of the new stream, butdoes not reset the SSRC, sequence numbers, or set the marker bit. When the timestamp on the new stream is older than the timestamp on the originalstream, the ITSP (which doesn't know there has been any change) discards the newframes because it thinks they are too old. This patch addresses this by changing the SSRC on a stream update unless constantssrc=true is set in sip.conf. Review: https://reviewboard.asterisk.org/r/374/ ........ ................ 2009-09-30 16:57 +0000 [r221203] Tilghman Lesher * main/channel.c, /: Merged revisions 221201 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r221201 | tilghman | 2009-09-30 11:56:42 -0500 (Wed, 30 Sep 2009) | 14 lines Merged revisions 221200 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221200 | tilghman | 2009-09-30 11:55:21 -0500 (Wed, 30 Sep 2009) | 7 lines Avoid a potential NULL dereference. (closes issue #15865) Reported by: kobaz Patches: 20090915__issue15865.diff.txt uploaded by tilghman (license 14) Tested by: kobaz ........ ................ 2009-09-30 14:55 +0000 [r221088] Sean Bright * apps/app_voicemail.c, /: Merged revisions 221085 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r221085 | seanbright | 2009-09-30 10:47:58 -0400 (Wed, 30 Sep 2009) | 9 lines Clarify documentation for VoiceMailMain()'s a() option. We require box numbers, not names as the documentation implies. (issue #14740) Reported by: pj Patches: __20090729-app_voicemail-documentation.patch uploaded by lmadsen (license 10) Tested by: seanbright, lmadsen ........ 2009-09-30 04:41 +0000 [r220998-221046] Tilghman Lesher * /, funcs/func_lock.c: Recorded merge of revisions 221044 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r221044 | tilghman | 2009-09-29 23:32:36 -0500 (Tue, 29 Sep 2009) | 8 lines Allow locks to be inherited through a masquerade without causing starvation. (closes issue #14859) Reported by: atis Patches: 20090821__issue14859.diff.txt uploaded by tilghman (license 14) 20090925__issue14859__1.6.1.diff.txt uploaded by tilghman (license 14) Tested by: atis, tilghman ........ * /, channels/chan_sip.c: Merged revisions 220906 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r220906 | tilghman | 2009-09-29 14:57:37 -0500 (Tue, 29 Sep 2009) | 16 lines Merged revisions 220873 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220873 | tilghman | 2009-09-29 12:59:26 -0500 (Tue, 29 Sep 2009) | 9 lines Reduce CPU usage related to building a peer merely for devicestates. This fixes a 100% CPU problem in the SIP driver, found by profiling the driver while the problem was occurring. (closes issue #14309) Reported by: pkempgen Patches: 20090924__issue14309.diff.txt uploaded by tilghman (license 14) Tested by: pkempgen, vrban ........ ................ 2009-09-29 20:25 +0000 [r220938] Matthew Nicholson * apps/app_chanspy.c: Avoid a deadlock in chanspy, just in case the spyee is masqueraded and chanspy_ds_chan_fixup() is called with the channel locked. (closes issue #15965) Reported by: atis Patches: chanspy-deadlock-fix1.diff uploaded by mnicholson (license 96) Tested by: atis 2009-09-29 17:05 +0000 [r220835] Jeff Peeler * apps/app_voicemail.c, /: Merged revisions 220833 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r220833 | jpeeler | 2009-09-29 11:58:29 -0500 (Tue, 29 Sep 2009) | 12 lines Make deletion of temporary greetings work properly with IMAP_STORAGE When imapgreetings was set to yes, the message was being deleted but wasn't actually being expunged. When imapgreetings was set to no, the file based message was not being deleted at all. All good now! (closes issue #14949) Reported by: noahisaac Patches: vm_tempgreeting_removal.patch uploaded by noahisaac (license 748), modified by me ........ 2009-09-28 19:13 +0000 [r220724] Sean Bright * /, Makefile.rules: Merged revisions 220721 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r220721 | seanbright | 2009-09-28 15:11:20 -0400 (Mon, 28 Sep 2009) | 10 lines Merged revisions 220717 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220717 | seanbright | 2009-09-28 15:09:25 -0400 (Mon, 28 Sep 2009) | 3 lines When selecting DONT_OPTIMIZE in menuselect, explicitly pass -O0 to the compiler so we override any default optimization levels for a particular install. ........ ................ 2009-09-26 15:12 +0000 [r220588] Tilghman Lesher * /, include/asterisk/aes.h: Merged revisions 220586 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r220586 | tilghman | 2009-09-26 10:10:28 -0500 (Sat, 26 Sep 2009) | 2 lines Allow AES to compile, when OpenSSL is not present. ........ 2009-09-24 20:38 +0000 [r220371] David Vossel * main/tcptls.c, /: Merged revisions 220365 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r220365 | dvossel | 2009-09-24 15:37:20 -0500 (Thu, 24 Sep 2009) | 8 lines fixes tcptls_session memory leak caused by ref count error (closes issue #15939) Reported by: dvossel Review: https://reviewboard.asterisk.org/r/375/ ........ 2009-09-24 19:42 +0000 [r220291] Tilghman Lesher * apps/app_playback.c, main/pbx.c, /, apps/app_disa.c: Merged revisions 220289 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r220289 | tilghman | 2009-09-24 14:41:02 -0500 (Thu, 24 Sep 2009) | 13 lines Merged revisions 220288 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220288 | tilghman | 2009-09-24 14:39:41 -0500 (Thu, 24 Sep 2009) | 6 lines Implicitly sending a progress signal breaks some applications. Call Progress() in your dialplan if you explicitly want progress to be sent. (Reverts change 216430, closes issue #15957) Reported by: Pavel Troller on the Asterisk-Dev mailing list http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html ........ ................ 2009-09-24 18:22 +0000 [r220102-220220] Sean Bright * Makefile, /: Merged revisions 220217 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r220217 | seanbright | 2009-09-24 14:19:41 -0400 (Thu, 24 Sep 2009) | 9 lines Merged revisions 220213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220213 | seanbright | 2009-09-24 14:18:18 -0400 (Thu, 24 Sep 2009) | 1 line Resolve parallel build warnings. Reported by Klaus Darilion on the asterisk-dev mailing list. ........ ................ * Makefile, build_tools/mkpkgconfig, /: Merged revisions 220100 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r220100 | seanbright | 2009-09-24 10:44:08 -0400 (Thu, 24 Sep 2009) | 9 lines Merged revisions 220099 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220099 | seanbright | 2009-09-24 10:41:57 -0400 (Thu, 24 Sep 2009) | 2 lines Remove the remaining bashisms in the Makefile/mkpkgconfig ........ ................ 2009-09-24 08:40 +0000 [r220030] Michiel van Baak * build_tools/mkpkgconfig, /: Merged revisions 220028 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r220028 | mvanbaak | 2009-09-24 10:36:18 +0200 (Thu, 24 Sep 2009) | 14 lines Merged revisions 220027 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220027 | mvanbaak | 2009-09-24 10:33:50 +0200 (Thu, 24 Sep 2009) | 7 lines mkpkgconfig does not need bash so make it use /bin/sh This fixes building on all systems that don't have bash at /bin/bash Reported by _ys on #asterisk-dev Tested by _ys on #asterisk-dev ........ ................ 2009-09-24 07:44 +0000 [r219988] Tilghman Lesher * apps/app_directory.c, /: Merged revisions 219987 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r219987 | tilghman | 2009-09-24 02:39:44 -0500 (Thu, 24 Sep 2009) | 8 lines Fix two possible crashes, one only in 1.6.1 and one in 1.6.1 forward. (closes issue #15739) Reported by: DLNoah, jeffg Patches: 20090914__issue15739.diff.txt uploaded by tilghman (license 14) 20090922__issue15739.diff.txt uploaded by tilghman (license 14) Tested by: DLNoah, jeffg ........ 2009-09-22 21:47 +0000 [r219820] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 219818 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219818 | tilghman | 2009-09-22 16:43:22 -0500 (Tue, 22 Sep 2009) | 17 lines Merged revisions 219816 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219816 | tilghman | 2009-09-22 16:37:03 -0500 (Tue, 22 Sep 2009) | 10 lines When IMAP variables were changed during a reload, Voicemail did not use the new values. This change introduces a configuration version variable, which ensures that connections with the old values are not reused but are allowed to expire normally. (closes issue #15934) Reported by: viniciusfontes Patches: 20090922__issue15934.diff.txt uploaded by tilghman (license 14) Tested by: viniciusfontes ........ ................ 2009-09-21 17:02 +0000 [r219723] David Vossel * channels/chan_iax2.c, /: Merged revisions 219721 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219721 | dvossel | 2009-09-21 11:59:05 -0500 (Mon, 21 Sep 2009) | 9 lines Merged revisions 219720 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219720 | dvossel | 2009-09-21 11:55:53 -0500 (Mon, 21 Sep 2009) | 3 lines Reverting merge 219520. This change was not necessary. ........ ................ 2009-09-20 18:21 +0000 [r219667] Tilghman Lesher * /, main/file.c: Merged revisions 219654 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219654 | tilghman | 2009-09-20 12:55:49 -0500 (Sun, 20 Sep 2009) | 15 lines Merged revisions 219653 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219653 | tilghman | 2009-09-20 12:52:05 -0500 (Sun, 20 Sep 2009) | 8 lines Really stop the stream, when ast_closestream() is called. (closes issue #15129) Reported by: bmh Patches: 20090918__issue15129.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/372/ ........ ................ 2009-09-19 03:10 +0000 [r219589] Russell Bryant * channels/chan_iax2.c, /: Merged revisions 219587 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219587 | russell | 2009-09-18 21:59:52 -0500 (Fri, 18 Sep 2009) | 13 lines Merged revisions 219586 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219586 | russell | 2009-09-18 21:51:13 -0500 (Fri, 18 Sep 2009) | 6 lines Make sure the iax_pvt exists before dereferencing it. This fixes the latest crash posted on issue 15609. (issue #15609) ........ ................ 2009-09-18 23:22 +0000 [r219453-219522] David Vossel * channels/chan_iax2.c, /: Merged revisions 219520 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219520 | dvossel | 2009-09-18 18:20:58 -0500 (Fri, 18 Sep 2009) | 15 lines Merged revisions 219519 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219519 | dvossel | 2009-09-18 18:19:50 -0500 (Fri, 18 Sep 2009) | 9 lines iax2 frame double free The iax frame's retrans sched id was written over right before iax2_frame_free was called. In iax2_frame_free that retrans id is used to delete the sched item. By writing over the retrans field before the sched item could be deleted, it was possible for a retransmit to occur on a freed frame. ........ ................ * /, channels/chan_sip.c: Merged revisions 219451 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219451 | dvossel | 2009-09-18 11:20:41 -0500 (Fri, 18 Sep 2009) | 20 lines Merged revisions 219450 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219450 | dvossel | 2009-09-18 11:19:15 -0500 (Fri, 18 Sep 2009) | 14 lines via-header branches not updated correctly on INVITE INVITE requests must always contain a new unique branch id. When a new branch id is created for an INVITE, the dialog's invite_branch variable must be updated so CANCEL requests use the correct branch id. (closes issue #15262) Reported by: maniax Patches: asterisk-1.6.1.0-sip-branch.patch uploaded by tweety (license 608) invite_new_branch_trunk.diff uploaded by dvossel (license 671) Tested by: maniax, dvossel ........ ................ 2009-09-18 13:57 +0000 [r219414] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 219412 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r219412 | tilghman | 2009-09-18 08:54:51 -0500 (Fri, 18 Sep 2009) | 6 lines Missing value setting line for maxsecs/maxmessage (closes issue #15696) Reported by: fhackenberger Patches: maxsecs.patch uploaded by fhackenberger (license 592) ........ 2009-09-17 22:36 +0000 [r219367] Joshua Colp * /, channels/chan_sip.c: Merged revisions 219324 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219324 | mmichelson | 2009-09-17 17:22:01 -0500 (Thu, 17 Sep 2009) | 12 lines Merged revisions 219320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219320 | mmichelson | 2009-09-17 17:20:50 -0500 (Thu, 17 Sep 2009) | 6 lines Send a 100 Trying response when we detect a spiral. This was problematic during spiral tests at SIPit... along with some other things as well. ........ ................ 2009-09-17 22:04 +0000 [r219306] David Vossel * /, channels/chan_sip.c: Merged revisions 219304 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219304 | dvossel | 2009-09-17 16:59:21 -0500 (Thu, 17 Sep 2009) | 27 lines Merged revisions 219303 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219303 | dvossel | 2009-09-17 16:29:37 -0500 (Thu, 17 Sep 2009) | 21 lines INVITE w/Replaces deadlock fix This patch cleans up the locking logic in chan_sip.c's handle_invite_replaces() function as well as making use of ast_do_masquerade() rather than forcing the masquerade on an ast_read(). The code had several redundant unlocks that would result in 'freed more times than we've locked!' errors. I cleaned these up as well as moving all the unlock logic to the end of the function. This patch should also resolve the issue people were having with the replacecall channel never being unlocked with one legged calls. (closes issue #15151) Reported by: irroot Patches: invite_w_replaces_1.4.diff uploaded by dvossel (license 671) Tested by: irroot, dvossel Review: https://reviewboard.asterisk.org/r/371/ ........ ................ 2009-09-17 19:58 +0000 [r219266] Joshua Colp * /, channels/chan_sip.c: Merged revisions 219264 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r219264 | file | 2009-09-17 14:57:39 -0500 (Thu, 17 Sep 2009) | 2 lines Ensure no spaces exist before "refresher=" when doing the comparison. ........ 2009-09-17 Leif Madsen * Released Asterisk 1.6.1.7-rc1 2009-09-17 15:44 +0000 [r219199] Matthew Nicholson * main/channel.c, /, include/asterisk/cdr.h, include/asterisk/channel.h: Merged revisions 219139 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219139 | mnicholson | 2009-09-17 10:18:01 -0500 (Thu, 17 Sep 2009) | 17 lines Merged revisions 219136 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219136 | mnicholson | 2009-09-17 09:58:39 -0500 (Thu, 17 Sep 2009) | 10 lines Prevent a potential race condition and crash when hanging up a channel by removing the channel from the channel list before begining channel tear down. This fix may potentially cause problems with CDR backends that access the channel a CDR is associated with via the channel list. This fix makes the channel unavabile at the time when the CDR backend is invoked. This has been documented in include/asterisk/cdr.h. (closes issue #15316) Reported by: vmarrone Tested by: mnicholson Review: https://reviewboard.asterisk.org/r/362/ ........ ................ 2009-09-16 23:52 +0000 [r219062] Tilghman Lesher * main/config.c, configs/extensions.conf.sample, /: Merged revisions 219061 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r219061 | tilghman | 2009-09-16 18:42:12 -0500 (Wed, 16 Sep 2009) | 15 lines Merged revisions 219023 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219023 | tilghman | 2009-09-16 18:21:53 -0500 (Wed, 16 Sep 2009) | 8 lines Properly deal with quotes in the arguments of '#exec' includes. (closes issue #15583) Reported by: pkempgen Patches: 20090726__issue15583.diff.txt uploaded by tilghman (license 14) 20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license 169) Tested by: pkempgen ........ ................ 2009-09-16 19:27 +0000 [r218936] Mark Michelson * /, channels/chan_sip.c: Merged revisions 218933 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r218933 | mmichelson | 2009-09-16 14:25:36 -0500 (Wed, 16 Sep 2009) | 12 lines Reverse order of args to fread. This way, we don't always write a null byte into byte 1 of the buffer (closes issue #15905) Reported by: ebroad Patches: freadfix.patch uploaded by ebroad (license 878) Tested by: ebroad ........ 2009-09-16 19:24 +0000 [r218932] Joshua Colp * /, channels/chan_sip.c: Merged revisions 218918 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r218918 | file | 2009-09-16 13:31:47 -0500 (Wed, 16 Sep 2009) | 5 lines On TCP and TLS connections do not attempt to stop retransmission of the packet internally. This was preventing responses from being properly processed because the packet was not being found causing handle_response to return prematurely. ........ 2009-09-16 18:23 +0000 [r218890] David Brooks * main/pbx.c, /: Merged revisions 218868 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218868 | dbrooks | 2009-09-16 13:06:42 -0500 (Wed, 16 Sep 2009) | 20 lines Merged revisions 218867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218867 | dbrooks | 2009-09-16 13:00:45 -0500 (Wed, 16 Sep 2009) | 13 lines Fixes CID pattern matching behavior to mirror that of extension pattern matching. Pattern matching for extensions uses a type of scoring system, giving values for specificity to each character in the pattern. Unfortunately, this is done character by character, in order. This does lead to some less specific patterns being first in line for matching, but it will usually get the job done. This patch merely brings CID matching to the same level as extension matching. This patch does not attempt to tackle the problem shared by extension matching. (closes issue #14708) Reported by: klaus3000 ........ ................ 2009-09-16 13:37 +0000 [r218801] Russell Bryant * contrib/firmware/iax/iaxy.bin (removed), /, UPGRADE.txt: Merged revisions 218799 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218799 | russell | 2009-09-16 08:34:41 -0500 (Wed, 16 Sep 2009) | 16 lines Merged revisions 218798 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218798 | russell | 2009-09-16 08:33:43 -0500 (Wed, 16 Sep 2009) | 9 lines Remove the IAXy firmware from Asterisk. The firmware can now be found on downloads.digium.com, where the rest of our binary downloads live. This was the last part of our Asterisk tarballs that was considered non-free by Debian. :-) (closes issue #15838) Reported by: paravoid ........ ................ 2009-09-15 22:46 +0000 [r218727-218734] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 218731 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218731 | tilghman | 2009-09-15 17:33:10 -0500 (Tue, 15 Sep 2009) | 13 lines Merged revisions 218730 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218730 | tilghman | 2009-09-15 17:27:41 -0500 (Tue, 15 Sep 2009) | 6 lines If the user enters the same password as before, don't signal an error when the change does nothing. (closes issue #15492) Reported by: cbbs70a Patches: 20090713__issue15492.diff.txt uploaded by tilghman (license 14) ........ ................ * /, channels/chan_gtalk.c: Merged revisions 139281,175058,175089 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk (closes issue #13985) ................ r139281 | phsultan | 2008-08-21 04:55:31 -0500 (Thu, 21 Aug 2008) | 5 lines Fix two memory leaks in chan_gtalk, thanks Eliel! (closes issue #13310) Reported by: eliel Patches: chan_gtalk.c.patch uploaded by eliel (license 64) ................ r175058 | phsultan | 2009-02-12 04:31:36 -0600 (Thu, 12 Feb 2009) | 20 lines Merged revisions 175029 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175029 | phsultan | 2009-02-12 11:16:21 +0100 (Thu, 12 Feb 2009) | 12 lines Set the initiator attribute to lowercase in our replies when receiving calls. This attribute contains a JID that identifies the initiator of the GoogleTalk voice session. The GoogleTalk client discards Asterisk's replies if the initiator attribute contains uppercase characters. (closes issue #13984) Reported by: jcovert Patches: chan_gtalk.2.patch uploaded by jcovert (license 551) Tested by: jcovert ........ ................ r175089 | phsultan | 2009-02-12 08:25:03 -0600 (Thu, 12 Feb 2009) | 6 lines Issue a warning message if our candidate's IP is the loopback address. (closes issue #13985) Reported by: jcovert Tested by: phsultan ................ 2009-09-15 19:27 +0000 [r218689] David Vossel * /, channels/chan_sip.c: Merged revisions 218687 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r218687 | dvossel | 2009-09-15 14:22:37 -0500 (Tue, 15 Sep 2009) | 2 lines upward bound checking for port string to int conversion ........ 2009-09-15 16:18 +0000 [r218592] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 218586 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218586 | mnicholson | 2009-09-15 11:15:02 -0500 (Tue, 15 Sep 2009) | 15 lines Merged revisions 218578 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218578 | mnicholson | 2009-09-15 11:03:54 -0500 (Tue, 15 Sep 2009) | 8 lines Send request contact header field with response to registrer queries instead of the address of record. (closes issue #14438) Reported by: ravindrad Patches: regquerypatch uploaded by ravindrad (license 684) Tested by: ravindrad ........ ................ 2009-09-15 16:05 +0000 [r218581] Tilghman Lesher * apps/app_followme.c, /: Merged revisions 218579 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218579 | tilghman | 2009-09-15 11:04:41 -0500 (Tue, 15 Sep 2009) | 16 lines Merged revisions 218577 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218577 | tilghman | 2009-09-15 11:01:17 -0500 (Tue, 15 Sep 2009) | 9 lines Ensure FollowMe sets language in channels it creates. Also, not in the original bug report, but related fields are accountcode and musicclass, and the inheritance of datastores. (closes issue #15372) Reported by: Romik Patches: 20090828__issue15372.diff.txt uploaded by tilghman (license 14) Tested by: cervajs ........ ................ 2009-09-15 15:42 +0000 [r218574] Mark Michelson * /, channels/chan_sip.c: Merged revisions 218566 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r218566 | mmichelson | 2009-09-15 10:40:14 -0500 (Tue, 15 Sep 2009) | 4 lines Use a better method of ensuring null-termination of the buffer while reading the SDP when using TCP. ........ 2009-09-15 15:41 +0000 [r218569] Jeff Peeler * channels/chan_dahdi.c: Merged revisions 218430 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218430 | jpeeler | 2009-09-14 17:38:25 -0500 (Mon, 14 Sep 2009) | 18 lines Merged revisions 218401 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218401 | jpeeler | 2009-09-14 16:47:11 -0500 (Mon, 14 Sep 2009) | 11 lines Fix handling of DAHDI_EVENT_REMOVED event to prevent crash in do_monitor. After talking to rmudgett about some of his recent iflist locking changes, it was determined that the only place that would destroy a channel without being explicitly to do so was in handle_init_event. The loop to walk the interface list has been modified to wait to destroy the channel until the dahdi_pvt of the channel to be destroyed is no longer needed. (closes issue #15378) Reported by: samy ........ ................ 2009-09-15 15:12 +0000 [r218506] Mark Michelson * /, channels/chan_sip.c: Merged revisions 218499,218504 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r218499 | mmichelson | 2009-09-15 09:59:50 -0500 (Tue, 15 Sep 2009) | 3 lines Fix off-by-one error when reading SDP sent over TCP. ........ r218504 | mmichelson | 2009-09-15 10:05:53 -0500 (Tue, 15 Sep 2009) | 3 lines Ensure that SDP read from TCP socket is null-terminated. ........ 2009-09-15 15:04 +0000 [r218502] Kevin P. Fleming * sounds/Makefile, /: Merged revisions 218500 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218500 | kpfleming | 2009-09-15 11:02:21 -0400 (Tue, 15 Sep 2009) | 9 lines Merged revisions 218497 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218497 | kpfleming | 2009-09-15 10:55:58 -0400 (Tue, 15 Sep 2009) | 1 line Use proper hostname for downloading sound files. ........ ................ 2009-09-14 19:49 +0000 [r218363] Tilghman Lesher * sounds/Makefile, apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged revisions 218361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218361 | tilghman | 2009-09-14 14:29:48 -0500 (Mon, 14 Sep 2009) | 11 lines Recorded merge of revisions 218331 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218331 | tilghman | 2009-09-14 14:16:35 -0500 (Mon, 14 Sep 2009) | 4 lines Don't say "Please try again" if we don't give the user another chance to try again. (issue #15055, SWP-129) Reported by: jthurman ........ ................ 2009-09-14 18:17 +0000 [r218297] Joshua Colp * /, main/features.c: Merged revisions 218295 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r218295 | file | 2009-09-14 13:16:39 -0500 (Mon, 14 Sep 2009) | 2 lines Do not attempt to add a parking extension if an error occurred while reading the configuration. ........ 2009-09-14 15:17 +0000 [r218227] Matthew Nicholson * /, apps/app_directed_pickup.c: Merged revisions 218224 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r218224 | mnicholson | 2009-09-14 09:57:23 -0500 (Mon, 14 Sep 2009) | 14 lines Merged revisions 218223 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218223 | mnicholson | 2009-09-14 09:53:57 -0500 (Mon, 14 Sep 2009) | 8 lines Ensure we don't pickup ourselves when doing pickup by exten. (closes issue #15100) Reported by: lmsteffan Patches: (modified) pickup.patch uploaded by lmsteffan (license 779) ........ ................ 2009-09-13 21:48 +0000 [r218218] Tzafrir Cohen * channels/chan_phone.c, /: gcc 4.4: Remove a nop memset size 0 that annoys gcc This memset doesn't write beyond the end of the buffer. (tmpbuf has size of 4). Merged revisions 218184 via svnmerge from http://svn.digium.com/svn/asterisk/trunk 2009-09-12 13:15 +0000 [r218112] Michiel van Baak * main/rtp.c: Use the ip for the new 'rtp set debug ip '. Since 1.6.X still has the deprecated 'rtp debug ip ' this patch is different from the fix that went into trunk (closes issue 0015711) Reported by: davidw Patches: 2009082800-rtpdebug.diff.txt uploaded by mvanbaak (license 7) Tested by: davidw 2009-09-11 05:59 +0000 [r217924-218054] Tilghman Lesher * main/pbx.c, /: Merged revisions 218050 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r218050 | tilghman | 2009-09-11 00:58:11 -0500 (Fri, 11 Sep 2009) | 3 lines Check the origination priority for more matches, not the current priority. Found by Pavel Troller on the -dev list. ........ * apps/app_queue.c, /: Merged revisions 217990 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r217990 | tilghman | 2009-09-10 18:54:51 -0500 (Thu, 10 Sep 2009) | 10 lines Merged revisions 217989 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r217989 | tilghman | 2009-09-10 18:52:22 -0500 (Thu, 10 Sep 2009) | 3 lines Don't ring another channel, if there's not enough time for a queue member to answer. (Fixes AST-228) ........ ................ * channels/chan_iax2.c, contrib/scripts/iax-friends.sql, /, channels/chan_sip.c: Merged revisions 217916 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r217916 | tilghman | 2009-09-10 18:12:16 -0500 (Thu, 10 Sep 2009) | 2 lines Make calltoken support work with realtime users and peers. ........ 2009-09-10 21:23 +0000 [r217826] David Vossel * channels/chan_iax2.c, /: Merged revisions 217807 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r217807 | dvossel | 2009-09-10 16:07:47 -0500 (Thu, 10 Sep 2009) | 28 lines Merged revisions 217806 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r217806 | dvossel | 2009-09-10 16:06:07 -0500 (Thu, 10 Sep 2009) | 22 lines IAX2 encryption regression The IAX2 Call Token security patch inadvertently broke the use of encryption due to the reorganization of code in the socket_process() function. When encryption is used, an incoming full frame must first be decrypted before the information elements can be parsed. The security release mistakenly moved IE parsing before decryption in order to process the new Call Token IE. To resolve this, decryption of full frames is once again done before looking into the frame. This involves searching for an existing callno, checking the pvt to see if encryption is turned on, and decrypting the packet before the internal fields of the full frame are accessed. (closes issue #15834) Reported by: karesmakro Patches: iax2_encryption_fix_1.4.diff uploaded by dvossel (license 671) Tested by: dvossel, karesmakro Review: https://reviewboard.asterisk.org/r/355/ ........ ................ 2009-09-10 19:55 +0000 [r217738] mnick : * /, res/res_musiconhold.c: Merged revisions 217730 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r217730 | mnick | 2009-09-10 14:39:41 -0500 (Thu, 10 Sep 2009) | 17 lines Sets the correct musicclass after an announcement (closes issue #15279) Reported by: mbeckwell Patches: patch.txt uploaded by mnick (license ) Tested by: mnick (closes issue #15832) Reported by: mbeckwell Patches: patch.txt uploaded by mnick (license 874) Tested by: mnick ........ 2009-09-10 18:18 +0000 [r217642] Tilghman Lesher * res/res_config_odbc.c, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 217638 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r217638 | tilghman | 2009-09-10 13:17:14 -0500 (Thu, 10 Sep 2009) | 4 lines Verify support for wide ODBC character types before using them. (closes issue #15870) Reported by: nic_bellamy ........ 2009-09-10 12:11 +0000 [r217595] Olle Johansson * /, channels/chan_sip.c: Merged revisions 217593 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r217593 | oej | 2009-09-10 14:06:55 +0200 (Tor, 10 Sep 2009) | 8 lines Include ActionID in all events that are responsed to AMI Action SIPShowRegistry (closes issue #15868) Reported by: nic_bellamy Patches: manager_SIPshowregistry_actionid.patch uploaded by nic bellamy (license 299) ........ 2009-09-09 20:30 +0000 [r217518] Tzafrir Cohen * /, res/res_phoneprov.c: gcc 4.4 fix: union instead of cast gcc 4.4 has more strict rules for aliasing. It doesn't like a struct sockaddr_in pointer pointing to a struct sockaddr. So we make it a union. Merged revisions 217445 via svnmerge from http://svn.digium.com/svn/asterisk/trunk 2009-09-09 11:02 +0000 [r217370] Olle Johansson * /, channels/chan_sip.c: Merged revisions 217368 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r217368 | oej | 2009-09-09 12:39:43 +0200 (Ons, 09 Sep 2009) | 2 lines Not having any TLS session to write to is a serious XMIT_ERROR. ........ 2009-09-08 22:20 +0000 [r217295] Sean Bright * /, apps/app_meetme.c: Merged revisions 217286 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r217286 | seanbright | 2009-09-08 18:17:08 -0400 (Tue, 08 Sep 2009) | 4 lines Fix compilation of app_meetme. Reported by ebroad in #asterisk-bugs ........ 2009-09-08 20:32 +0000 [r217213] Tilghman Lesher * /, apps/app_meetme.c: Merged revisions 217199 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r217199 | tilghman | 2009-09-08 15:28:41 -0500 (Tue, 08 Sep 2009) | 14 lines Merged revisions 217156 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r217156 | tilghman | 2009-09-08 15:01:45 -0500 (Tue, 08 Sep 2009) | 7 lines When MOH is playing on the channel, announcements sent through the conference are not heard. (closes issue #14588) Reported by: voipas Patches: 20090716__issue14588__2.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen, twisted, tilghman ........ ................ 2009-09-08 16:39 +0000 [r217076] Kevin P. Fleming * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 217074 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r217074 | kpfleming | 2009-09-08 11:37:28 -0500 (Tue, 08 Sep 2009) | 9 lines Ensure that the default autoconf CFLAGS are not used. A recent change to the configure script that allows the user to specify CFLAGS and/or LDFLAGS to the script had the unfortunate side effect of letting autoconf's default CFLAGS (-g -O2) feed in to the rest of the build system, thereby overriding the DONT_OPTIMIZE setting in menuselect. That problem is now corrected. ........ 2009-09-08 15:36 +0000 [r217035] Tilghman Lesher * /, res/res_limit.c: Merged revisions 217033 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r217033 | tilghman | 2009-09-08 10:30:18 -0500 (Tue, 08 Sep 2009) | 4 lines Remove what appears to be an unnecessary define. (closes issue #15851) Reported by: tzafrir ........ 2009-09-08 14:27 +0000 [r216995] David Vossel * /, channels/chan_sip.c: Merged revisions 216993 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216993 | dvossel | 2009-09-08 09:26:30 -0500 (Tue, 08 Sep 2009) | 14 lines caller id number empty parse_uri was not being given the correct scheme's, as a result, uri parsing did not parse the username correctly. One of the side effects of this is an empty caller id. (closes issue #15839) Reported by: ebroad Patches: blank_cidv2.patch uploaded by ebroad (license 878) parse_uri_fix.diff uploaded by dvossel (license 671) Tested by: ebroad, dvossel ........ 2009-09-07 16:41 +0000 [r216646-216844] Olle Johansson * /, channels/chan_sip.c: Merged revisions 216842 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216842 | oej | 2009-09-07 18:35:12 +0200 (MÃ¥n, 07 Sep 2009) | 2 lines Make sure we reset global_exclude_static at channel reload ........ * /, channels/chan_sip.c: Merged revisions 216695 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216695 | oej | 2009-09-07 15:06:19 +0200 (MÃ¥n, 07 Sep 2009) | 8 lines If there is no session timer setting in the INVITE, set it to default value (not unset minimum = -1) Patch by oej closes issue #15621 Reported by: fnordian Tested by: atis ........ * configs/sip.conf.sample: Make code and documentation agree with each other * CHANGES, channels/chan_sip.c: Turning off premature media by default * apps/app_playback.c, main/pbx.c, /, channels/chan_sip.c, apps/app_disa.c, configs/sip.conf.sample: Merged revisions 216438 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r216438 | oej | 2009-09-04 16:02:34 +0200 (Fre, 04 Sep 2009) | 35 lines Merged revisions 216430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 lines Make apps send PROGRESS control frame for early media and fix too early media issue in SIP The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI links *before* any call progress. The SIP channel receives these frames and by default signals 183 Session progress and starts sending media. This will cause phones to play silence and ignore the later 180 ringing message. A bad user experience. The fix is twofold: - We discovered that asterisk apps that support early media ("noanswer") did not send any PROGRESS frame to indicate early media. Fixed. - We introduce a setting in chan_sip so that users can disable any relay of media frames before the outbound channel actually indicates any sort of call progress. In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions of Asterisk, this will be enabled. We don't assume that it will change your Asterisk phone experience - only for the better. We encourage third-party application developers to make sure that if they have applications that wants to send early media, add a PROGRESS control frame transmission to make sure that all channel drivers actually will start sending early media. This has not been the default in Asterisk previous to this patch, so if you got inspiration from our code, you need to update accordingly. Sorry for the trouble and thanks for your support. This code has been running for a few months in a large scale installation (over 250 servers with PRI and/or BRI links to old PBX systems). That's no proof that this is an excellent patch, but, well, it's tested :-) ........ ................ 2009-09-04 19:51 +0000 [r216599] David Vossel * /, channels/chan_sip.c: Merged revisions 216594 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216594 | dvossel | 2009-09-04 14:32:07 -0500 (Fri, 04 Sep 2009) | 7 lines sip peer matching by address only with TCP/TLS This patch removes the contact header matching logic and adds logic to match all tcp/tls connections by ip only Review: https://reviewboard.asterisk.org/r/354/ ........ 2009-09-04 19:32 +0000 [r216596] Sean Bright * apps/app_voicemail.c, /: Merged revisions 216593 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216593 | seanbright | 2009-09-04 15:29:02 -0400 (Fri, 04 Sep 2009) | 1 line Use ast_free() instead of free(). ........ 2009-09-04 17:53 +0000 [r216549-216552] Tilghman Lesher * /, include/asterisk/lock.h: Merged revisions 216551 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216551 | tilghman | 2009-09-04 12:50:21 -0500 (Fri, 04 Sep 2009) | 2 lines Fix trunk breakage. ........ * main/pbx.c, /, UPGRADE-1.6.txt: Merged revisions 216547 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216547 | tilghman | 2009-09-04 12:31:44 -0500 (Fri, 04 Sep 2009) | 3 lines Enable turning off the application delimiter warning with the 'dontwarn' option. Suggested on the -dev list, and implemented in an alternate way by me. ........ 2009-09-04 15:09 +0000 [r216440-216508] Michiel van Baak * /, main/utils.c: Merged revisions 216506 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r216506 | mvanbaak | 2009-09-04 17:05:05 +0200 (Fri, 04 Sep 2009) | 9 lines Merged revisions 216435 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r216435 | mvanbaak | 2009-09-04 15:56:10 +0200 (Fri, 04 Sep 2009) | 2 lines make asterisk compile under devmode with DEBUG_THREADS enabled on OpenBSD ........ ................ * /, include/asterisk/lock.h: Merged revisions 216437 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216437 | mvanbaak | 2009-09-04 16:00:38 +0200 (Fri, 04 Sep 2009) | 2 lines make sure canlog is set so we can compile with DEBUG_THREADS enabled on OpenBSD ........ 2009-09-04 13:56 +0000 [r216266-216434] Russell Bryant * /, channels/chan_sip.c: Merged revisions 216368 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216368 | russell | 2009-09-04 08:14:25 -0500 (Fri, 04 Sep 2009) | 12 lines Do not treat every SIP peer as if they were configured with insecure=port. There was a problem in the function responsible for doing peer matching by IP address and port number such that during the second pass for checking for a peer configured with insecure=port, it would end up treating every peer as if it had been configured that way. These changes fix the logic in the peer IP and port comparison callback to handle insecure=port checking properly. This problem was introduced when SIP peers were converted to astobj2. Many thanks to dvossel for noticing this while working on another peer matching issue. ........ * doc/IAX2-security.txt (added), /: Merged revisions 216264 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r216264 | russell | 2009-09-04 05:48:44 -0500 (Fri, 04 Sep 2009) | 16 lines Merged revisions 216263 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r216263 | russell | 2009-09-04 05:48:00 -0500 (Fri, 04 Sep 2009) | 9 lines Merged revisions 216262 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r216262 | russell | 2009-09-04 05:47:37 -0500 (Fri, 04 Sep 2009) | 2 lines Add a plain text version of the IAX2 security document. ........ ................ ................ 2009-09-04 06:13 +0000 [r216224] Michiel van Baak * main/astobj2.c, /: Merged revisions 216222 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r216222 | mvanbaak | 2009-09-04 08:08:33 +0200 (Fri, 04 Sep 2009) | 3 lines make sure 'start' is always initialized. Makes asterisk compile with --enable-dev-mode ........ 2009-09-03 19:42 +0000 [r216013-216098] Russell Bryant * UPGRADE.txt: tweak * /, UPGRADE.txt: Merged revisions 216092 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r216092 | russell | 2009-09-03 14:38:35 -0500 (Thu, 03 Sep 2009) | 16 lines Merged revisions 216085 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r216085 | russell | 2009-09-03 14:36:46 -0500 (Thu, 03 Sep 2009) | 9 lines Merged revisions 216080 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r216080 | russell | 2009-09-03 14:35:23 -0500 (Thu, 03 Sep 2009) | 2 lines Add a note about IAX2 to UPGRADE.txt. ........ ................ ................ * /, doc/IAX2-security.pdf (added): Merged revisions 216009 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r216009 | russell | 2009-09-03 13:45:54 -0500 (Thu, 03 Sep 2009) | 16 lines Merged revisions 216008 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r216008 | russell | 2009-09-03 13:44:58 -0500 (Thu, 03 Sep 2009) | 9 lines Merged revisions 216005 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r216005 | russell | 2009-09-03 13:42:24 -0500 (Thu, 03 Sep 2009) | 2 lines Add IAX2 security document related to AST-2009-006. ........ ................ ................ 2009-09-03 18:41 +0000 [r216004] David Vossel * channels/chan_iax2.c, channels/iax2-parser.c, main/astobj2.c, configs/iax.conf.sample, include/asterisk/acl.h, channels/iax2-parser.h, /, include/asterisk/astobj2.h, channels/iax2.h, main/acl.c: Merged revisions 215955 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r215955 | dvossel | 2009-09-03 11:31:54 -0500 (Thu, 03 Sep 2009) | 6 lines Merge code associated with AST-2009-006 (closes issue #12912) Reported by: rathaus Tested by: tilghman, russell, dvossel, dbrooks ........ 2009-09-03 Leif Madsen * Asterisk 1.6.1.6 released * AST-2009-006 2009-08-28 Leif Madsen * Asterisk 1.6.1.5 released 2009-08-11 Tilghman Lesher * Asterisk 1.6.1.5-rc1 released 2009-08-10 19:51 +0000 [r211569-211586] Tilghman Lesher * doc/CODING-GUIDELINES, /: Merged revisions 211584 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r211584 | tilghman | 2009-08-10 14:49:41 -0500 (Mon, 10 Aug 2009) | 9 lines Merged revisions 211583 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r211583 | tilghman | 2009-08-10 14:48:48 -0500 (Mon, 10 Aug 2009) | 1 line Conversion specifiers, not format specifiers ........ ................ * channels/chan_iax2.c, res/ael/pval.c, main/cdr.c, main/channel.c, main/manager.c, apps/app_setcallerid.c, apps/app_rpt.c, main/asterisk.c, res/res_config_pgsql.c, apps/app_dahdibarge.c, funcs/func_rand.c, funcs/func_timeout.c, apps/app_record.c, codecs/codec_speex.c, apps/app_morsecode.c, main/acl.c, funcs/func_cut.c, cdr/cdr_pgsql.c, apps/app_followme.c, main/enum.c, res/res_config_sqlite.c, main/config.c, agi/eagi-sphinx-test.c, channels/misdn_config.c, channels/chan_dahdi.c, funcs/func_channel.c, apps/app_macro.c, apps/app_sms.c, pbx/pbx_config.c, apps/app_verbose.c, main/dsp.c, apps/app_voicemail.c, apps/app_adsiprog.c, funcs/func_speex.c, channels/chan_sip.c, res/res_limit.c, channels/chan_agent.c, agi/eagi-test.c, funcs/func_math.c, main/utils.c, channels/iax2-provision.c, apps/app_talkdetect.c, main/indications.c, channels/chan_oss.c, main/cli.c, res/res_config_curl.c, pbx/pbx_loopback.c, res/res_smdi.c, apps/app_osplookup.c, channels/chan_misdn.c, channels/chan_skinny.c, pbx/pbx_dundi.c, utils/extconf.c, apps/app_mixmonitor.c, channels/chan_mgcp.c, main/timing.c, main/pbx.c, doc/CODING-GUIDELINES, utils/muted.c, apps/app_readfile.c, /, apps/app_meetme.c, apps/app_privacy.c, apps/app_waituntil.c, cdr/cdr_adaptive_odbc.c, pbx/dundi-parser.c, res/res_http_post.c, res/res_musiconhold.c, apps/app_queue.c, main/netsock.c, utils/frame.c, channels/chan_usbradio.c, funcs/func_enum.c, channels/chan_phone.c, apps/app_waitforring.c, pbx/pbx_spool.c, funcs/func_odbc.c, apps/app_minivm.c, main/features.c, res/res_agi.c, main/http.c, res/snmp/agent.c, res/res_config_ldap.c, apps/app_chanspy.c, apps/app_stack.c, res/res_odbc.c, funcs/func_dialplan.c, main/dnsmgr.c, main/frame.c, apps/app_waitforsilence.c, funcs/func_strings.c, apps/app_disa.c, apps/app_alarmreceiver.c: AST-2009-005 2009-08-10 14:12 +0000 [r211349] Joshua Colp * /, channels/chan_sip.c: Merged revisions 211347 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r211347 | file | 2009-08-10 11:07:44 -0300 (Mon, 10 Aug 2009) | 5 lines Fix retrieval of the port used for the video stream when adding SDP to a SIP message. (closes issue #15121) Reported by: jsmith ........ 2009-08-09 15:43 +0000 [r211234-211277] Tilghman Lesher * /, main/astfd.c: Merged revisions 211275 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r211275 | tilghman | 2009-08-09 10:42:02 -0500 (Sun, 09 Aug 2009) | 9 lines Merged revisions 211274 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r211274 | tilghman | 2009-08-09 10:41:01 -0500 (Sun, 09 Aug 2009) | 2 lines Small oops. Clear the flags which have been checked. ........ ................ * apps/app_stack.c, /: Merged revisions 211232 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r211232 | tilghman | 2009-08-09 02:11:22 -0500 (Sun, 09 Aug 2009) | 4 lines Check for NULL frame, before dereferencing pointer. (closes issue #15617) Reported by: rain ........ 2009-08-07 20:17 +0000 [r211115] Russell Bryant * apps/app_chanspy.c, /: Merged revisions 211113 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r211113 | russell | 2009-08-07 15:12:21 -0500 (Fri, 07 Aug 2009) | 11 lines Recorded merge of revisions 211112 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r211112 | russell | 2009-08-07 15:11:31 -0500 (Fri, 07 Aug 2009) | 4 lines Resolve a deadlock involving app_chanspy and masquerades. (ABE-1936) ........ ................ 2009-08-07 18:19 +0000 [r211047] Tilghman Lesher * apps/app_queue.c, /: Merged revisions 211040 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r211040 | tilghman | 2009-08-07 13:17:41 -0500 (Fri, 07 Aug 2009) | 21 lines Merged revisions 211038 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r211038 | tilghman | 2009-08-07 13:16:28 -0500 (Fri, 07 Aug 2009) | 14 lines QUEUE_MEMBER_LIST _really_ wants the interface name, not the membername. This is a partial revert of revision 82590, which was an attempted cleanup, but in reality, it broke QUEUE_MEMBER_LIST, which has always been intended as a method by which component interfaces could be queried from the queue. Membername isn't useful here, because that field cannot be used to obtain further information about the member. See the documentation on QUEUE_MEMBER_LIST, RemoveQueueMember, QUEUE_MEMBER_PENALTY, and the various AMI commands which take a member argument for further justification. (closes issue #15664) Reported by: rain Patches: app_queue-queue_member_list.diff uploaded by rain (license 327) ........ ................ 2009-08-07 13:09 +0000 [r210994] Kevin P. Fleming * main/udptl.c, /: Merged revisions 210992 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r210992 | kpfleming | 2009-08-07 08:08:00 -0500 (Fri, 07 Aug 2009) | 13 lines Workaround broken T.38 endpoints that offer tiny MaxDatagram sizes. Some T.38 endpoints treat T38FaxMaxDatagram as the maximum IFP size that should be sent to them, rather than the maximum packet payload size. If such an endpoint also requests UDPRedundancy as the error correction mode, we'll end up calculating a tiny maximum IFP size, so small as to be unusable. This patch sets a lower bound on what we'll consider the remote's maximum IFP size to be, assuming that endpoints that do this really can accept larger packets than they've offered to accept. (closes issue #15649) Reported by: dazza76 ........ 2009-08-06 21:47 +0000 [r210910-210916] Tilghman Lesher * main/channel.c, /: Merged revisions 210914 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r210914 | tilghman | 2009-08-06 16:46:01 -0500 (Thu, 06 Aug 2009) | 14 lines Merged revisions 210913 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r210913 | tilghman | 2009-08-06 16:45:01 -0500 (Thu, 06 Aug 2009) | 7 lines Because channel information can be accessed outside of the channel thread, we must lock the channel prior to modifying it. (closes issue #15397) Reported by: caspy Patches: 20090714__issue15397.diff.txt uploaded by tilghman (license 14) Tested by: caspy ........ ................ * apps/app_stack.c, include/asterisk/app.h, /, main/app.c: Merged revisions 210908 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r210908 | tilghman | 2009-08-06 16:29:26 -0500 (Thu, 06 Aug 2009) | 9 lines Allow Gosub to recognize quote delimiters without consuming them. (closes issue #15557) Reported by: rain Patches: 20090723__issue15557.diff.txt uploaded by tilghman (license 14) Tested by: rain Review: https://reviewboard.asterisk.org/r/316/ ........ 2009-08-06 17:48 +0000 [r210819] Joshua Colp * /, channels/chan_sip.c: Merged revisions 210817 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r210817 | file | 2009-08-06 14:47:04 -0300 (Thu, 06 Aug 2009) | 11 lines Accept additional T.38 reinvites after an initial one has been handled. Discussion of this subject has yielded that it is not actually acceptable to change T.38 parameters after the initial reinvite but declining is harsh and can cause the fax to fail when it may be possible to allow it to continue. This patch changes things so that additional T.38 reinvites are accepted but parameter changes ignored. This gives the fax a fighting chance. (closes issue #15610) Reported by: huangtx2009 ........ 2009-08-05 20:28 +0000 [r210681] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 210640 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r210640 | rmudgett | 2009-08-05 14:40:03 -0500 (Wed, 05 Aug 2009) | 21 lines Merged revisions 210575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r210575 | rmudgett | 2009-08-05 14:18:56 -0500 (Wed, 05 Aug 2009) | 14 lines Dialplan starts execution before the channel setup is complete. * Issue 15655: For the case where dialing is complete for an incoming call, dahdi_new() was asked to start the PBX and then the code set more channel variables. If the dialplan hungup before these channel variables got set, asterisk would likely crash. * Fixed potential for overlap incoming call to erroneously set channel variables as global dialplan variables if the ast_channel structure failed to get allocated. * Added missing set of CALLINGSUBADDR in the dialing is complete case. (closes issue #15655) Reported by: alecdavis ........ ................ 2009-08-05 18:57 +0000 [r210567] Leif Madsen * doc/tex/imapstorage.tex, /: Merged revisions 210564 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r210564 | lmadsen | 2009-08-05 13:49:58 -0500 (Wed, 05 Aug 2009) | 19 lines Merged revisions 210563 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r210563 | lmadsen | 2009-08-05 13:46:21 -0500 (Wed, 05 Aug 2009) | 11 lines Update imapstorage.txt documentation. Updated the imapstorage.txt documentation to reflect that issues with c-client versions older than 2007 seem to cause crashing issues that are not seen with more recent versions. Documentation has been updated to reflect this. (closes issue #14496) Reported by: vbcrlfuser Patches: __20090727-imap-documentation-patch.txt uploaded by lmadsen (license 10) Tested by: lmadsen, mmichelson, dbrooks ........ ................ 2009-08-04 14:54 +0000 [r210240] Kevin P. Fleming * Makefile, /: Merged revisions 210238 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r210238 | kpfleming | 2009-08-04 09:53:00 -0500 (Tue, 04 Aug 2009) | 16 lines Merged revisions 210237 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r210237 | kpfleming | 2009-08-04 09:51:39 -0500 (Tue, 04 Aug 2009) | 10 lines Eliminate spurious compiler warnings from system headers on *BSD platforms. Ensure that system headers located in /usr/local/include are actually treated as system headers by the compiler, and not as local headers which are subject to warnings from the -Wundef compiler option and others. (closes issue #15606) Reported by: mvanbaak ........ ................ 2009-08-01 11:32 +0000 [r209836-209900] Russell Bryant * main/db1-ast/mpool/mpool.c, /: Merged revisions 209887 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r209887 | russell | 2009-08-01 06:29:25 -0500 (Sat, 01 Aug 2009) | 12 lines Merged revisions 209879 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r209879 | russell | 2009-08-01 06:27:25 -0500 (Sat, 01 Aug 2009) | 5 lines Resolve a valgrind warning about a read from uninitialized memory. (issue #15396) Reported by: aragon ........ ................ * apps/app_milliwatt.c, /: Merged revisions 209839 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r209839 | russell | 2009-08-01 06:02:07 -0500 (Sat, 01 Aug 2009) | 20 lines Merged revisions 209838 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r209838 | russell | 2009-08-01 05:59:05 -0500 (Sat, 01 Aug 2009) | 13 lines Modify how Playtones() is used in Milliwatt() to resolve gain issue. When Milliwatt() was changed internally to use Playtones() so that the proper tone was used, it introduced a drop in gain in the output signal. So, use the playtones API directly and specify a volume argument such that the output matches the gain of the original Milliwatt() code. (closes issue #15386) Reported by: rue_mohr Patches: issue_15386.rev2.diff uploaded by russell (license 2) Tested by: rue_mohr ........ ................ * /, main/event.c: Merged revisions 209835 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209835 | russell | 2009-08-01 05:43:40 -0500 (Sat, 01 Aug 2009) | 6 lines Fix ast_event_queue_and_cache() to actually do the cache() part. (closes issue #15624) Reported by: ffossard Tested by: russell ........ 2009-08-01 01:25 +0000 [r209781] Kevin P. Fleming * channels/misdn/isdn_lib.c, utils/frame.c, /, main/Makefile, channels/misdn/ie.c: Merged revisions 209760-209761 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r209760 | kpfleming | 2009-07-31 20:03:07 -0500 (Fri, 31 Jul 2009) | 13 lines Merged revisions 209759 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r209759 | kpfleming | 2009-07-31 19:52:00 -0500 (Fri, 31 Jul 2009) | 7 lines Minor changes inspired by testing with latest GCC. The latest GCC (what will become 4.5.x) has a few new warnings, that in these cases found some either downright buggy code, or at least seriously poorly designed code that could be improved. ........ ................ r209761 | kpfleming | 2009-07-31 20:04:06 -0500 (Fri, 31 Jul 2009) | 1 line Revert accidental Makefile change. ................ 2009-07-31 21:58 +0000 [r209714] Russell Bryant * /, main/event.c: Merged revisions 209711 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209711 | russell | 2009-07-31 16:53:31 -0500 (Fri, 31 Jul 2009) | 2 lines Fix some places where ast_event_type was used instead of ast_event_ie_type. ........ 2009-07-30 18:46 +0000 [r209593] David Brooks * include/asterisk/abstract_jb.h, channels/chan_dahdi.c, contrib/init.d/rc.debian.asterisk, /, apps/app_sms.c, codecs/lpc10/pitsyn.c, channels/chan_console.c: Merged revisions 209554 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209554 | dbrooks | 2009-07-30 11:07:05 -0500 (Thu, 30 Jul 2009) | 6 lines Fixes numerous spelling errors. Patch submitted by alecdavis. (closes issue #15595) Reported by: alecdavis ........ 2009-07-30 14:40 +0000 [r209517] Mark Michelson * /, channels/chan_sip.c: Merged revisions 209516 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209516 | mmichelson | 2009-07-30 09:38:21 -0500 (Thu, 30 Jul 2009) | 8 lines Fix a crash that can result if text codecs are allowed but textsupport is disabled. (closes issue #15596) Reported by: fabled Patches: sip-red.patch uploaded by fabled (license 448) ........ 2009-07-28 00:19 +0000 [r209327] Tilghman Lesher * /, sounds/sounds.xml: Merged revisions 209317 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r209317 | tilghman | 2009-07-27 19:14:12 -0500 (Mon, 27 Jul 2009) | 9 lines Merged revisions 209315 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r209315 | tilghman | 2009-07-27 19:12:03 -0500 (Mon, 27 Jul 2009) | 2 lines Publish French extra sounds ........ ................ 2009-07-27 21:44 +0000 [r209262-209281] Kevin P. Fleming * /, apps/app_fax.c: Merged revisions 209279 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209279 | kpfleming | 2009-07-27 16:43:36 -0500 (Mon, 27 Jul 2009) | 7 lines Cleanup T.38 negotiation changes. Convert LOG_NOTICE messages about T.38 negotiation in debug level 1 messages, clean up some looping logic, and correct an improper use of ast_free() for freeing an ast_frame. ........ * /, apps/app_fax.c: Merged revisions 209256 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209256 | kpfleming | 2009-07-27 16:21:43 -0500 (Mon, 27 Jul 2009) | 10 lines Make T.38 switchover in ReceiveFAX synchronous. In receive mode, if the channel that ReceiveFAX is running on supports T.38, we should *always* attempt to switch T.38, rather than listening for an incoming CNG tone and only triggering on that. The channel may be using a low-bitrate codec that distorts the CNG tone, the sending FAX endpoint may not send CNG at all, or there could be a variety of other reasons that we don't detect it, but in all those cases if T.38 is available we certainly want to use it. ........ 2009-07-27 20:57 +0000 [r209237] Mark Michelson * main/rtp.c, /: Merged revisions 209235 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209235 | mmichelson | 2009-07-27 15:54:54 -0500 (Mon, 27 Jul 2009) | 5 lines Gracefully handle malformed RTP text packets. AST-2009-004 ........ 2009-07-27 20:28 +0000 [r209233] David Brooks * res/res_jabber.c, main/loader.c, channels/chan_dahdi.c, channels/chan_vpb.cc, res/res_smdi.c, /, include/asterisk/module.h, main/features.c, res/res_agi.c: Merged revisions 209098 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209098 | dbrooks | 2009-07-27 11:33:50 -0500 (Mon, 27 Jul 2009) | 6 lines Fixing typos. Replaces "recieved" with "received" and "initilize" with "initialize" (closes issue #15571) Reported by: alecdavis ........ 2009-07-27 20:17 +0000 [r209134-209199] Mark Michelson * /, res/res_musiconhold.c: Merged revisions 209197 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209197 | mmichelson | 2009-07-27 15:11:42 -0500 (Mon, 27 Jul 2009) | 9 lines Honor channel's music class when using realtime music on hold. (closes issue #15051) Reported by: alexh Patches: 15051.patch uploaded by mmichelson (license 60) Tested by: alexh ........ * main/udptl.c, /, configs/udptl.conf.sample: Merged revisions 209132 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r209132 | mmichelson | 2009-07-27 12:50:04 -0500 (Mon, 27 Jul 2009) | 24 lines Merged revisions 209131 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r209131 | mmichelson | 2009-07-27 12:44:06 -0500 (Mon, 27 Jul 2009) | 18 lines Allow for UDPTL to use only even-numbered ports if desired. There are some VoIP providers out there that will not accept SDP offers with odd numbered UDPTL ports. While it is my personal opinion that these VoIP providers are misinterpreting RFC 2327, it really is not a big deal to play along with their silly little games. Of course, since restricting UDPTL ports to only even numbers reduces the range of available ports by half, so the option to use only even port numbers is off by default. A user can enable the behavior by setting use_even_ports=yes in udptl.conf. (closes issue #15182) Reported by: CGMChris Patches: 15182.patch uploaded by mmichelson (license 60) Tested by: CGMChris ........ ................ 2009-07-27 15:40 +0000 [r209058] Kevin P. Fleming * Makefile, /: Merged revisions 209056 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209056 | kpfleming | 2009-07-27 10:38:59 -0500 (Mon, 27 Jul 2009) | 10 lines Restore explicit export of ASTCFLAGS/ASTLDFLAGS and underscore-variants to sub-makes. During the recent Makefile improvements I made, it seemed the 'make' was automatically carrying down the ASTCFLAGS/ASTLDFLAGS settings to sub-makes, so I removed the explict export of them. However, there are some circumstances where make does this, and some where it does not, so I've brought them back to ensure they are always exported. I also removed an extraneous double setting of _ASTLDFLAGS on *BSD platforms. ........ 2009-07-27 01:22 +0000 [r208926] Jeff Peeler * channels/chan_iax2.c, /, main/translate.c: Merged revisions 208924 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208924 | jpeeler | 2009-07-26 20:20:37 -0500 (Sun, 26 Jul 2009) | 9 lines Merged revisions 208923 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208923 | jpeeler | 2009-07-26 20:18:31 -0500 (Sun, 26 Jul 2009) | 2 lines Fix logic errors from 208746 ........ ................ 2009-07-26 14:04 +0000 [r208888] Michiel van Baak * contrib/scripts/install_prereq, /: Merged revisions 208886 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208886 | mvanbaak | 2009-07-26 16:00:52 +0200 (Sun, 26 Jul 2009) | 2 lines add OpenBSD to the install_prereq script ........ 2009-07-25 06:25 +0000 [r208754] Jeff Peeler * channels/chan_iax2.c, /, channels/chan_skinny.c, main/translate.c: Merged revisions 208749 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208749 | jpeeler | 2009-07-25 01:23:18 -0500 (Sat, 25 Jul 2009) | 13 lines Merged revisions 208746 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208746 | jpeeler | 2009-07-25 01:19:50 -0500 (Sat, 25 Jul 2009) | 7 lines Fix compiling under dev-mode with gcc 4.4.0. Mostly trivial changes, but I did not know of any other way to fix the "dereferencing type-punned pointer will break strict-aliasing rules" error without creating a tmp variable in chan_skinny. ........ ................ 2009-07-24 18:52 +0000 [r208595] Russell Bryant * apps/app_dial.c, /: Merged revisions 208593 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208593 | russell | 2009-07-24 13:42:32 -0500 (Fri, 24 Jul 2009) | 14 lines Merged revisions 208592 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208592 | russell | 2009-07-24 13:38:24 -0500 (Fri, 24 Jul 2009) | 7 lines Do not log an ERROR if autoservice_stop() returns -1. This does not indicate an error. A return of -1 just means that the channel has been hung up. (reported in #asterisk-dev) ........ ................ 2009-07-24 18:32 +0000 [r208590] Mark Michelson * /, channels/chan_sip.c: Merged revisions 208588 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208588 | mmichelson | 2009-07-24 13:31:04 -0500 (Fri, 24 Jul 2009) | 16 lines Merged revisions 208587 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208587 | mmichelson | 2009-07-24 13:26:50 -0500 (Fri, 24 Jul 2009) | 10 lines Only send a BYE when hanging up a channel that is up. For cases where Asterisk sends an INVITE and receives a non 2XX final response, Asterisk would follow the INVITE transaction by immediately sending a BYE, which was unnecessary. (closes issue #14575) Reported by: chris-mac ........ ................ 2009-07-24 15:05 +0000 [r208550] Kevin P. Fleming * main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h: Merged revisions 208548 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208548 | kpfleming | 2009-07-24 10:02:53 -0500 (Fri, 24 Jul 2009) | 8 lines Resolve a T.38 negotiation issue left over from the udptl-updates merge. The udptl-updates branch that was merged yesterday failed to properly send back T.38 SDP responses with the correct error correction mode, if the incoming SDP from the other end caused us to change error correction modes. This patch corrects that situation. ........ 2009-07-24 14:38 +0000 [r208544] Michiel van Baak * contrib/scripts/install_prereq, /: Merged revisions 208542 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208542 | mvanbaak | 2009-07-24 16:35:49 +0200 (Fri, 24 Jul 2009) | 13 lines use aptitude for debian based systems The function to check wether we need to install packages was using dpkg-query which was gives wrong output on Debian 5 Also, the apt-get has been replaced with aptitude because aptitude is now the preferred way to handle packages on Debian (closes issue #15570) Reported by: mvanbaak Patches: 2009072400_installprereq-aptitude.diff uploaded by mvanbaak (license 7) ........ 2009-07-23 22:32 +0000 [r208484-208503] Kevin P. Fleming * UPGRADE.txt: Use correct formatting for T.38 change note in UPGRADE.txt * include/asterisk/frame.h, main/rtp.c, main/channel.c, main/udptl.c, main/frame.c, /, channels/chan_sip.c, apps/app_fax.c, UPGRADE.txt, include/asterisk/udptl.h: Merged revisions 208464 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208464 | kpfleming | 2009-07-23 16:57:24 -0500 (Thu, 23 Jul 2009) | 46 lines Rework of T.38 negotiation and UDPTL API to address interoperability problems Over the past couple of months, a number of issues with Asterisk negotiating (and successfully completing) T.38 sessions with various endpoints have been found. This patch attempts to address many of them, primarily focused around ensuring that the endpoints' MaxDatagram size is honored, and in addition by ensuring that T.38 session parameter negotiation is performed correctly according to the ITU T.38 Recommendation. The major changes here are: 1) T.38 applications in Asterisk (app_fax) only generate/receive IFP packets, they do not ever work with UDPTL packets. As a result of this, they cannot be allowed to generate packets that would overflow the other endpoints' MaxDatagram size after the UDPTL stack adds any error correction information. With this patch, the application is told the maximum *IFP* size it can generate, based on a calculation using the far end MaxDatagram size and the active error correction mode on the T.38 session. The same is true for sending *our* MaxDatagram size to the remote endpoint; it is computed from the value that the application says it can accept (for a single IFP packet) combined with the active error correction mode. 2) All treatment of T.38 session parameters as 'capabilities' in chan_sip has been removed; these parameters are not at all like audio/video stream capabilities. There are strict rules to follow for computing an answer to a T.38 offer, and chan_sip now follows those rules, using the desired parameters from the application (or channel) that wants to accept the T.38 negotiation. 3) chan_sip now stores and forwards ast_control_t38_parameters structures for tracking 'our' and 'their' T.38 session parameters; this greatly simplifies negotiation, especially for pass-through calls. 4) Since T.38 negotiation without specifying parameters or receiving the final negotiated parameters is not very worthwhile, the AST_CONTROL_T38 control frame has been removed. A note has been added to UPGRADE.txt about this removal, since any out-of-tree applications that use it will no longer function properly until they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review: https://reviewboard.asterisk.org/r/310/ ........ 2009-07-23 20:45 +0000 [r208459] David Brooks * apps/app_rpt.c, res/res_smdi.c, pbx/pbx_dundi.c: Just replacing typos "recieved" with "received". (closes issue #15360) Reported by: okrief 2009-07-23 19:35 +0000 [r208390] Mark Michelson * /, channels/chan_sip.c: Merged revisions 208388 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208388 | mmichelson | 2009-07-23 14:34:49 -0500 (Thu, 23 Jul 2009) | 24 lines Merged revisions 208386 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208386 | mmichelson | 2009-07-23 14:24:21 -0500 (Thu, 23 Jul 2009) | 17 lines Fix a problem where a 491 response could be sent out of dialog. This generalizes the fix for issue 13849. The initial fix corrected the problem that Asterisk would reply with a 491 if a reinvite were received from an endpoint and we had not yet received an ACK from that endpoint for the initial INVITE it had sent us. This expansion also allows Asterisk to appropriately handle an INVITE with authorization credentials if Asterisk had not received an ACK from the previous transaction in which Asterisk had responded to an unauthorized INVITE with a 407. (closes issue #14239) Reported by: klaus3000 Patches: 14239.patch uploaded by mmichelson (license 60) Tested by: klaus3000 ........ ................ 2009-07-23 19:24 +0000 [r208385] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 208383 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208383 | jpeeler | 2009-07-23 14:21:50 -0500 (Thu, 23 Jul 2009) | 12 lines Merged revisions 208380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208380 | jpeeler | 2009-07-23 14:19:53 -0500 (Thu, 23 Jul 2009) | 6 lines Only set the priindication setting when not performing a reload (closes issue #14696) Reported by: fdecher ........ ................ 2009-07-23 16:30 +0000 [r208265-208318] Mark Michelson * /, channels/chan_sip.c: Merged revisions 208314 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208314 | mmichelson | 2009-07-23 11:29:37 -0500 (Thu, 23 Jul 2009) | 9 lines Merged revisions 208312 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208312 | mmichelson | 2009-07-23 11:29:18 -0500 (Thu, 23 Jul 2009) | 3 lines Remove inaccurate XXX comment. ........ ................ * /, channels/chan_sip.c: Merged revisions 208263 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208263 | mmichelson | 2009-07-23 10:46:34 -0500 (Thu, 23 Jul 2009) | 15 lines Merged revisions 208262 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208262 | mmichelson | 2009-07-23 10:43:07 -0500 (Thu, 23 Jul 2009) | 8 lines Properly handle 183 responses which do not contain an SDP. (closes issue #15442) Reported by: ffloimair Patches: 15442.patch uploaded by mmichelson (license 60) Tested by: tkarl, ffloimair ........ ................ 2009-07-22 21:45 +0000 [r208115] Jason Parker * /, apps/app_festival.c: Merged revisions 208113 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208113 | qwell | 2009-07-22 16:43:57 -0500 (Wed, 22 Jul 2009) | 9 lines Restore an int declaration on PPC platforms. This x is one crafty little bugger... It was used for 2 different things (one of which was only done on PPC) in 1.4. One of the uses were removed in trunk, and with it went the declaration. (closes issue #14038) Reported by: ffloimair ........ 2009-07-21 22:48 +0000 [r207948] Tilghman Lesher * /, funcs/func_strings.c: Merged revisions 207946 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207946 | tilghman | 2009-07-21 17:45:32 -0500 (Tue, 21 Jul 2009) | 15 lines Merged revisions 207945 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207945 | tilghman | 2009-07-21 17:38:54 -0500 (Tue, 21 Jul 2009) | 8 lines Force an error if a blank is passed to QUOTE (because the documentation states the argument is not optional). This change makes URIENCODE and QUOTE behave similarly, since the documentation states that the argument is not optional, for both. (closes issue #15439) Reported by: pkempgen Patches: 20090706__issue15439.diff.txt uploaded by tilghman (license 14) ........ ................ 2009-07-21 20:29 +0000 [r207784-207861] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 207854 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207854 | jpeeler | 2009-07-21 15:26:02 -0500 (Tue, 21 Jul 2009) | 16 lines Merged revisions 207827 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207827 | jpeeler | 2009-07-21 15:16:55 -0500 (Tue, 21 Jul 2009) | 9 lines Wait for wink before dialing when using E&M wink signaling There was already code for other signaling types in dahdi_handle_event to handle dialing if a dial operation dial string was present. Simply add SIG_EMWINK to the list. (closes issue #14434) Reported by: araasch ........ ................ * channels/chan_dahdi.c: Revert r207637, this approach could potentially block for an unacceptable amount of time. 2009-07-21 14:31 +0000 [r207726] Mark Michelson * main/manager.c, /: Merged revisions 207723 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207723 | mmichelson | 2009-07-21 09:29:40 -0500 (Tue, 21 Jul 2009) | 11 lines Merged revisions 207714 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207714 | mmichelson | 2009-07-21 09:26:00 -0500 (Tue, 21 Jul 2009) | 5 lines Document default timeout for AMI originations. AST-224 ........ ................ 2009-07-21 13:48 +0000 [r207684] Kevin P. Fleming * channels/Makefile, doc/video_console.txt, Makefile, agi/Makefile, codecs/Makefile, utils/Makefile, funcs/Makefile, codecs/lpc10/Makefile, main/db1-ast/Makefile, /, main/Makefile, codecs/gsm/Makefile, Makefile.moddir_rules, Makefile.rules, pbx/Makefile, res/Makefile: Merged revisions 207680 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207680 | kpfleming | 2009-07-21 08:28:04 -0500 (Tue, 21 Jul 2009) | 18 lines Merged revisions 207647 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul 2009) | 12 lines Ensure that user-provided CFLAGS and LDFLAGS are honored. This commit changes the build system so that user-provided flags (in ASTCFLAGS and ASTLDFLAGS) are supplied to the compiler/linker *after* all flags provided by the build system itself, so that the user can effectively override the build system's flags if desired. In addition, ASTCFLAGS and ASTLDFLAGS can now be provided *either* in the environment before running 'make', or as variable assignments on the 'make' command line. As a result, the use of COPTS and LDOPTS is no longer necessary, so they are no longer documented, but are still supported so as not to break existing build systems that supply them when building Asterisk. ........ ................ 2009-07-21 04:45 +0000 [r207637] Jeff Peeler * channels/chan_dahdi.c: Wait for wink before dialing when using E&M wink signaling This patch adds a new dahdi_wait function to specifically wait for the wink event. If the wink is not eventually received the channel is hung up. (closes issue #14434) Reported by: araasch Patches: emwinkmod uploaded by araasch (license 693) 2009-07-20 20:02 +0000 [r207426] Mark Michelson * /, channels/chan_sip.c: Merged revisions 207424 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207424 | mmichelson | 2009-07-20 14:48:12 -0500 (Mon, 20 Jul 2009) | 39 lines Merged revisions 207423 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207423 | mmichelson | 2009-07-20 14:39:59 -0500 (Mon, 20 Jul 2009) | 33 lines Answer video SDP offers properly when videosupport is not enabled. Copied from Review board: In issue 12434, the reporter describes a situation in which audio and video is offered on the call, but because videosupport is disabled in sip.conf, Asterisk gives no response at all to the video offer. According to RFC 3264, all media offers should have a corresponding answer. For offers we do not intend to actually reply to with meaningful values, we should still reply with the port for the media stream set to 0. In this patch, we take note of what types of media have been offered and save the information on the sip_pvt. The SDP in the response will take into account whether media was offered. If we are not otherwise going to answer a media offer, we will insert an appropriate m= line with the port set to 0. It is important to note that this patch is pretty much a bandage being applied to a broken bone. The patch *only* helps for situations where video is offered but videosupport is disabled and when udptl_pt is disabled but T.38 is offered. Asterisk is not guaranteed to respond to every media offer. Notable cases are when multiple streams of the same type are offered. The 2 media stream limit is still present with this patch, too. In trunk and the 1.6.X branches, things will be a bit different since Asterisk also supports text in SDPs as well. (closes issue #12434) Reported by: mnnojd Review: https://reviewboard.asterisk.org/r/311 Review: https://reviewboard.asterisk.org/r/313 ........ ................ 2009-07-20 16:40 +0000 [r207363] Russell Bryant * main/channel.c, /: Merged revisions 207361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207361 | russell | 2009-07-20 11:36:15 -0500 (Mon, 20 Jul 2009) | 16 lines Merged revisions 207360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207360 | russell | 2009-07-20 11:26:24 -0500 (Mon, 20 Jul 2009) | 9 lines Only do the chan->fdno check in ast_read() in a developer build. I changed this check to only happen in a dev-mode build. I also added a comment explaining what is going on. I also made it so that detection of this situation does not affect ast_read() operation. (closes issue #14723) Reported by: seadweller ........ ................ 2009-07-18 04:17 +0000 [r207321] Tilghman Lesher * apps/app_voicemail.c, /: Recorded merge of revisions 207317 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r207317 | tilghman | 2009-07-17 23:16:44 -0500 (Fri, 17 Jul 2009) | 3 lines Flag field in wrong position. Reported by "Hoggins!" on asterisk-dev list. ........ 2009-07-18 02:09 +0000 [r207287] Richard Mudgett * channels/misdn/isdn_lib.c, channels/misdn_config.c, channels/misdn/isdn_lib_intern.h, channels/misdn/isdn_lib.h, doc/tex/misdn.tex, channels/chan_misdn.c, main/callerid.c, configs/misdn.conf.sample: Merged revisions 145293,158010 from https://origsvn.digium.com/svn/asterisk/branches/1.4 to make merging easier. These changes are already on trunk. ................ r145293 | rmudgett | 2008-09-30 18:55:24 -0500 (Tue, 30 Sep 2008) | 54 lines channels/chan_misdn.c channels/misdn/isdn_lib.c * Miscellaneous other fixes from trunk to make merging easier later. ........ r145200 | rmudgett | 2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) | 7 lines * Miscellaneous formatting changes to make v1.4 and trunk more merge compatible in the mISDN area. channels/chan_misdn.c * Eliminated redundant code in cb_events() EVENT_SETUP ........ r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008) | 9 lines improved helptext of misdn_set_opt. ........ r142181 | rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line Cleaned up comment ........ r138738 | rmudgett | 2008-08-18 16:07:28 -0500 (Mon, 18 Aug 2008) | 30 lines channels/chan_misdn.c * Made bearer2str() use allowed_bearers_array[] * Made use the causes.h defines instead of hardcoded numbers. * Made use Asterisk presentation indicator values if either of the mISDN presentation or screen options are negative. * Updated the misdn_set_opt application option descriptions. * Renamed the awkward Caller ID presentation misdn_set_opt application option value not_screened to restricted. Deprecated the not_screened option value. channels/misdn/isdn_lib.c * Made use the causes.h defines instead of hardcoded numbers. * Fixed some spelling errors and typos. * Added all defined facility code strings to fac2str(). channels/misdn/isdn_lib.h * Added doxygen comments to struct misdn_bchannel. channels/misdn/isdn_lib_intern.h * Added doxygen comments to struct misdn_stack. channels/misdn_config.c configs/misdn.conf.sample * Updated the mISDN presentation and screen parameter descriptions. doc/misdn.txt (doc/tex/misdn.tex) * Updated the misdn_set_opt application option descriptions. * Fixed some spelling errors and typos. ................ r158010 | rmudgett | 2008-11-19 19:46:09 -0600 (Wed, 19 Nov 2008) | 9 lines Merged revision 157977 from https://origsvn.digium.com/svn/asterisk/team/group/issue8824 ........ Fixes JIRA ABE-1726 The dial extension could be empty if you are using MISDN_KEYPAD to control ISDN provider features. ................ 2009-07-17 22:30 +0000 [r207227-207256] Tilghman Lesher * /, doc/voicemail_odbc_postgresql.txt: Merged revisions 207255 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r207255 | tilghman | 2009-07-17 17:29:50 -0500 (Fri, 17 Jul 2009) | 2 lines Add flag here, too (as requested by jsmith) ........ * /, doc/tex/odbcstorage.tex, UPGRADE.txt: Recorded merge of revisions 207224 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r207224 | tilghman | 2009-07-17 17:04:43 -0500 (Fri, 17 Jul 2009) | 2 lines Document the "flag" field in the voicemessages table. ........ 2009-07-17 19:39 +0000 [r207101-207158] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 207156 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r207156 | jpeeler | 2009-07-17 14:37:38 -0500 (Fri, 17 Jul 2009) | 14 lines Merged revisions 207155 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207155 | jpeeler | 2009-07-17 14:36:19 -0500 (Fri, 17 Jul 2009) | 7 lines Fix format specifier to print out an unsigned long long. Yep, it's even ifdefed out code. But it made it to the RR list... (closes issue #14726) Reported by: lmadsen ........ ................ * configs/chan_dahdi.conf.sample, /: Merged revisions 207095 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r207095 | jpeeler | 2009-07-17 14:16:35 -0500 (Fri, 17 Jul 2009) | 2 lines Update some missing allowed options for overlapdial ........ 2009-07-17 17:53 +0000 [r206870-207031] David Vossel * /, channels/chan_sip.c: Merged revisions 207029 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r207029 | dvossel | 2009-07-17 12:51:44 -0500 (Fri, 17 Jul 2009) | 6 lines sip option flags handled incorrectly (closes issue #15376) Reported by: Takehiko Ooshima Tested by: dvossel, Takehiko_Ooshima ........ * /, channels/chan_sip.c: Merged revisions 206939 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206939 | dvossel | 2009-07-17 11:13:22 -0500 (Fri, 17 Jul 2009) | 20 lines Merged revisions 206938 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206938 | dvossel | 2009-07-17 11:05:06 -0500 (Fri, 17 Jul 2009) | 14 lines SIP incorrect From: header information when callpres is prohib Some ITSP make use of the "Anonymous" display name to detect a requirement to withhold caller id across the PSTN. This does not work if the display name is "Unknown". (closes issue #14465) Reported by: Nick_Lewis Patches: chan_sip.c-callerpres.patch uploaded by Nick (license 657) chan_sip.c-callerpres_trunk.patch uploaded by dvossel (license 671) Tested by: Nick_Lewis, dvossel ........ ................ * /, funcs/func_timeout.c: Merged revisions 206877 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r206877 | dvossel | 2009-07-16 16:45:14 -0500 (Thu, 16 Jul 2009) | 6 lines TIMEOUT(absolute) returned negative value. (closes issue #15513) Reported by: ys ........ * configs/iax.conf.sample, /: Merged revisions 206873 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206873 | dvossel | 2009-07-16 16:33:51 -0500 (Thu, 16 Jul 2009) | 12 lines Merged revisions 206872 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206872 | dvossel | 2009-07-16 16:33:19 -0500 (Thu, 16 Jul 2009) | 6 lines error in iax.conf related IP-based access control (closes issue #15518) Reported by: pkempgen ........ ................ * /, main/callerid.c: Merged revisions 206868 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206868 | dvossel | 2009-07-16 16:25:22 -0500 (Thu, 16 Jul 2009) | 14 lines Merged revisions 206867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206867 | dvossel | 2009-07-16 16:24:16 -0500 (Thu, 16 Jul 2009) | 8 lines avoid segfault caused by user error If the CALLERPRES() dialplan function is set to nothing, a segfault occurs. This is user error to begin with, but I'd rather see a cli warning message than have Asterisk crash on me. ........ ................ 2009-07-16 16:53 +0000 [r206810] Tilghman Lesher * funcs/func_realtime.c, /: Merged revisions 206808 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206808 | tilghman | 2009-07-16 11:51:05 -0500 (Thu, 16 Jul 2009) | 13 lines Merged revisions 206807 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206807 | tilghman | 2009-07-16 11:27:35 -0500 (Thu, 16 Jul 2009) | 6 lines Fix a memory leak. (closes issue #15517) Reported by: adomjan Patches: func_realtime.c-ast_variable_destroy.diff uploaded by adomjan (license 487) ........ ................ 2009-07-15 22:06 +0000 [r206774] David Vossel * /, channels/chan_sip.c: Merged revisions 206768 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r206768 | dvossel | 2009-07-15 17:04:13 -0500 (Wed, 15 Jul 2009) | 8 lines Session timer were not activated if Supported header field in INVITE had both "timer" and other options. (closes issue #15403) Reported by: makoto Patches: sip-session-timer.patch uploaded by makoto (license ........ 2009-07-15 21:40 +0000 [r206764] Richard Mudgett * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h, /: Merged revisions 206707 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206707 | rmudgett | 2009-07-15 16:14:41 -0500 (Wed, 15 Jul 2009) | 33 lines Merged revisions 206706 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r206706 | rmudgett | 2009-07-15 15:44:55 -0500 (Wed, 15 Jul 2009) | 26 lines Merged revision 206700 from https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... .......... Fixed chan_misdn crash because mISDNuser library is not thread safe. With Asterisk the mISDNuser library is driven by two threads concurrently: 1. channels/misdn/isdn_lib.c::manager_event_handler() 2. channels/misdn/isdn_lib.c::misdn_lib_isdn_event_catcher() Calls into the library are done concurrently and recursively from isdn_lib.c. Both threads can fiddle with the master/child layer3_proc_t lists. One thread may traverse the list when the other interrupts it and then removes the list element which the first thread was currently handling. This is exactly what caused the crash. About 60 calls were needed to a Gigaset CX475 before it occurred once. This patch adds locking when calling into the mISDNuser library. This also fixes some cb_log calls with wrong port parameter. JIRA ABE-1913 Patches: misdn-locking.patch (Modified with mostly cosmetic changes) .......... ................ ................ 2009-07-15 20:21 +0000 [r206704] David Vossel * /, channels/chan_sip.c: Merged revisions 206702 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r206702 | dvossel | 2009-07-15 15:20:01 -0500 (Wed, 15 Jul 2009) | 10 lines callerid(num) is wrong when username is missing A domain only sip uri would return 123.123.123.123 as callid num. Now, if the username is missing from a uri, the callerid num field is left empty. (closes issue #15476) Reported by: viraptor ........ 2009-07-15 16:03 +0000 [r206638] Sean Bright * /, codecs/codec_dahdi.c: Merged revisions 206636 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206636 | seanbright | 2009-07-15 12:00:24 -0400 (Wed, 15 Jul 2009) | 9 lines Merged revisions 206635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206635 | seanbright | 2009-07-15 11:57:51 -0400 (Wed, 15 Jul 2009) | 1 line Only print debug info in codec_dahdi if we are asking for it. ........ ................ 2009-07-14 20:25 +0000 [r206596] Tilghman Lesher * /, contrib/scripts/meetme.sql: Recorded merge of revisions 206567 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r206567 | tilghman | 2009-07-14 15:14:45 -0500 (Tue, 14 Jul 2009) | 6 lines Document all meetme realtime fields, and in the process, make some field lengths more consistent. (closes issue #15493) Reported by: lasko Patches: meetme.diff uploaded by lasko (license 833) ........ 2009-07-14 18:32 +0000 [r206558] Richard Mudgett * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, channels/chan_misdn.c, /: Merged revisions 206489 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206489 | rmudgett | 2009-07-14 12:01:48 -0500 (Tue, 14 Jul 2009) | 35 lines Merged revisions 206487 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206487 | rmudgett | 2009-07-14 11:44:47 -0500 (Tue, 14 Jul 2009) | 28 lines Fixes several call transfer issues with chan_misdn. * issue #14355 - Crash if attempt to transfer a call to an application. Masquerade the other pair of the four asterisk channels involved in the two calls. The held call already must be a bridged call (not an applicaton) or it would have been rejected. * issue #14692 - Held calls are not automatically cleared after transfer. Allow the core to initate disconnect of held calls to the ISDN port. This also fixes a similar case where the party on hold hangs up before being transferred or taken off hold. * JIRA ABE-1903 - Orphaned held calls left in music-on-hold. Do not simply block passing the hangup event on held calls to asterisk core. * Fixed to allow held calls to be transferred to ringing calls. Previously, held calls could only be transferred to connected calls. * Eliminated unused call states to simplify hangup code. * Eliminated most uses of "holded" because it is not a word. (closes issue #14355) (closes issue #14692) Reported by: sodom Patches: misdn_xfer_v14_r205839.patch uploaded by rmudgett (license 664) Tested by: rmudgett ........ ................ 2009-07-14 14:56 +0000 [r206388] Russell Bryant * channels/chan_iax2.c, /: Merged revisions 206386 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206386 | russell | 2009-07-14 09:51:44 -0500 (Tue, 14 Jul 2009) | 20 lines Merged revisions 206385 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r206385 | russell | 2009-07-14 09:48:00 -0500 (Tue, 14 Jul 2009) | 13 lines Merged revisions 206384 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r206384 | russell | 2009-07-14 09:45:47 -0500 (Tue, 14 Jul 2009) | 6 lines Ensure apathetic replies are sent out on the proper socket. chan_iax2 supports multiple address bindings. The send_apathetic_reply() function did not attempt to send its response on the same socket that the incoming message came in on. ........ ................ ................ 2009-07-14 01:35 +0000 [r206372] Richard Mudgett * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged revisions 206341 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r206341 | rmudgett | 2009-07-13 19:48:59 -0500 (Mon, 13 Jul 2009) | 11 lines Merged revisions 206284 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206284 | rmudgett | 2009-07-13 19:17:28 -0500 (Mon, 13 Jul 2009) | 4 lines Fix some memory leaks in chan_misdn. JIRA ABE-1911 ........ ................ 2009-07-13 23:33 +0000 [r206282] David Vossel * /, channels/chan_sip.c: Merged revisions 206280 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r206280 | dvossel | 2009-07-13 18:26:51 -0500 (Mon, 13 Jul 2009) | 9 lines dns lookup of peername rather than peer's host in transmit_register() (closes issue #15052) Reported by: fsantulli Patches: chan_sip_bug_15052_[20090626204511].patch uploaded by fsantulli (license 818) Tested by: fsantulli ........ 2009-07-13 16:24 +0000 [r206186] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 206185 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r206185 | tilghman | 2009-07-13 11:23:07 -0500 (Mon, 13 Jul 2009) | 2 lines Remove reference to non-existent help file ........ 2009-07-10 21:52 +0000 [r205987] David Vossel * /, channels/chan_sip.c: Merged revisions 205985 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205985 | dvossel | 2009-07-10 16:42:10 -0500 (Fri, 10 Jul 2009) | 16 lines SIP register not using peer's outbound proxy If callbackextension is defined for a peer it successfully causes a registration to occur, but the registration ignores the outboundproxy settings for the peer. This patch allows the peer to be passed to obproxy_get() in transmit_register(). (closes issue #14344) Reported by: Nick_Lewis Patches: callbackextension_peer_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel Review: https://reviewboard.asterisk.org/r/294/ ........ 2009-07-10 18:45 +0000 [r205941] Kevin P. Fleming * main/udptl.c, /: Merged revisions 205939 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205939 | kpfleming | 2009-07-10 13:44:09 -0500 (Fri, 10 Jul 2009) | 1 line Update comments about the level of T.38 support in Asterisk. ........ 2009-07-10 17:50 +0000 [r205881] Mark Michelson * /, channels/chan_sip.c: Merged revisions 205878 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205878 | mmichelson | 2009-07-10 12:39:57 -0500 (Fri, 10 Jul 2009) | 30 lines Merged revisions 205877 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r205877 | mmichelson | 2009-07-10 12:39:13 -0500 (Fri, 10 Jul 2009) | 23 lines Merged revisions 205776 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205776 | mmichelson | 2009-07-10 10:56:45 -0500 (Fri, 10 Jul 2009) | 16 lines Merged revisions 205775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul 2009) | 10 lines Ensure that outbound NOTIFY requests are properly routed through stateful proxies. With this change, we make note of Record-Route headers present in any SUBSCRIBE request that we receive so that our outbound NOTIFY requests will have the proper Route headers in them. (closes issue #14725) Reported by: ibc ........ ................ ................ ................ 2009-07-10 16:48 +0000 [r205842] David Vossel * /, channels/chan_sip.c: Merged revisions 205840 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205840 | dvossel | 2009-07-10 11:42:04 -0500 (Fri, 10 Jul 2009) | 37 lines Merged revisions 205804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205804 | dvossel | 2009-07-10 11:23:59 -0500 (Fri, 10 Jul 2009) | 31 lines SIP registration auth loop caused by stale nonce If an endpoint sends two registration requests in a very short period of time with the same nonce, both receive 401 responses from Asterisk, each with a different nonce (the second 401 containing the current nonce and the first one being stale). If the endpoint responds to the first 401, it does not match the current nonce so Asterisk sends a third 401 with a newly generated nonce (which updates the current nonce)... Now if the endpoint responds to the second 401, it does not match the current nonce either and Asterisk sends a fourth 401 with a newly generated nonce... This loop goes on and on. There appears to be a simple fix for this. If the nonce from the request does not match our nonce, but is a good response to a previous nonce, instead of sending a 401 with a newly generated nonce, use the current one instead. This breaks the loop as the nonce is not updated until a response is received. Additional logic has been added to make sure no nonce can be responded to twice though. (closes issue #15102) Reported by: Jamuel Patches: patch-bug_0015102 uploaded by Jamuel (license 809) nonce_sip.diff uploaded by dvossel (license 671) Tested by: Jamuel Review: https://reviewboard.asterisk.org/r/289/ ........ ................ 2009-07-10 15:57 +0000 [r205778] Mark Michelson * /, channels/chan_sip.c: Merged revisions 205776 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205776 | mmichelson | 2009-07-10 10:56:45 -0500 (Fri, 10 Jul 2009) | 16 lines Merged revisions 205775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul 2009) | 10 lines Ensure that outbound NOTIFY requests are properly routed through stateful proxies. With this change, we make note of Record-Route headers present in any SUBSCRIBE request that we receive so that our outbound NOTIFY requests will have the proper Route headers in them. (closes issue #14725) Reported by: ibc ........ ................ 2009-07-10 15:36 +0000 [r205772] Kevin P. Fleming * /, apps/app_fax.c: Merged revisions 205770 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205770 | kpfleming | 2009-07-10 10:28:11 -0500 (Fri, 10 Jul 2009) | 12 lines Fix some remaining T.38 negotiation problems in app_fax. Revision 205696 did not quite fix all the issues with the T.38 negotiation changes and app_fax; this patch corrects them, along with a couple of other minor issues. (closes issue #15480) Reported by: dimas Patches: test2-15480.patch uploaded by dimas (license 88) ........ 2009-07-09 23:51 +0000 [r205730] Richard Mudgett * channels/chan_dahdi.c: Merged revisions 205728 via svn merge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205728 | rmudgett | 2009-07-09 18:37:53 -0500 (Thu, 09 Jul 2009) | 21 lines No audio on calls from Asterisk to various ISDN devices until DTMF sent by caller. Add missing clearing of the dialing flag when the ISDN call is CONNECTED. (i.e. When libpri generates the event PRI_EVENT_ANSWER.) (closes issue #15420) Reported by: scottbmilne Patches: bug15420-1.4.25.1-diff2.txt uploaded by alecdavis (license 585) Tested by: scottbmilne, alecdavis (closes issue #15416) Reported by: avinoash (closes issue #15389) Reported by: alecdavis This patch should also fix the following issue: (issue #15205) Reported by: vinsik ........ 2009-07-09 21:27 +0000 [r205698] Kevin P. Fleming * include/asterisk/frame.h, /, channels/chan_sip.c, apps/app_fax.c: Merged revisions 205696 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205696 | kpfleming | 2009-07-09 16:20:23 -0500 (Thu, 09 Jul 2009) | 16 lines Repair ability of SendFAX/ReceiveFAX to respond to T.38 switchover. Recent changes in T.38 negotiation in Asterisk caused these applications to not respond when the other endpoint initiated a switchover to T.38; this resulted in the T.38 switchover failing, and the FAX attempt to be made using an audio connection, instead of T.38 (which would usually cause the FAX to fail completely). This patch corrects this problem, and the applications will now correctly respond to the T.38 switchover request. In addition, the response will include the appopriate T.38 session parameters based on what the other end offered and what our end is capable of. (closes issue #14849) Reported by: afosorio ........ 2009-07-09 16:20 +0000 [r205596-205605] David Vossel * include/asterisk/time.h, /: Merged revisions 205600 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205600 | dvossel | 2009-07-09 11:19:09 -0500 (Thu, 09 Jul 2009) | 9 lines Merged revisions 205599 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205599 | dvossel | 2009-07-09 11:18:09 -0500 (Thu, 09 Jul 2009) | 2 lines Changing ast_samp2tv to not use floating point. ........ ................ * channels/chan_iax2.c, include/asterisk/frame.h, main/rtp.c, /: Merged revisions 205479 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205479 | dvossel | 2009-07-08 18:19:09 -0500 (Wed, 08 Jul 2009) | 16 lines Merged revisions 205471 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205471 | dvossel | 2009-07-08 18:15:54 -0500 (Wed, 08 Jul 2009) | 10 lines Fixes 8khz assumptions Many calculations assume 8khz is the codec rate. This is not always the case. This patch only addresses chan_iax.c and res_rtp_asterisk.c, but I am sure there are other areas that make this assumption as well. Review: https://reviewboard.asterisk.org/r/306/ ........ ................ 2009-07-09 08:33 +0000 [r205534] Michiel van Baak * /, main/ssl.c: Merged revisions 205532 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205532 | mvanbaak | 2009-07-09 10:31:24 +0200 (Thu, 09 Jul 2009) | 5 lines pthread_self returns a pthread_t which is not an unsigned int on all pthread implementations. Casting it to an unsigned int fixes compiler warnings. Tested on OpenBSD and Linux both 32 and 64 bit ........ 2009-07-08 22:16 +0000 [r205414] David Vossel * include/asterisk/devicestate.h, main/pbx.c, /, main/devicestate.c, include/asterisk/pbx.h: Merged revisions 205412 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205412 | dvossel | 2009-07-08 17:15:06 -0500 (Wed, 08 Jul 2009) | 12 lines Merged revisions 205409 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205409 | dvossel | 2009-07-08 16:35:12 -0500 (Wed, 08 Jul 2009) | 6 lines moving ast_devstate_to_extenstate to pbx.c from devicestate.c ast_devstate_to_extenstate belongs in pbx.c. This change fixes a compile time error with chan_vpb as well. ........ ................ 2009-07-08 19:27 +0000 [r205352] Mark Michelson * apps/app_queue.c, /: Merged revisions 205350 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205350 | mmichelson | 2009-07-08 14:26:55 -0500 (Wed, 08 Jul 2009) | 20 lines Merged revisions 205349 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205349 | mmichelson | 2009-07-08 14:26:13 -0500 (Wed, 08 Jul 2009) | 14 lines Prevent phantom calls to queue members. If a caller were to hang up while a periodic announcement or position were being said, the return value for those functions would incorrectly indicate that the caller was still in the queue. With these changes, the problem does not occur. (closes issue #14631) Reported by: latinsud Patches: queue_announce_ghost_call2.diff uploaded by latinsud (license 745) (with small modification from me) ........ ................ 2009-07-08 18:21 +0000 [r205299] Jason Parker * config.guess, config.sub, /: Merged revisions 205291 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205291 | qwell | 2009-07-08 13:19:46 -0500 (Wed, 08 Jul 2009) | 9 lines Merged revisions 205288 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205288 | qwell | 2009-07-08 13:19:03 -0500 (Wed, 08 Jul 2009) | 1 line Update config.guess and config.sub from the savannah.gnu.org git repo. ........ ................ 2009-07-08 18:07 +0000 [r205279] David Brooks * /, main/features.c: Merged revisions 205254 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205254 | dbrooks | 2009-07-08 12:26:26 -0500 (Wed, 08 Jul 2009) | 8 lines Fixes Park() argument handling Park() was not respecting the arguments passed to it. Any extension/context/priority given to it was being ignored. This patch remedies this. (closes issue #15380) Reported by: DLNoah ........ 2009-07-08 16:59 +0000 [r205222] Tilghman Lesher * main/say.c: oops, fixing build 2009-07-08 16:56 +0000 [r205218] David Vossel * include/asterisk/time.h, /: Merged revisions 205216 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205216 | dvossel | 2009-07-08 11:54:24 -0500 (Wed, 08 Jul 2009) | 17 lines Merged revisions 205215 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205215 | dvossel | 2009-07-08 11:53:40 -0500 (Wed, 08 Jul 2009) | 10 lines ast_samp2tv needs floating point for 16khz audio In ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate is 16000. The .5 is currently stripped off because we don't calculate using floating points. This causes madness with 16khz audio. (issue ABE-1899) Review: https://reviewboard.asterisk.org/r/305/ ........ ................ 2009-07-08 16:29 +0000 [r205203] Tilghman Lesher * /, main/say.c: Merged revisions 205196 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205196 | tilghman | 2009-07-08 11:27:50 -0500 (Wed, 08 Jul 2009) | 9 lines Merged revisions 205188 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205188 | tilghman | 2009-07-08 11:26:15 -0500 (Wed, 08 Jul 2009) | 2 lines Add redirection warnings for the invalid language codes previously removed. ........ ................ 2009-07-08 15:57 +0000 [r205147-205153] Russell Bryant * /, main/ssl.c: Merged revisions 205151 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205151 | russell | 2009-07-08 10:56:28 -0500 (Wed, 08 Jul 2009) | 2 lines Use tabs instead of spaces for indentation. ........ * res/res_jabber.c, main/asterisk.c, /, main/Makefile, res/res_crypto.c, main/ssl.c (added), include/asterisk/_private.h: Merged revisions 205120 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r205120 | russell | 2009-07-08 10:17:19 -0500 (Wed, 08 Jul 2009) | 16 lines Move OpenSSL initialization to a single place, make library usage thread-safe. While doing some reading about OpenSSL, I noticed a couple of things that needed to be improved with our usage of OpenSSL. 1) We had initialization of the library done in multiple modules. This has now been moved to a core function that gets executed during Asterisk startup. We already link OpenSSL into the core for TCP/TLS functionality, so this was the most logical place to do it. 2) OpenSSL is not thread-safe by default. However, making it thread safe is very easy. We just have to provide a couple of callbacks. One callback returns a thread ID. The other handles locking. For more information, start with the "Is OpenSSL thread-safe?" question on the FAQ page of openssl.org. ........ 2009-07-06 14:24 +0000 [r204976] Ryan Brindley * main/config.c, /: Merged revisions 202753 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r202753 | rbrindley | 2009-06-23 16:25:17 -0500 (Tue, 23 Jun 2009) | 9 lines If we delete the info, lets also delete the lines (closes issue 0014509) Reported by: timeshell Patches: 20090504__bug14509.diff.txt uploaded by tilghman (license 14) Tested by: awk, timeshell ........ 2009-07-06 13:40 +0000 [r204950] Kevin P. Fleming * main/channel.c, /: Merged revisions 204948 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r204948 | kpfleming | 2009-07-06 08:38:29 -0500 (Mon, 06 Jul 2009) | 7 lines Improve handling of AST_CONTROL_T38 and AST_CONTROL_T38_PARAMETERS for non-T.38-capable channels. This change allows applications that request T.38 negotiation on a channel that does not support it to get the proper indication that it is not supported, rather than thinking that negotiation was started when it was not. ........ 2009-07-02 22:05 +0000 [r204837] Richard Mudgett * channels/chan_misdn.c, /: Merged revisions 204835 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r204835 | rmudgett | 2009-07-02 17:01:28 -0500 (Thu, 02 Jul 2009) | 17 lines Merged revisions 204834 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204834 | rmudgett | 2009-07-02 16:59:43 -0500 (Thu, 02 Jul 2009) | 10 lines Removed confusing warning message "Got Busy in Connected State" If an incoming mISDN call is answered with the Answer application and a subsequent Dial gets a busy endpoint then it is valid for that already connected channel to get the busy indication. Asterisk will play the busy tones until the dialplan plays something else or hangs up the call. (closes issue #11974) Reported by: fvdb ........ ................ 2009-07-02 16:28 +0000 [r204736] David Vossel * include/asterisk/devicestate.h, main/pbx.c, /, main/devicestate.c: Merged revisions 204710 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r204710 | dvossel | 2009-07-02 11:03:44 -0500 (Thu, 02 Jul 2009) | 21 lines Merged revisions 204681 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204681 | dvossel | 2009-07-02 10:05:57 -0500 (Thu, 02 Jul 2009) | 14 lines Improved mapping of extension states from combined device states. This fixes a few issues with incorrect extension states and adds a cli command, core show device2extenstate, to display all possible state mappings. (closes issue #15413) Reported by: legart Patches: exten_helper.diff uploaded by dvossel (license 671) Tested by: dvossel, legart, amilcar Review: https://reviewboard.asterisk.org/r/301/ ........ ................ 2009-06-30 21:30 +0000 [r204612] Tilghman Lesher * /, main/say.c, UPGRADE.txt: Merged revisions 204563 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r204563 | tilghman | 2009-06-30 15:41:04 -0500 (Tue, 30 Jun 2009) | 13 lines Merged revisions 204556 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204556 | tilghman | 2009-06-30 15:23:51 -0500 (Tue, 30 Jun 2009) | 6 lines More incorrect language codes, plus ensuring that regionalizations use the specified language, and not English for grammar. (closes issue #15022) Reported by: greenfieldtech Patches: 20090519__issue15022.diff.txt uploaded by tilghman (license 14) ........ ................ 2009-06-30 18:52 +0000 [r204477] Jason Parker * /, main/say.c: Merged revisions 204475 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r204475 | qwell | 2009-06-30 13:48:35 -0500 (Tue, 30 Jun 2009) | 9 lines Merged revisions 204474 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204474 | qwell | 2009-06-30 13:47:06 -0500 (Tue, 30 Jun 2009) | 1 line Fix ast_say_counted_noun to correctly handle Polish. Fix a comment typo in passing. ........ ................ 2009-06-30 18:44 +0000 [r204472] Tilghman Lesher * apps/app_voicemail.c, /, main/say.c, UPGRADE.txt: Recorded merge of revisions 204470 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r204470 | tilghman | 2009-06-30 13:36:24 -0500 (Tue, 30 Jun 2009) | 18 lines Recorded merge of revisions 204469 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204469 | tilghman | 2009-06-30 13:23:35 -0500 (Tue, 30 Jun 2009) | 11 lines "tw" is the language specification for Twi (from Ghana) not Taiwanese. (closes issue #15346) Reported by: volivier Patches: 20090617__issue15346__1.4.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__trunk.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__1.6.0.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__1.6.1.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__1.6.2.diff.txt uploaded by tilghman (license 14) Tested by: volivier ........ ................ 2009-06-29 22:53 +0000 [r204249-204303] Mark Michelson * /, channels/chan_sip.c: Merged revisions 204301 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r204301 | mmichelson | 2009-06-29 17:50:35 -0500 (Mon, 29 Jun 2009) | 15 lines Merged revisions 204300 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204300 | mmichelson | 2009-06-29 17:45:34 -0500 (Mon, 29 Jun 2009) | 9 lines Add error message so that it is clear why a SIP peer was not processed when a DNS lookup fails on a host or outboundproxy. (closes issue #13432) Reported by: p_lindheimer Patches: outboundproxy.patch uploaded by p (license 558) ........ ................ * /, channels/chan_sip.c: Merged revisions 204247 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r204247 | mmichelson | 2009-06-29 16:48:54 -0500 (Mon, 29 Jun 2009) | 32 lines Merged revisions 204243,204246 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204243 | mmichelson | 2009-06-29 16:23:43 -0500 (Mon, 29 Jun 2009) | 22 lines Fix a problem where chan_sip would ignore "old" but valid responses. chan_sip has had a problem for quite a long time that would manifest when Asterisk would send multiple SIP responses on the same dialog before receiving a response. The problem occurred because chan_sip only kept track of the highest outgoing sequence number used on the dialog. If Asterisk sent two requests out, and a response arrived for the first request sent, then Asterisk would ignore the response. The result was that Asterisk would continue retransmitting the requests and ignoring the responses until the maximum number of retransmissions had been reached. The fix here is to rearrange the code a bit so that instead of simply comparing the sequence number of the response to our latest outgoing sequence number, we walk our list of outstanding packets and determine if there is a match. If there is, we continue. If not, then we ignore the response. In doing this, I found a few completely useless variables that I have now removed. (closes issue #11231) Reported by: flefoll Review: https://reviewboard.asterisk.org/r/298 ........ r204246 | mmichelson | 2009-06-29 16:37:05 -0500 (Mon, 29 Jun 2009) | 3 lines Fix build oops. ........ ................ 2009-06-27 01:18 +0000 [r203918] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 203909 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r203909 | rmudgett | 2009-06-26 20:07:52 -0500 (Fri, 26 Jun 2009) | 23 lines Merged revisions 203908 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203908 | rmudgett | 2009-06-26 19:55:12 -0500 (Fri, 26 Jun 2009) | 16 lines The ISDN CPE side should not exclusively pick B channels normally. Before this patch, Asterisk unconditionally picked B channels exclusively on the CPE side and normally allowed alternative B channels on the network side. Now Asterisk does the opposite. Reasons for the CPE side to normally not pick B channels exclusively: * For CPE point-to-multipoint mode (i.e. phone side), the CPE side does not have enough information to exclusively pick B channels. (There may be other devices on the line.) * Q.931 gives preference to the network side picking B channels. * Some telcos require the CPE side to not pick B channels exclusively. (closes issue #14383) Reported by: mbrancaleoni ........ ................ 2009-06-26 22:13 +0000 [r203856] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 203853 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r203853 | jpeeler | 2009-06-26 17:11:31 -0500 (Fri, 26 Jun 2009) | 12 lines Merged revisions 203848 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203848 | jpeeler | 2009-06-26 17:09:19 -0500 (Fri, 26 Jun 2009) | 5 lines Make sure to recreate the dahdi pseudo channel after dahdi restart (closes issue #14477) Reported by: timking ........ ................ 2009-06-26 21:26 +0000 [r203781-203823] Russell Bryant * /, main/file.c: Merged revisions 203802 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r203802 | russell | 2009-06-26 16:21:48 -0500 (Fri, 26 Jun 2009) | 22 lines Merged revisions 203785 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203785 | russell | 2009-06-26 16:16:39 -0500 (Fri, 26 Jun 2009) | 15 lines Don't fast forward past the end of a message. This is nice change for users of the voicemail application. If someone gets a little carried away with fast forwarding through a message, they can easily get to the end and accidentally exit the voicemail application by hitting the fast forward key during the following prompt. This adds some safety by not allowing a fast forward past the end of a message. (closes issue #14554) Reported by: lacoursj Patches: 21761.patch uploaded by lacoursj (license 707) Tested by: lacoursj ........ ................ * /, channels/chan_sip.c: Merged revisions 203779 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203779 | russell | 2009-06-26 15:45:00 -0500 (Fri, 26 Jun 2009) | 5 lines Ensure the TCP read buffer is fully initialized before handling each packet. (closes issue #14452) Reported by: umberto71 ........ 2009-06-26 20:18 +0000 [r203727] David Brooks * apps/app_voicemail.c, /: Merged revisions 203721 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203721 | dbrooks | 2009-06-26 15:13:51 -0500 (Fri, 26 Jun 2009) | 16 lines Fixing voicemail's error in checking max silence vs min message length Max silence was represented in milliseconds, yet vmminsecs (minmessage) was represented as seconds. Also, the inequality was reversed. The warning, if triggered, was "Max silence should be less than minmessage or you may get empty messages", which should have been logged if max silence was greater than minmessage, but the check was for less than. Also, conforming if statement to coding guidelines. closes issue #15331) Reported by: markd Review: https://reviewboard.asterisk.org/r/293/ ........ 2009-06-26 19:56 +0000 [r203718] Jeff Peeler * channels/chan_dahdi.c: reverse whitespace change 203713 that was based on looking at sig_analog (which has about a 1000 line indentation change that is not worth doing here) 2009-06-26 19:48 +0000 [r203714] David Vossel * channels/chan_iax2.c, /: Merged revisions 203710 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203710 | dvossel | 2009-06-26 14:47:11 -0500 (Fri, 26 Jun 2009) | 7 lines moving debug message from level 0 to 1. (closes issue #15404) Reported by: leobrown Patches: iax_codec_debug.patch uploaded by leobrown (license 541) ........ 2009-06-26 19:48 +0000 [r203713] Jeff Peeler * channels/chan_dahdi.c: whitespace fix 2009-06-26 19:37 +0000 [r203704] Russell Bryant * include/asterisk/devicestate.h, main/pbx.c, /, main/devicestate.c: Merged revisions 203702 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203702 | russell | 2009-06-26 14:31:14 -0500 (Fri, 26 Jun 2009) | 5 lines Make invalid hints report Unavailable instead of Idle. (closes issue #14413) Reported by: pj ........ 2009-06-26 19:31 +0000 [r203703] Joshua Colp * include/asterisk/frame.h, main/rtp.c, main/channel.c, main/frame.c, /, channels/chan_sip.c, apps/app_fax.c, configs/sip.conf.sample: Merged revisions 203699 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203699 | file | 2009-06-26 16:27:24 -0300 (Fri, 26 Jun 2009) | 2 lines Improve T.38 negotiation by exchanging session parameters between application and channel. ........ 2009-06-26 19:28 +0000 [r203700] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 203672 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203672 | jpeeler | 2009-06-26 14:03:25 -0500 (Fri, 26 Jun 2009) | 16 lines Check if polarityonanswerdelay has elapsed before setting a channel as answered after a polarity reversal. Previously on a polarity switch event chan_dahdi would set the channel immediately as answered. This would cause problems if a polarity reversal occurred when the line was picked up as the dial would not have yet occurred. Now if the polarity reversal occurs before delay has elapsed after coming off hook or an answer, it is ignored. Also, some refactoring was done in _handle_event. (closes issue #13917) Reported by: alecdavis Patches: chan_dahdi.bug13917.feb09.diff2.txt uploaded by alecdavis (license 585) Tested by: alecdavis ........ 2009-06-25 21:46 +0000 [r203446] David Vossel * main/ast_expr2.fl, main/ast_expr2.c, /: Merged revisions 203444 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203444 | dvossel | 2009-06-25 16:45:32 -0500 (Thu, 25 Jun 2009) | 4 lines fixes a few redundant conditions (issue #15269) ........ 2009-06-25 21:19 +0000 [r203393] Terry Wilson * main/cli.c, /: Merged revisions 203381 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r203381 | twilson | 2009-06-25 16:15:11 -0500 (Thu, 25 Jun 2009) | 11 lines Merged revisions 203380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203380 | twilson | 2009-06-25 16:13:10 -0500 (Thu, 25 Jun 2009) | 4 lines I didn't see that Mark already fixed the underlying issue! Yay for removing useless code. ........ ................ 2009-06-25 21:07 +0000 [r203378] Russell Bryant * /, main/features.c: Merged revisions 203376 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r203376 | russell | 2009-06-25 16:04:55 -0500 (Thu, 25 Jun 2009) | 16 lines Merged revisions 203375 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203375 | russell | 2009-06-25 16:02:18 -0500 (Thu, 25 Jun 2009) | 9 lines Fix a case where CDR answer time could be before the start time involving parking. (closes issue #13794) Reported by: davidw Patches: 13794.patch uploaded by murf (license 17) 13794.patch.160 uploaded by murf (license 17) Tested by: murf, dbrooks ........ ................ 2009-06-25 19:27 +0000 [r203274] Jason Parker * channels/chan_dahdi.c, /: Merged revisions 203258 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203258 | qwell | 2009-06-25 14:22:46 -0500 (Thu, 25 Jun 2009) | 10 lines Unmute when we get a dtmfup (we muted on dtmfdown) event. This would occasionally cause one-way audio when using hardware DTMF detection. (closes issue #14761) Reported by: tzafrir Patches: v1-14761.patch uploaded by dimas (license 88) Tested by: tzafrir, dimas ........ 2009-06-25 16:07 +0000 [r203118] Russell Bryant * /, channels/chan_sip.c: Merged revisions 203116 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r203116 | russell | 2009-06-25 11:04:10 -0500 (Thu, 25 Jun 2009) | 18 lines Merged revisions 203115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203115 | russell | 2009-06-25 11:02:16 -0500 (Thu, 25 Jun 2009) | 11 lines Resolve a crash related to a T.38 reinvite race condition. This change resolves a crash observed locally during some T.38 testing. A call was set up using a call file, and when the T.38 reinvite came in, the channel state was still AST_STATE_DOWN. The reason is explained by a comment in the code that previously lived in the handling of AST_STATE_RINGING. This change modifies the logic to handle the same race condition for any channel state that is not UP. (closes ABE-1895) ........ ................ 2009-06-24 21:22 +0000 [r203057] Richard Mudgett * channels/chan_dahdi.c, /: Merged revisions 203037 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r203037 | rmudgett | 2009-06-24 16:08:55 -0500 (Wed, 24 Jun 2009) | 15 lines Merged revisions 203036 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203036 | rmudgett | 2009-06-24 16:01:43 -0500 (Wed, 24 Jun 2009) | 8 lines Improved chan_dahdi.conf pritimer error checking. Valid format is: pritimer=timer_name,timer_value * Fixed segfault if the ',' is missing. * Completely check the range returned by pri_timer2idx() to prevent possible access outside array bounds. ........ ................ 2009-06-24 18:30 +0000 [r202969] Mark Michelson * /, channels/chan_sip.c: Merged revisions 202967 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r202967 | mmichelson | 2009-06-24 13:29:10 -0500 (Wed, 24 Jun 2009) | 9 lines Merged revisions 202966 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202966 | mmichelson | 2009-06-24 13:28:47 -0500 (Wed, 24 Jun 2009) | 3 lines Use the handy UNLINK macro instead of hand-coding the same thing in-line. ........ ................ 2009-06-24 18:10 +0000 [r202927] Joshua Colp * /, channels/chan_sip.c: Merged revisions 202925 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r202925 | file | 2009-06-24 15:08:17 -0300 (Wed, 24 Jun 2009) | 2 lines Ensure the default settings are applied for T.38 when we set it up for a peer. ........ 2009-06-23 22:11 +0000 [r202764] Matthew Fredrickson * channels/chan_dahdi.c, /: Merged revisions 202761 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r202761 | mattf | 2009-06-23 17:08:43 -0500 (Tue, 23 Jun 2009) | 1 line I could have sworn I committed this patch ages ago, but... bug fix with setting NAI properly on linksets in certain situations. ........ 2009-06-23 16:34 +0000 [r202674] David Vossel * /, channels/chan_sip.c: Merged revisions 202672 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r202672 | dvossel | 2009-06-23 11:31:30 -0500 (Tue, 23 Jun 2009) | 18 lines Merged revisions 202671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202671 | dvossel | 2009-06-23 11:28:46 -0500 (Tue, 23 Jun 2009) | 12 lines MWI NOTIFY contains a wrong URI if Asterisk listens to non-standard port and transport (closes issue #14659) Reported by: klaus3000 Patches: patch_chan_sip_fixMWIuri_1.4.txt uploaded by klaus3000 (license 65) mwi_port-transport_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel, klaus3000 Review: https://reviewboard.asterisk.org/r/288/ ........ ................ 2009-06-22 20:18 +0000 [r202503] Russell Bryant * main/channel.c, /: Merged revisions 202497 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r202497 | russell | 2009-06-22 15:11:04 -0500 (Mon, 22 Jun 2009) | 11 lines Merged revisions 202496 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202496 | russell | 2009-06-22 15:08:53 -0500 (Mon, 22 Jun 2009) | 4 lines Report CallerID change during a masquerade. Reported by: markster ........ ................ 2009-06-22 16:31 +0000 [r202472] Sean Bright * cdr/cdr_sqlite3_custom.c, /: Merged revisions 202417 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r202417 | seanbright | 2009-06-22 12:09:50 -0400 (Mon, 22 Jun 2009) | 4 lines Fix lock usage in cdr_sqlite3_custom to avoid potential crashes during reload. Pointed out by Russell while working on the CEL branch. ........ 2009-06-22 16:14 +0000 [r202418] Russell Bryant * /, channels/chan_sip.c: Merged revisions 202415 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r202415 | russell | 2009-06-22 11:05:08 -0500 (Mon, 22 Jun 2009) | 9 lines Merged revisions 202414 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202414 | russell | 2009-06-22 11:00:00 -0500 (Mon, 22 Jun 2009) | 2 lines Make Polycom subscription type override check more explicit. ........ ................ 2009-06-22 15:41 +0000 [r202412] David Vossel * main/loader.c, /, include/asterisk/module.h: Merged revisions 202410 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r202410 | dvossel | 2009-06-22 10:33:35 -0500 (Mon, 22 Jun 2009) | 5 lines attempting to load running modules Modules placed in the priority heap for loading were not properly removed from the linked list. This resulted in some modules attempting to load twice. ........ 2009-06-22 15:10 +0000 [r202339-202345] Mark Michelson * /, channels/chan_sip.c: Merged revisions 202343 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r202343 | mmichelson | 2009-06-22 09:58:24 -0500 (Mon, 22 Jun 2009) | 36 lines Merged revisions 202341-202342 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202341 | mmichelson | 2009-06-22 09:42:55 -0500 (Mon, 22 Jun 2009) | 26 lines Fix a situation in which Asterisk would not stop retransmitting 487s. If a CANCEL were received by Asterisk, we would send a 487 in response to the original INVITE and a 200 OK for the CANCEL. If there were a network hiccup which caused the 200 OK and the 487 to be lost, then the UA communicating with Asterisk may try to retransmit its CANCEL. Asterisk's response to this used to be to try sending another 487 to the canceled INVITE and another 200 OK to the CANCEL. The problem here is that the originally-sent 487 was sent "reliably" meaning that it will be retransmitted until it is received properly. So when we receive the second CANCEL it is likely that the first batch of 487s we sent is still going strong and reaches the UA. The result was that the second set of 487s would be retransmitted constantly until the maximum number of retries had been reached. The fix for this is that if we receive a second CANCEL for an INVITE, then we cancel the retransmission of the first set of 487s and start a second set. This causes the dialog to be terminated reasonably. (closes issue #14584) Reported by: klaus3000 Patches: 14584_v2.patch uploaded by mmichelson (license 60) Tested by: klaus3000 ........ r202342 | mmichelson | 2009-06-22 09:44:58 -0500 (Mon, 22 Jun 2009) | 3 lines Remove an extra debug line left from previous commit. ........ ................ * /, channels/chan_sip.c: Merged revisions 202337 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r202337 | mmichelson | 2009-06-22 09:35:09 -0500 (Mon, 22 Jun 2009) | 31 lines Merged revisions 202336 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202336 | mmichelson | 2009-06-22 09:34:05 -0500 (Mon, 22 Jun 2009) | 25 lines Fix a possible infinite loop in SDP parsing during glare situation. There was a while loop in get_ip_and_port_from_sdp which was controlled by a call to get_sdp_iterate. The loop would exit either if what we were searching for was found or if the return was NULL. The problem is that get_sdp_iterate never returns NULL. This means that if what we were searching for was not present, the loop would run infinitely. This modification of the loop fixes the problem. (closes issue #15213) Reported by: schmidts (closes issue #15349) Reported by: samy (closes issue #14464) Reported by: pj (closes issue #15345) Reported by: aragon Patches: sip_inf_loop.patch uploaded by mmichelson (license 60) Tested by: aragon ........ ................ 2009-06-21 16:15 +0000 [r202260-202264] Russell Bryant * cdr/cdr_manager.c, /: Merged revisions 202262 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r202262 | russell | 2009-06-21 11:11:48 -0500 (Sun, 21 Jun 2009) | 2 lines Fix possibility of crashiness during reload in custom fields handling. ........ * cdr/cdr_manager.c, /: Merged revisions 202258 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r202258 | russell | 2009-06-21 11:00:23 -0500 (Sun, 21 Jun 2009) | 2 lines Standardize return values of load_config() so reload() doesn't report an error on success. ........ 2009-06-20 19:14 +0000 [r202185] Sean Bright * /, apps/app_fax.c: Merged revisions 202183 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r202183 | seanbright | 2009-06-20 15:09:47 -0400 (Sat, 20 Jun 2009) | 5 lines Fix version detection for API changes in spandsp. (closes issue #15355) Reported by: deuffy ........ 2009-06-19 21:08 +0000 [r202008] Matthew Nicholson * channels/chan_sip.c: Added deadlock protection to try_suggested_sip_codec in chan_sip.c. Review: https://reviewboard.asterisk.org/r/287/ 2009-06-19 20:26 +0000 [r201996] David Vossel * channels/chan_iax2.c, /: Merged revisions 201994 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r201994 | dvossel | 2009-06-19 15:24:37 -0500 (Fri, 19 Jun 2009) | 14 lines Merged revisions 201993 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201993 | dvossel | 2009-06-19 15:22:02 -0500 (Fri, 19 Jun 2009) | 8 lines timestamp was being converted to host order as a short rather than a long (closes issue #15361) Reported by: ffloimair Patches: ts_issue.diff uploaded by dvossel (license 671) ........ ................ 2009-06-19 15:48 +0000 [r201784-201905] Tilghman Lesher * res/res_config_odbc.c, /: Merged revisions 201904 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201904 | tilghman | 2009-06-19 10:47:55 -0500 (Fri, 19 Jun 2009) | 4 lines Fix 2 typos and add support for wide character types. Reported by Benny Amorsen via the asterisk-users mailing list. http://lists.digium.com/pipermail/asterisk-users/2009-June/233622.html ........ * main/features.c: If the "h" extension fails, give it another chance in main/pbx.c. If the "h" extension fails, give it another chance in main/pbx.c, when it returns from the bridge code. Fixes an issue where the "h" extension may occasionally not fire, when a Dial is executed from a Macro. Debugged in #asterisk with user tompaw. * /, apps/Makefile: Merged revisions 201783 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201783 | tilghman | 2009-06-18 15:52:36 -0500 (Thu, 18 Jun 2009) | 6 lines One of the changes in 1.6.1 was to allow app_directory to use functionality within app_voicemail for directory functions. It is therefore no longer necessary for app_directory to be linked against the ODBC libraries (and it never was necessary for app_directory to be linked against IMAP, though it was). ........ 2009-06-18 16:51 +0000 [r201680] David Vossel * channels/misdn/isdn_lib.c, utils/conf2ael.c, main/ast_expr2.c, utils/stereorize.c, main/ast_expr2f.c, res/ael/ael_lex.c, utils/ael_main.c, utils/extconf.c, channels/xpmr/xpmr.c, pbx/pbx_config.c, res/res_config_ldap.c, apps/app_rpt.c, main/asterisk.c, codecs/gsm/src/gsm_destroy.c, /, channels/h323/ast_h323.cxx: Merged revisions 201678 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201678 | dvossel | 2009-06-18 11:37:42 -0500 (Thu, 18 Jun 2009) | 11 lines fixes some memory leaks and redundant conditions (closes issue #15269) Reported by: contactmayankjain Patches: patch.txt uploaded by contactmayankjain (license 740) memory_leak_stuff.trunk.diff uploaded by dvossel (license 671) Tested by: contactmayankjain, dvossel ........ 2009-06-18 15:36 +0000 [r201613] Russell Bryant * /, res/res_musiconhold.c: Merged revisions 201610 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r201610 | russell | 2009-06-18 10:27:10 -0500 (Thu, 18 Jun 2009) | 36 lines Merged revisions 201600 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201600 | russell | 2009-06-18 10:24:31 -0500 (Thu, 18 Jun 2009) | 29 lines Fix memory corruption and leakage related reloads of non files mode MoH classes. For Music on Hold classes that are not files mode, meaning that we are executing an application that will feed us audio data, we use a thread to monitor the external application and read audio from it. This thread also makes use of the MoH class object. In the MoH class destructor, we used pthread_cancel() to ask the thread to exit. Unfortunately, the code did not wait to ensure that the thread actually went away. What needed to be done is a pthread_join() to ensure that the thread fully cleans up before we proceed. By adding this one line, we resolve two significant problems: 1) Since the thread was never joined, it never fully goes away. So, on every reload of non-files mode MoH, an unused thread was sticking around. 2) There was a race condition here where the application monitoring thread could still try to access the MoH class, even though the thread executing the MoH reload has already destroyed it. (issue #15109) Reported by: jvandal (issue #15123) Reported by: axisinternet (issue #15195) Reported by: amorsen (issue AST-208) ........ ................ 2009-06-18 15:24 +0000 [r201601] David Vossel * /, channels/chan_sip.c: Merged revisions 201570 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201570 | dvossel | 2009-06-18 10:16:05 -0500 (Thu, 18 Jun 2009) | 11 lines parsing extension correctly from sip register lines If a transport type was specified, but no extension, parsing of the extension would return whatever was after the transport rather than defaulting to 's'. (closes issue #15111) Reported by: ffs Patches: chan_sip.c_register-parser.patch uploaded by ffs (license 730) Tested by: ffs, dvossel ........ 2009-06-17 21:32 +0000 [r201532] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 201531 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201531 | tilghman | 2009-06-17 16:31:39 -0500 (Wed, 17 Jun 2009) | 7 lines Initialize additional variables, to prevent a possible crash. (closes issue #15186) Reported by: ajohnson Patches: 20090528__issue15186.diff.txt uploaded by tilghman (license 14) Tested by: ajohnson ........ 2009-06-17 20:11 +0000 [r201460-201464] Mark Michelson * /, channels/chan_sip.c: Merged revisions 201462 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201462 | mmichelson | 2009-06-17 15:10:01 -0500 (Wed, 17 Jun 2009) | 12 lines Fix problem with no audio due to ignoring the SDP. A recent change to our SDP version comparison made audio not function on some calls. This was because of a test wherein we were trying to see if an unsigned value was less than 0. This is a dumb comparison and arguably the compiler should have warned about it. Alas, though, it slipped past. Now it's fixed by changing the variable to be a signed type. Found by several developers. Tested by mnicholson and dbrooks. ........ * main/channel.c, /: Merged revisions 201458 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r201458 | mmichelson | 2009-06-17 15:04:12 -0500 (Wed, 17 Jun 2009) | 15 lines Merged revisions 201450 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201450 | mmichelson | 2009-06-17 14:59:31 -0500 (Wed, 17 Jun 2009) | 9 lines Change the datastore traversal in ast_do_masquerade to use a safe list traversal. It is possible for datastore fixup functions to remove the datastore from the list and free it. In particular, the queue_transfer_fixup in app_queue does this. While I don't yet know of this causing any crashes, it certainly could. Found while discussing a separate issue with Brian Degenhardt. ........ ................ 2009-06-17 20:01 +0000 [r201448-201456] David Vossel * doc/datastores.txt, /: Merged revisions 201453 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201453 | dvossel | 2009-06-17 15:00:51 -0500 (Wed, 17 Jun 2009) | 3 lines ast_channel_datastore_alloc is no longer used. updating datastores.txt to reflect that. ........ * apps/app_mixmonitor.c, /: Merged revisions 201445 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r201445 | dvossel | 2009-06-17 14:45:35 -0500 (Wed, 17 Jun 2009) | 25 lines Merged revisions 201423 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201423 | dvossel | 2009-06-17 14:28:12 -0500 (Wed, 17 Jun 2009) | 19 lines StopMixMonitor race condition (not giving up file immediately) StopMixMonitor only indicates to the MixMonitor thread to stop writing to the file. It does not guarantee that the recording's file handle is available to the dialplan immediately after execution. This results in a race condition. To resolve this, the filestream pointer is placed in a datastore on the channel. When StopMixMonitor is called, the datastore is retrieved from the channel and the filestream is closed immediately before returning to the dialplan. Documentation indicating the use of StopMixMonitor to free files has been updated as well. (closes issue #15259) Reported by: travisghansen Tested by: dvossel Review: https://reviewboard.asterisk.org/r/283/ ........ ................ 2009-06-17 19:39 +0000 [r201444] David Brooks * /, channels/chan_sip.c: Merged revisions 201381 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r201381 | dbrooks | 2009-06-17 14:15:07 -0500 (Wed, 17 Jun 2009) | 16 lines Merged revisions 201380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201380 | dbrooks | 2009-06-17 13:45:50 -0500 (Wed, 17 Jun 2009) | 9 lines Checks for NULL sip_pvt pointer in chan_sip.c->acf_channel_read() Zombie channels could be passed, and chan_sip.c wasn't checking for it. Could crash Asterisk. Now checking for NULL pointer. (closes issue #15330) Reported by: okrief Tested by: dbrooks ........ ................ 2009-06-17 15:32 +0000 [r201365] David Vossel * /, channels/chan_sip.c: Merged revisions 201344 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201344 | dvossel | 2009-06-17 10:20:26 -0500 (Wed, 17 Jun 2009) | 16 lines SIP registry ref count error During a sip reload, the list of sip_registry objects are supposed to be traversed, unlinked, and destroyed, but destruction never takes place due to a ref counting error. This causes a memory leak when registry items are removed from sip.conf and reloaded. While the registries are removed from the global list, they are not removed from the scheduler. Because of this, SIP register attempts continue to be sent out for the item even though it may no longer be in the .conf. (closes issue #15295) Reported by: amorsen Review: https://reviewboard.asterisk.org/r/282/ ........ 2009-06-17 12:05 +0000 [r201264] Kevin P. Fleming * /, include/asterisk/linkedlists.h: Merged revisions 201262 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r201262 | kpfleming | 2009-06-17 07:04:17 -0500 (Wed, 17 Jun 2009) | 15 lines Merged revisions 201261 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201261 | kpfleming | 2009-06-17 07:03:25 -0500 (Wed, 17 Jun 2009) | 9 lines Correct AST_LIST_APPEND_LIST behavior when list to be appended is empty. When the list to be appended is empty, and the list to be appended to is *not*, AST_LIST_APPEND_LIST would actually cause the target list to become broken, and no longer have a pointer to its last entry. This patch fixes the problem. (reported by Stanislaw Pitucha on the asterisk-dev mailing list) ........ ................ 2009-06-16 22:31 +0000 [r201225] David Vossel * /, channels/chan_sip.c: Merged revisions 201223 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201223 | dvossel | 2009-06-16 17:29:30 -0500 (Tue, 16 Jun 2009) | 2 lines fix issue with build_contact introduced by the "SIP trasnport type issues" commit ........ 2009-06-16 19:42 +0000 [r200989-201096] Kevin P. Fleming * include/asterisk/frame.h, apps/app_chanspy.c, apps/app_mixmonitor.c, main/channel.c, main/autoservice.c, main/frame.c, /, apps/app_meetme.c, main/slinfactory.c, include/asterisk/linkedlists.h, main/file.c, include/asterisk/channel.h: Merged revisions 201056 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r201056 | kpfleming | 2009-06-16 13:54:30 -0500 (Tue, 16 Jun 2009) | 18 lines Merged revisions 200991 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun 2009) | 11 lines Improve support for media paths that can generate multiple frames at once. There are various media paths in Asterisk (codec translators and UDPTL, primarily) that can generate more than one frame to be generated when the application calling them expects only a single frame. This patch addresses a number of those cases, at least the primary ones to solve the known problems. In addition it removes the broken TRACE_FRAMES support, fixes a number of bugs in various frame-related API functions, and cleans up various code paths affected by these changes. https://reviewboard.asterisk.org/r/175/ ........ ................ * /, configure, autoconf/ast_gcc_attribute.m4, configure.ac: Merged revisions 201090 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r201090 | kpfleming | 2009-06-16 14:27:12 -0500 (Tue, 16 Jun 2009) | 5 lines Another minor fix to compiler attribute checking. Defaulting to 'static' for the function scope was bad... so remove it. ........ * /, configure, autoconf/ast_gcc_attribute.m4, configure.ac: Merged revisions 200985 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200985 | kpfleming | 2009-06-16 11:32:36 -0500 (Tue, 16 Jun 2009) | 7 lines Fix problems with new compiler attribute checking in configure script. The last changes to ast_gcc_attribute.m4 caused some problems checking for various attributes, because the scope of the symbol the attribute is applied to can be important; this patch allows the scope to be specified for the check. ........ 2009-06-16 16:34 +0000 [r200987] David Vossel * /, channels/chan_sip.c: Merged revisions 200946 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200946 | dvossel | 2009-06-16 11:03:30 -0500 (Tue, 16 Jun 2009) | 32 lines SIP transport type issues What this patch addresses: 1. ast_sip_ouraddrfor() by default binds to the UDP address/port reguardless if the sip->pvt is of type UDP or not. Now when no remapping is required, ast_sip_ouraddrfor() checks the sip_pvt's transport type, attempting to set the address and port to the correct TCP/TLS bindings if necessary. 2. It is not necessary to send the port number in the Contact header unless the port is non-standard for the transport type. This patch fixes this and removes the todo note. 3. In sip_alloc(), the default dialog built always uses transport type UDP. Now sip_alloc() looks at the sip_request (if present) and determines what transport type to use by default. 4. When changing the transport type of a sip_socket, the file descriptor must be set to -1 and in some cases the tcptls_session's ref count must be decremented and set to NULL. I've encountered several issues associated with this process and have created a function, set_socket_transport(), to handle the setting of the socket type. (closes issue #13865) Reported by: st Patches: dont_add_port_if_tls.patch uploaded by Kristijan (license 753) 13865.patch uploaded by mmichelson (license 60) tls_port_v5.patch uploaded by vrban (license 756) transport_issues.diff uploaded by dvossel (license 671) Tested by: mmichelson, Kristijan, vrban, jmacz, dvossel Review: https://reviewboard.asterisk.org/r/278/ ........ 2009-06-16 16:04 +0000 [r200947] Michiel van Baak * apps/app_voicemail.c, /: Merged revisions 200943 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200943 | mvanbaak | 2009-06-16 17:51:36 +0200 (Tue, 16 Jun 2009) | 9 lines add FILE_STORAGE to Voicemail Build Options Voicemail can only use one storage module at the moment. Because it's unclear that selecting one of the storage modules in menuselect will disable filesystem storage we now have a FILE_STORAGE option that conflicts with the other modules. (closes issue #15333) ........ 2009-06-16 01:32 +0000 [r200707-200766] Kevin P. Fleming * /, configure, include/asterisk/autoconfig.h.in, autoconf/ast_gcc_attribute.m4: Merged revisions 200764 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200764 | kpfleming | 2009-06-15 20:28:08 -0500 (Mon, 15 Jun 2009) | 11 lines Ensure that configure-script testing for compiler attributes actually works. The configure script tests for compiler attributes didn't actually enable enough warnings or provide a proper test harness to determine whether the compiler supports the attribute in question or not; this caused gcc 4.1 to report that it supports 'weakref', but it doesn't actually support it in the way that is needed for our optional API mechanism. The new configure script test will properly distinguish between full support and partial support for this attribute, among others. ........ * CHANGES, /: Merged revisions 200726 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200726 | kpfleming | 2009-06-15 20:03:22 -0500 (Mon, 15 Jun 2009) | 6 lines Document the new automatic 'ignoresdpversion' behavior. Asterisk will now automatically ignore incorrect incoming SDP version numbers when necessary to complete a T.38 re-INVITE operation. ........ * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 165180,200689 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r165180 | mnicholson | 2008-12-17 12:49:12 -0600 (Wed, 17 Dec 2008) | 14 lines This patch adds a new 'ignoresdpversion' option to sip.conf. When this is enabled (either globally or for a specific peer), chan_sip will treat any SDP data it receives as new data and update the media stream accordingly. By default, Asterisk will only modify the media stream if the SDP session version received is different from the current SDP session version. This option is required to interoperate with devices that have non-standard SDP session version implementations (observed by toc on the bug tracker with Microsoft OCS which always uses 0 as the session version). http://reviewboard.digium.com/r/94/ (closes issue #13958) Reported by: toc Tested by: toc ........ r200689 | kpfleming | 2009-06-15 15:42:38 -0500 (Mon, 15 Jun 2009) | 12 lines Accept T.38 re-INVITE responses with invalid SDP versions. This commit changes the 'incoming SDP version' check logic a bit more; when 'ignoresdpversion' is *not* set for a peer, if we initiate a re-INVITE to switch to T.38, we'll always accept the peer's SDP response, even if they don't properly increment the SDP version number as they should. If this situation occurs, a warning message will be generated suggesting that the peer's configuration be changed to include the 'ignoresdpversion' configuration option (although ideally they'd fix their SIP implementation to be RFC compliant). AST-221 ........ 2009-06-15 15:23 +0000 [r200516] Mark Michelson * /, channels/chan_sip.c: Merged revisions 200514 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r200514 | mmichelson | 2009-06-15 10:22:11 -0500 (Mon, 15 Jun 2009) | 11 lines Merged revisions 200513 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200513 | mmichelson | 2009-06-15 10:21:46 -0500 (Mon, 15 Jun 2009) | 5 lines Add INFO to our allowed methods so that endpoints know they may send it to us. AST-223 ........ ................ 2009-06-12 19:08 +0000 [r200363] Mark Michelson * main/channel.c, /: Merged revisions 200361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r200361 | mmichelson | 2009-06-12 14:07:51 -0500 (Fri, 12 Jun 2009) | 16 lines Merged revisions 200360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200360 | mmichelson | 2009-06-12 14:06:41 -0500 (Fri, 12 Jun 2009) | 10 lines Suppress a warning message and give a better return code when generating inband ringing after a call is answered. (closes issue #15158) Reported by: madkins Patches: 15158.patch uploaded by mmichelson (license 60) Tested by: madkins ........ ................ 2009-06-11 22:44 +0000 [r200229] Sean Bright * Makefile, /: Merged revisions 199781 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r199781 | seanbright | 2009-06-09 14:08:53 -0400 (Tue, 09 Jun 2009) | 2 lines Fix all of the parallel build warnings issued when running make -j#. ........ 2009-06-11 21:25 +0000 [r200171] Terry Wilson * main/rtp.c: Don't access rtp->rtcp->* if rtp->rtcp is null 2009-06-11 21:18 +0000 [r200152] Mark Michelson * /, channels/chan_sip.c: Merged revisions 200146 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200146 | mmichelson | 2009-06-11 16:17:14 -0500 (Thu, 11 Jun 2009) | 5 lines Fix a crash due to a potentially NULL p->options. Thanks to mnicholson for pointing it out. ........ 2009-06-11 12:16 +0000 [r200041] Leif Madsen * build_tools/make_version_h, /, build_tools/make_version_c: Merged revisions 200039 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r200039 | lmadsen | 2009-06-11 08:15:09 -0400 (Thu, 11 Jun 2009) | 8 lines Fix path for .flavor and .version (issue #14737) Reported by: davidw Patches: flavor.patch uploaded by davidw (license 780) Tested by: davidw ........ 2009-06-10 20:35 +0000 [r199996] David Brooks * main/pbx.c, /: Fixes the argument order in definition of new_find_extension(). In the definition of new_find_extension(), the arguments 'callerid' and 'label' were swapped. The prototype declaration and all calls to the function are ordered 'callerid' then 'label', but the function itself was ordered 'label' then 'callerid'. (closes issue #15303) Reported by: JimDickenson 2009-06-10 20:18 +0000 [r199963] Mark Michelson * /, channels/chan_sip.c: Merged revisions 199958 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r199958 | mmichelson | 2009-06-10 15:15:48 -0500 (Wed, 10 Jun 2009) | 6 lines Only try to use the invite_branch on outgoing INVITEs with auth credentials. I have added a comment to the code to help ease understanding of the logic here as well. ........ 2009-06-10 16:13 +0000 [r199859] Sean Bright * include/asterisk/utils.h, /: Merged revisions 199857 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r199857 | seanbright | 2009-06-10 12:10:23 -0400 (Wed, 10 Jun 2009) | 9 lines Merged revisions 199856 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199856 | seanbright | 2009-06-10 12:08:35 -0400 (Wed, 10 Jun 2009) | 2 lines __WORDSIZE is not available on all platforms, so use sizeof(void *) instead. ........ ................ 2009-06-09 20:50 +0000 [r199745-199820] David Vossel * /, channels/chan_sip.c: Merged revisions 199818 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r199818 | dvossel | 2009-06-09 15:47:57 -0500 (Tue, 09 Jun 2009) | 11 lines CLI NOTIFY sending wrong transport type. SIP's cli NOTIFY command only used UDP rather than copying the transport type from the peer. (closes issue #15283) Reported by: jthurman Patches: sip-notify-tcp-svn199728.patch uploaded by jthurman (license 614) Tested by: jthurman, dvossel ........ * main/loader.c, /, res/res_timing_pthread.c, include/asterisk/module.h, res/res_timing_dahdi.c: Merged revisions 199743 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r199743 | dvossel | 2009-06-09 11:22:04 -0500 (Tue, 09 Jun 2009) | 11 lines module load priority This patch adds the option to give a module a load priority. The value represents the order in which a module's load() function is initialized. The lower the value, the higher the priority. The value is only checked if the AST_MODFLAG_LOAD_ORDER flag is set. If the AST_MODFLAG_LOAD_ORDER flag is not set, the value will never be read and the module will be given the lowest possible priority on load. Since some modules are reliant on a timing interface, the timing modules have been given a high load priorty. (closes issue #15191) Reported by: alecdavis Tested by: dvossel Review: https://reviewboard.asterisk.org/r/262/ ........ 2009-06-08 19:39 +0000 [r199633] Sean Bright * include/asterisk/utils.h, /: Merged revisions 199630 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r199630 | seanbright | 2009-06-08 15:33:09 -0400 (Mon, 08 Jun 2009) | 32 lines Merged revisions 199626,199628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199626 | seanbright | 2009-06-08 15:24:32 -0400 (Mon, 08 Jun 2009) | 21 lines Increase the size of our thread stack on 64 bit processors. We were setting the stack size for each thread to 240KB regardless of architecture, which meant that in some scenarios we actually had less available stack space on 64 bit processors (pointers use 8 bytes instead of 4). So now we calculate the stack size we reserve based on the platform's __WORDSIZE, which gives us: 32 bit -> 240KB 64 bit -> 496KB 128 bit -> 1008KB (that's right, we're ready for 128 bit processors) Patch typed by me but written by several members of #asterisk-dev, including Kevin, Tilghman, and Qwell. (closes issue #14932) Reported by: jpiszcz Patches: 06052009_issue14932.patch uploaded by seanbright (license 71) Tested by: seanbright ........ r199628 | seanbright | 2009-06-08 15:28:33 -0400 (Mon, 08 Jun 2009) | 2 lines Fix a typo in the stack size calculation just introduced. ........ ................ 2009-06-08 17:35 +0000 [r199590] Mark Michelson * /, channels/chan_sip.c: Recorded merge of revisions 199588 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r199588 | mmichelson | 2009-06-08 12:32:04 -0500 (Mon, 08 Jun 2009) | 9 lines Fix a deadlock that could occur when setting rtp stats on SIP calls. (closes issue #15143) Reported by: cristiandimache Patches: 15143.patch uploaded by mmichelson (license 60) Tested by: cristiandimache ........ 2009-06-05 21:32 +0000 [r199300] David Vossel * include/asterisk/devicestate.h, /, main/devicestate.c: Merged revisions 199298 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r199298 | dvossel | 2009-06-05 16:21:22 -0500 (Fri, 05 Jun 2009) | 21 lines Merged revisions 199297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199297 | dvossel | 2009-06-05 16:19:56 -0500 (Fri, 05 Jun 2009) | 14 lines Fixes issue with hints giving unexpected results. Hints with two or more devices that include ONHOLD gave unexpected results. (closes issue #15057) Reported by: p_lindheimer Patches: onhold_trunk.diff uploaded by dvossel (license 671) pbx.c.1.4.patch uploaded by p (license 558) devicestate.c.trunk.patch uploaded by p (license 671) Tested by: p_lindheimer, dvossel Review: https://reviewboard.asterisk.org/r/254/ ........ ................ 2009-06-05 13:51 +0000 [r199229] Mark Michelson * channels/chan_dahdi.c, /: Merged revisions 199227 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r199227 | mmichelson | 2009-06-05 08:51:08 -0500 (Fri, 05 Jun 2009) | 14 lines Correct "dahdi show channels" output when specifying a group. Since a DAHDI channel may belong to multiple groups, we need to use a bitwise and instead of equivalence to determine whether to display the channel information. (closes issue #15248) Reported by: gentian Patches: 15248.patch uploaded by mmichelson (license 60) Tested by: gentian ........ 2009-06-04 19:16 +0000 [r199141] David Vossel * channels/chan_iax2.c, /: Merged revisions 199139 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r199139 | dvossel | 2009-06-04 14:10:16 -0500 (Thu, 04 Jun 2009) | 9 lines Merged revisions 199138 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199138 | dvossel | 2009-06-04 14:00:15 -0500 (Thu, 04 Jun 2009) | 3 lines Additional updates to AST-2009-001 ........ ................ 2009-06-04 14:53 +0000 [r199053] Sean Bright * main/asterisk.c, main/loader.c, /, include/asterisk/_private.h: Merged revisions 199051 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r199051 | seanbright | 2009-06-04 10:31:24 -0400 (Thu, 04 Jun 2009) | 47 lines Merged revisions 199022 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun 2009) | 40 lines Safely handle AMI connections/reload requests that occur during startup. During asterisk startup, a lock on the list of modules is obtained by the primary thread while each module is initialized. Issue 13778 pointed out a problem with this approach, however. Because the AMI is loaded before other modules, it is possible for a module reload to be issued by a connected client (via Action: Command), causing a deadlock. The resolution for 13778 was to move initialization of the manager to happen after the other modules had already been lodaded. While this fixed this particular issue, it caused a problem for users (like FreePBX) who call AMI scripts via an #exec in a configuration file (See issue 15189). The solution I have come up with is to defer any reload requests that come in until after the server is fully booted. When a call comes in to ast_module_reload (from wherever) before we are fully booted, the request is added to a queue of pending requests. Once we are done booting up, we then execute these deferred requests in turn. Note that I have tried to make this a bit more intelligent in that it will not queue up more than 1 request for the same module to be reloaded, and if a general reload request comes in ('module reload') the queue is flushed and we only issue a single deferred reload for the entire system. As for how this will impact existing installations - Before 13778, a reload issued before module initialization was completed would result in a deadlock. After 13778, you simply couldn't connect to the manager during startup (which causes problems with #exec-that-calls-AMI configuration files). I believe this is a good general purpose solution that won't negatively impact existing installations. (closes issue #15189) (closes issue #13778) Reported by: p_lindheimer Patches: 06032009_15189_deferred_reloads.diff uploaded by seanbright (license 71) Tested by: p_lindheimer, seanbright Review: https://reviewboard.asterisk.org/r/272/ ........ ................ 2009-06-03 15:26 +0000 [r198826-198887] David Vossel * main/channel.c, /, main/features.c, include/asterisk/channel.h: Merged revisions 198856 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198856 | dvossel | 2009-06-02 16:17:49 -0500 (Tue, 02 Jun 2009) | 10 lines Generic call forward api, ast_call_forward() The function ast_call_forward() forwards a call to an extension specified in an ast_channel's call_forward string. After an ast_channel is called, if the channel's call_forward string is set this function can be used to forward the call to a new channel and terminate the original one. I have included this api call in both channel.c's ast_request_and_dial() and feature.c's feature_request_and_dial(). App_dial and app_queue already contain call forward logic specific for their application and options. (closes issue #13630) Reported by: festr Review: https://reviewboard.asterisk.org/r/271/ ........ * channels/chan_iax2.c, /: Merged revisions 198824 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198824 | dvossel | 2009-06-02 12:55:35 -0500 (Tue, 02 Jun 2009) | 8 lines fixes issue with channels not going down after transfer Iax2 currently does not support native bridging if the timeoutms value is set. We check for that in iax2_bridge, but then set timeoutms to 0 by default. If the timeoutms is not provided it is set to -1. By setting timeoutms to 0 it is processed causing a bridging retry loop. (closes issue #15216) Reported by: oxymoron Tested by: dvossel ........ 2009-06-02 13:50 +0000 [r198793] Joshua Colp * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 198791 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198791 | file | 2009-06-02 10:48:06 -0300 (Tue, 02 Jun 2009) | 5 lines Correct documentation for the register line, specifically where the domain should be specified. (closes issue #14367) Reported by: Nick_Lewis ........ 2009-06-01 18:44 +0000 [r198628] Tilghman Lesher * /, contrib/scripts/meetme.sql: Merged revisions 198626 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198626 | tilghman | 2009-06-01 13:40:35 -0500 (Mon, 01 Jun 2009) | 2 lines Add information for new meetme realtime fields ........ 2009-05-31 01:58 +0000 [r198441] Eliel C. Sardanons * /, res/res_timing_dahdi.c: Merged revisions 198437 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198437 | eliel | 2009-05-30 21:22:15 -0400 (Sat, 30 May 2009) | 11 lines Avoid a crash when res_timing_dahdi is unloaded but wasn't properly loaded. if dahdi_test_timer() fails, timing_funcs_handle remains NULL causing a crash when calling ast_unregister_timing_interface() with a NULL pointer. (closes issue #15234) Reported by: eliel Patches: timing_dahdi1.diff uploaded by eliel (license 64) ........ 2009-05-30 20:21 +0000 [r198373-198390] Sean Bright * res/res_jabber.c, /: Merged revisions 198375 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198375 | seanbright | 2009-05-30 16:11:33 -0400 (Sat, 30 May 2009) | 13 lines Properly terminate the receive buffer before sending to iksemel. aji_io_recv takes the maximum number of bytes to read (instead of the total buffer size), so we have to subtract 1 from our buffer size. Without this, when we receive packets that are larger than our buffer, iksemel will choke and things get wonky. (closes issue #15232) Reported by: lp0 Patches: 05302009_res_jabber.c.patch uploaded by seanbright (license 71) Tested by: seanbright, lp0 ........ * res/res_jabber.c, /: Merged revisions 198371 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r198371 | seanbright | 2009-05-30 15:38:58 -0400 (Sat, 30 May 2009) | 19 lines Merged revisions 198370 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198370 | seanbright | 2009-05-30 15:36:20 -0400 (Sat, 30 May 2009) | 12 lines Properly terminate AMI JabberSend response messages. The response message (either Error or Success) needs an extra trailing \r\n after the fields to inform the client that the message is complete. (closes issue #14876) Reported by: srt Patches: 05302009_1.4_res_jabber.c.diff uploaded by seanbright (license 71) asterisk_14876.patch uploaded by srt (license 378) trunk-14876-2.diff uploaded by phsultan (license 73) ........ ................ 2009-05-30 03:49 +0000 [r198314] Russell Bryant * res/res_smdi.c, /: Merged revisions 198312 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r198312 | russell | 2009-05-29 22:43:23 -0500 (Fri, 29 May 2009) | 12 lines Merged revisions 198311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198311 | russell | 2009-05-29 22:42:46 -0500 (Fri, 29 May 2009) | 5 lines Fix a crash that occurred when MWI SMDI messages expired. (closes issue #14561) Reported by: cmoss28 ........ ................ 2009-05-30 03:28 +0000 [r198295] Sean Bright * apps/app_dial.c, /: Merged revisions 198285 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r198285 | seanbright | 2009-05-29 23:26:06 -0400 (Fri, 29 May 2009) | 15 lines Merged revisions 198251 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198251 | seanbright | 2009-05-29 22:46:41 -0400 (Fri, 29 May 2009) | 8 lines Treat an empty FORWARD_CONTEXT the same way we treat a missing one. (closes issue #15056) Reported by: p_lindheimer Patches: 05292009_bug15056.diff uploaded by seanbright (license 71) Tested by: p_lindheimer ........ ................ 2009-05-30 02:34 +0000 [r198249] Joshua Colp * /, channels/chan_sip.c: Merged revisions 198248 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198248 | file | 2009-05-29 23:31:48 -0300 (Fri, 29 May 2009) | 2 lines When removing all packets from a dialog we also need to free the data if present. ........ 2009-05-29 23:05 +0000 [r198147-198187] Russell Bryant * /, configs/modules.conf.sample: Merged revisions 198186 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198186 | russell | 2009-05-29 18:04:31 -0500 (Fri, 29 May 2009) | 2 lines Suggesting that only a single timing module be loaded is no longer necessary. ........ * /, res/res_timing_pthread.c: Merged revisions 198183 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198183 | russell | 2009-05-29 17:33:31 -0500 (Fri, 29 May 2009) | 2 lines Improve handling of trying to ACK too many timer expirations. ........ * /, res/res_timing_pthread.c: Merged revisions 198146 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198146 | russell | 2009-05-29 15:06:59 -0500 (Fri, 29 May 2009) | 38 lines Resolve issues with choppy sound when using res_timing_pthread. The situation that caused this problem was when continuous mode was being turned on and off while a rate was set for a timing interface. A very easy way to replicate this bug was to do a Playback() from behind a Local channel. In this scenario, a rate gets set on the channel for doing file playback. At the same time, continuous mode gets turned on and off about every 20 ms as frames get queued on to the PBX side channel from the other side of the Local channel. Essentially, this module treated continuous mode and a set rate as mutually exclusive states for the timer to be in. When I dug deep enough, I observed the following pattern: 1) Set timer to tick every 20 ms. 2) Wait almost 20 ms ... 3) Continuous mode gets turned on for a queued up frame 4) Continuous mode gets turned off 5) The timer goes back to its tick per 20 ms. state but starts counting at 0 ms. 6) Goto step 2. Sometimes, res_timing_pthread would make it 20 ms and produce a timer tick, but not most of the time. This is what produced the choppy sound (or sometimes no sound at all). Now, the module treats continuous mode and a set rate as completely independent timer modes. They can be enabled and disabled independently of each other and things work as expected. (closes issue #14412) Reported by: dome Patches: issue14412.diff.txt uploaded by russell (license 2) issue14412-1.6.1.0.diff.txt uploaded by russell (license 2) Tested by: DennisD, russell ........ 2009-05-29 19:13 +0000 [r198074] Matthew Nicholson * main/cdr.c, main/channel.c, /, include/asterisk/cdr.h: Merged revisions 198072 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r198072 | mnicholson | 2009-05-29 14:04:24 -0500 (Fri, 29 May 2009) | 21 lines Merged revisions 198068 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198068 | mnicholson | 2009-05-29 13:53:01 -0500 (Fri, 29 May 2009) | 15 lines Use AST_CDR_NOANSWER instead of AST_CDR_NULL as the default CDR disposition. This change also involves the addition of an AST_CDR_FLAG_ORIGINATED flag that is used on originated channels to distinguish: them from dialed channels. (closes issue #12946) Reported by: meral Patches: null-cdr2.diff uploaded by mnicholson (license 96) Tested by: mnicholson, dbrooks (closes issue #15122) Reported by: sum Tested by: sum ........ ................ 2009-05-29 18:39 +0000 [r198065] Joshua Colp * /, main/file.c: Merged revisions 198064 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r198064 | file | 2009-05-29 15:39:04 -0300 (Fri, 29 May 2009) | 2 lines Fix a memory leak of the write buffer when writing a file. ........ 2009-05-29 18:17 +0000 [r198005] Sean Bright * Makefile, /: Merged revisions 198000 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r198000 | seanbright | 2009-05-29 14:15:15 -0400 (Fri, 29 May 2009) | 15 lines Merged revisions 197998 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197998 | seanbright | 2009-05-29 14:14:12 -0400 (Fri, 29 May 2009) | 8 lines Fix 'make config' target for Slackware. There was a missing semi-colon after the echo statement in the Makefile that was causing problems for some users. Fix suggested by reporter. (closes issue #15225) Reported by: pdavis ........ ................ 2009-05-29 16:19 +0000 [r197969] Russell Bryant * /, res/res_timing_pthread.c: Merged revisions 197960 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r197960 | russell | 2009-05-29 11:15:30 -0500 (Fri, 29 May 2009) | 2 lines Trim trailing whitespace so that I can work on this bug without it bothering me. :-) ........ 2009-05-28 23:59 +0000 [r197897] Leif Madsen * apps/app_mixmonitor.c: Update MixMonitor documentation. Updated the MixMonitor documentation for the 'b' option so that it is more obvious that you must not optimize awat the Local channel when using this option. (issue #14829) 2009-05-28 18:47 +0000 [r197700] Joshua Colp * channels/chan_iax2.c, /: Merged revisions 197697 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r197697 | file | 2009-05-28 15:45:11 -0300 (Thu, 28 May 2009) | 2 lines Fix a bug where the trunkmtu setting was not set to the default value of 1240 on load but was on reload. ........ 2009-05-28 18:26 +0000 [r197696] Eliel C. Sardanons * /, channels/chan_sip.c: Merged revisions 197621 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r197621 | eliel | 2009-05-28 12:01:48 -0400 (Thu, 28 May 2009) | 19 lines Merged revisions 197562 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197562 | eliel | 2009-05-28 11:21:32 -0400 (Thu, 28 May 2009) | 13 lines Use the address we already know when reloading a peer with nat=yes. If we already have an address for a peer, and we are reloading the sip configuration, try to use that address to contact the peer, instead of getting it from the Contact. (closes issue #15194) Reported by: ibc Patches: sip.patch uploaded by eliel (license 64) Tested by: manwe ........ ................ 2009-05-28 16:08 +0000 [r197623] David Vossel * channels/chan_iax2.c: 'iax show peer blah' now outputs whether or not peer 'blah' is in trunk mode or not. 2009-05-28 15:39 +0000 [r197545-197618] Mark Michelson * main/rtp.c, /, channels/chan_sip.c, include/asterisk/rtp.h: Merged revisions 197606 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r197606 | mmichelson | 2009-05-28 10:32:19 -0500 (Thu, 28 May 2009) | 22 lines Recorded merge of revisions 197588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197588 | mmichelson | 2009-05-28 10:27:49 -0500 (Thu, 28 May 2009) | 16 lines Allow for media to arrive from an alternate source when responding to a reinvite with 491. When we receive a SIP reinvite, it is possible that we may not be able to process the reinvite immediately since we have also sent a reinvite out ourselves. The problem is that whoever sent us the reinvite may have also sent a reinvite out to another party, and that reinvite may have succeeded. As a result, even though we are not going to accept the reinvite we just received, it is important for us to not have problems if we suddenly start receiving RTP from a new source. The fix for this is to grab the media source information from the SDP of the reinvite that we receive. This information is passed to the RTP layer so that it will know about the alternate source for media. Review: https://reviewboard.asterisk.org/r/252 ........ ................ * apps/app_chanspy.c, /, include/asterisk/audiohook.h, main/audiohook.c: Merged revisions 197543 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r197543 | mmichelson | 2009-05-28 09:58:06 -0500 (Thu, 28 May 2009) | 27 lines Merged revisions 197537 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197537 | mmichelson | 2009-05-28 09:49:13 -0500 (Thu, 28 May 2009) | 21 lines Add flags to chanspy audiohook so that audio stays in sync. There are two flags being added to the chanspy audiohook here. One is the pre-existing AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set, we ensure that the read and write slinfactories on the audiohook do not skew beyond a certain tolerance. In addition, there is a new audiohook flag added here, AST_AUDIOHOOK_SMALL_QUEUE. With this flag set, we do not allow for a slinfactory to build up a substantial amount of audio before flushing it. For this particular issue, this means that the person spying on the call will hear the conversations in real time with very little delay in the audio. (closes issue #13745) Reported by: geoffs Patches: 13745.patch uploaded by mmichelson (license 60) Tested by: snblitz ........ ................ 2009-05-28 14:54 +0000 [r197470-197540] Joshua Colp * /, main/utils.c: Merged revisions 197538 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r197538 | file | 2009-05-28 11:51:43 -0300 (Thu, 28 May 2009) | 5 lines Fix a bug in stringfields where it did not actually free the pools of memory. (closes issue #15074) Reported by: pj ........ * /, channels/chan_sip.c: Merged revisions 197467 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r197467 | file | 2009-05-28 10:47:45 -0300 (Thu, 28 May 2009) | 15 lines Merged revisions 197466 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197466 | file | 2009-05-28 10:44:58 -0300 (Thu, 28 May 2009) | 8 lines Fix a bug where the flag indicating the presence of rport would get overwritten by the nat setting. The presence of rport is now stored as a separate flag. Once the dialog is setup and authenticated (or it passes through unauthenticated) the proper nat flag is set. (closes issue #13823) Reported by: dimas ........ ................ 2009-05-28 11:40 +0000 [r197440] Gavin Henry * contrib/scripts/asterisk.ldap-schema, contrib/scripts/asterisk.ldif, doc/ldap.txt, configs/res_ldap.conf.sample: issue #15155 and issue #15156 from trunk 2009-05-27 20:11 +0000 [r197262] Sean Bright * Makefile, /: Merged revisions 197260 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r197260 | seanbright | 2009-05-27 16:08:16 -0400 (Wed, 27 May 2009) | 6 lines Use bash explicitly when calling build_tools/mkpkgconfig from the Makefile. Since we use bashisms in build_tools/mkpkgconfig, we should call on bash explicitly when running from the Makefile, otherwise we get errors during a 'make install.' (closes issue #15209) Reported by: seandarcy ........ 2009-05-27 19:29 +0000 [r197245] Tilghman Lesher * /, funcs/func_cut.c: Recorded merge of revisions 197209 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r197209 | tilghman | 2009-05-27 14:20:56 -0500 (Wed, 27 May 2009) | 12 lines Recorded merge of revisions 197194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197194 | tilghman | 2009-05-27 14:09:42 -0500 (Wed, 27 May 2009) | 5 lines Use a different determinator on whether to print the delimiter, since leading fields may be blank. (closes issue #15208) Reported by: ramonpeek Patch by me, though inspired in part by a patch from ramonpeek ........ ................ 2009-05-27 17:21 +0000 [r197145] Jeff Peeler * main/channel.c, include/asterisk/channel.h: Fix broken attended transfers The bridge was terminating immediately after the attended transfer was completed. The problem was because upon reentering ast_channel_bridge nexteventts was checked to see if it was set and if so could possibly return AST_BRIDGE_COMPLETE. (closes issue #15183) Reported by: andrebarbosa Tested by: andrebarbosa, tootai, loloski 2009-05-27 16:12 +0000 [r197091] Sean Bright * configs/smdi.conf.sample, configs/extensions.conf.sample, configs/sla.conf.sample, configs/chan_dahdi.conf.sample, /, configs/vpb.conf.sample: Merged revisions 197089 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r197089 | seanbright | 2009-05-27 12:07:57 -0400 (Wed, 27 May 2009) | 6 lines Fix references to /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf in the sample configuration files. (closes issue #15207) Reported by: seandarcy ........ 2009-05-27 15:59 +0000 [r197087] David Vossel * channels/chan_sip.c: Fixes merge issue for r196453. 2009-05-27 13:05 +0000 [r196990] Sean Bright * /, channels/chan_alsa.c: Merged revisions 196988 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r196988 | seanbright | 2009-05-27 09:02:54 -0400 (Wed, 27 May 2009) | 9 lines Display an error message when chan_alsa fails to load due to a missing or inaccessible configuration file. Before this change, when chan_alsa failed to load due to a missing or inaccessible configuration file, no message would be displayed. With this change, when chan_alsa fails to load due to a missing or inaccessible configuration file, a message will be displayed. (closes issue #14760) Reported by: Nick_Lewis Patches: chan_alsa.c-confload.patch uploaded by Nick (license 657) ........ 2009-05-26 22:42 +0000 [r196869-196947] Russell Bryant * /, autoconf/ast_check_osptk.m4 (added), configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 196946 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r196946 | russell | 2009-05-26 17:40:34 -0500 (Tue, 26 May 2009) | 8 lines Update configure script to check for OSP toolkit 3.5.0. (closes issue #14988) Reported by: tzafrir Patches: configure.ac.diff uploaded by homesick (license 91) new_ast_check_osptk.m4 uploaded by homesick (license 91) ........ * /, res/res_convert.c: Merged revisions 196843 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r196843 | russell | 2009-05-26 13:20:57 -0500 (Tue, 26 May 2009) | 16 lines Merged revisions 196826 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r196826 | russell | 2009-05-26 13:14:36 -0500 (Tue, 26 May 2009) | 9 lines Resolve a file handle leak. The frames here should have always been freed. However, out of luck, there was never any memory leaked. However, after file streams became reference counted, this code would leak the file stream for the file being read. (closes issue #15181) Reported by: jkroon ........ ................ 2009-05-26 13:46 +0000 [r196660-196723] Joshua Colp * /, channels/chan_sip.c: Merged revisions 196721 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r196721 | file | 2009-05-26 10:43:13 -0300 (Tue, 26 May 2009) | 7 lines Fix a bug where the sip unregister CLI command did not completely unregister the peer. (closes issue #15118) Reported by: alecdavis Patches: chan_sip_unregister.diff2.txt uploaded by alecdavis (license 585) ........ * contrib/scripts/safe_asterisk, /: Merged revisions 196658 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r196658 | file | 2009-05-26 10:06:50 -0300 (Tue, 26 May 2009) | 14 lines Merged revisions 196657 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r196657 | file | 2009-05-26 10:06:09 -0300 (Tue, 26 May 2009) | 7 lines Remove some bash specific stuff from safe_asterisk. (closes issue #10812) Reported by: paravoid Patches: safe_asterisk_bashism.diff uploaded by tzafrir (license 46) ........ ................ 2009-05-22 22:35 +0000 [r196453] David Vossel * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 196416 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r196416 | dvossel | 2009-05-22 16:09:45 -0500 (Fri, 22 May 2009) | 19 lines SIP set outbound transport type from Registration In sip.conf the transport option allows for the configuration of what transport types (udp, tcp, and tls) a peer will accept, but only the first type listed was used for outbound connections. This patch changes this. Now the default transport type is only used until the peer registers. When registration takes place the transport type is parsed out of the Contact header. If the Contact header's transport type is equal to one that the peer supports, the peer's default transport type for outbound connections is set to match the Contact header's type. If the Contact header's transport type is not present, then the peer's default transport type is set to match the one the peer registered with. When a peer unregisters or the registration expires, the default transport type for that peer is reset. (closes issue #12282) Reported by: rjain Patches: reg_patch_1.diff uploaded by dvossel (license 671) Tested by: dvossel (closes issue #14727) Reported by: pj Patches: reg_patch_3.diff uploaded by dvossel (license 671) Tested by: pj, dvossel Review: https://reviewboard.asterisk.org/r/249/ ........ 2009-05-22 13:58 +0000 [r196119] Joshua Colp * channels/chan_misdn.c, /: Merged revisions 196117 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r196117 | file | 2009-05-22 10:56:47 -0300 (Fri, 22 May 2009) | 12 lines Merged revisions 196116 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r196116 | file | 2009-05-22 10:54:17 -0300 (Fri, 22 May 2009) | 5 lines Fix a bug where using immediate with mISDN caused a cause code of 16 to get sent back instead of 1 if the 's' extension did not exist. (closes issue #12286) Reported by: lmamane ........ ................ 2009-05-21 19:13 +0000 [r195998] David Vossel * channels/chan_iax2.c, /: Merged revisions 195995 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r195995 | dvossel | 2009-05-21 14:11:49 -0500 (Thu, 21 May 2009) | 20 lines Merged revisions 195991 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195991 | dvossel | 2009-05-21 14:04:56 -0500 (Thu, 21 May 2009) | 14 lines Sign problem calculating timestamp for iax frame leads to no audio on the receiving peer. There are rare cases in which a frame's delivery timestamp is slightly less than the iax2_pvt's offset. This causes the pvt's timestamp to be a small negative number, but since the timestamp value is unsigned it looks like a huge positive number. This patch checks for this negative case and sets the ms to zero. A similar check is already done right below this one in the 'else' statement. (closes issue #15032) Reported by: guillecabeza Patches: chan_iax2.c.patch_timestamp uploaded by guillecabeza (license 380) Tested by: guillecabeza (closes issue #14216) Reported by: Andrey Sofronov ........ ................ 2009-05-21 16:19 +0000 [r195892] Matthew Nicholson * main/cdr.c, /, include/asterisk/cdr.h: Merged revisions 195882 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r195882 | mnicholson | 2009-05-21 10:33:55 -0500 (Thu, 21 May 2009) | 20 lines Merged revisions 195881 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195881 | mnicholson | 2009-05-21 10:25:50 -0500 (Thu, 21 May 2009) | 13 lines This commit prevents cdr records with AST_CDR_FLAG_ANSLOCKED and AST_CDR_FLAG_LOCKED from being updated in certain cases. This is accomplished by adding two functions to update the answer time and disposition of calls that checks for the proper lock flags. These functions are used in the ast_bridge_call() function so that ForkCDR(A) calls are respected. This patch also modifies the way ast_bridge_call() chooses the cdr record to base the bridged_cdr on. Previously the first unlocked cdr record would be chosen, now instead the first cdr record is chosen and forked cdr records are moved to the bridge_cdr. This allows the original cdr record and any forked cdr records to be properly updated with answer and end times. (closes issue #13797) Reported by: sh0t Tested by: sh0t (closes issue #14744) Reported by: deepesh ........ ................ 2009-05-20 23:31 +0000 [r195841] Tilghman Lesher * apps/app_stack.c, /: Merged revisions 195839 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r195839 | tilghman | 2009-05-20 18:30:05 -0500 (Wed, 20 May 2009) | 3 lines If a variable had a blank value upon the initial setting, then it would do nothing. Identified by Dmitry Andrianov via private email, fixed by me. ........ 2009-05-20 17:34 +0000 [r195638-195705] Joshua Colp * /, main/features.c: Merged revisions 195698 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r195698 | file | 2009-05-20 14:33:02 -0300 (Wed, 20 May 2009) | 12 lines Merged revisions 195688 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195688 | file | 2009-05-20 14:30:25 -0300 (Wed, 20 May 2009) | 5 lines Fix some code that wrongly assumed a pointer would always be non-NULL when dealing with CDRs after a bridge. (closes issue #15079) Reported by: barryf ........ ................ * /, apps/app_meetme.c: Merged revisions 195636 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r195636 | file | 2009-05-20 14:14:42 -0300 (Wed, 20 May 2009) | 12 lines Merged revisions 195635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195635 | file | 2009-05-20 14:14:00 -0300 (Wed, 20 May 2009) | 5 lines Fix a bug where the MeetMe option 'D' did not actually prompt for the pin. (closes issue #15050) Reported by: pmhaddad ........ ................ 2009-05-19 20:18 +0000 [r195526] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 195521 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r195521 | tilghman | 2009-05-19 15:16:01 -0500 (Tue, 19 May 2009) | 14 lines Merged revisions 195520 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195520 | tilghman | 2009-05-19 15:12:20 -0500 (Tue, 19 May 2009) | 7 lines Ensure thread keys are initialized before attempting to access them. (closes issue #14889) Reported by: jaroth Patches: app_voicemail.c.patch uploaded by msirota (license 758) Tested by: msirota, BlargMaN ........ ................ 2009-05-19 14:47 +0000 [r195451] Joshua Colp * /, channels/chan_sip.c: Merged revisions 195449 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r195449 | file | 2009-05-19 11:43:54 -0300 (Tue, 19 May 2009) | 14 lines Merged revisions 195448 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195448 | file | 2009-05-19 11:41:45 -0300 (Tue, 19 May 2009) | 7 lines Fix a bug where direct RTP setup would partially occur even when disabled if the calling channel was answered. (issue #13545) Reported by: davidw (issue #14244) Reported by: mbnwa ........ ................ 2009-05-18 21:31 +0000 [r195429] Eliel C. Sardanons * main/manager.c, /: Merged revisions 195369 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r195369 | eliel | 2009-05-18 16:49:20 -0400 (Mon, 18 May 2009) | 8 lines Fix the CLI command 'manager show command' documentation and functionality. The CLI command 'manager show command' supports passing multiple action names in the same line, but it was not allowing that because of a incorrect check in the argumentes counter. Also the documentation was updated to show that this usage of the command is possible. ........ 2009-05-18 20:54 +0000 [r195358-195372] Tilghman Lesher * apps/app_queue.c, include/asterisk/smdi.h, apps/app_voicemail.c, res/res_smdi.c, /, include/asterisk/monitor.h: Recorded merge of revisions 195370 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r195370 | tilghman | 2009-05-18 15:52:33 -0500 (Mon, 18 May 2009) | 15 lines Recorded merge of revisions 195366 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195366 | tilghman | 2009-05-18 15:24:13 -0500 (Mon, 18 May 2009) | 8 lines Add a similar dependency on SMDI for voicemail as already exists for ADSI. (closes issue #14846) Reported by: pj Patches: 20090413__bug14846__1.4.diff.txt uploaded by tilghman (license 14) 20090507__issue14846__1.6.0.diff.txt uploaded by tilghman (license 14) 20090507__issue14846__1.6.1.diff.txt uploaded by tilghman (license 14) ........ ................ * main/asterisk.c, /: Merged revisions 195320 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r195320 | tilghman | 2009-05-18 14:17:15 -0500 (Mon, 18 May 2009) | 9 lines Move the spawn of astcanary down, until after the call to daemon(3). This avoids possible conflicts with the internal implementation of daemon(3). (closes issue #15093) Reported by: tzafrir Patches: 20090513__issue15093__2.diff.txt uploaded by tilghman (license 14) Tested by: tzafrir ........ 2009-05-18 19:00 +0000 [r195318] Mark Michelson * /, apps/app_externalivr.c: Merged revisions 195316 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r195316 | mmichelson | 2009-05-18 13:58:26 -0500 (Mon, 18 May 2009) | 18 lines Fix externalivr's setvariable command so that it properly sets multiple variables. The command had a for loop that was guaranteed to only execute once since the continuation operation of the loop would set the input buffer NULL. I rewrote the loop so that its operation was more obvious, and it would set multiple variables correctly. I also reduced stack space required for the function, constified the input string, and modified the function so that it would not modify the input string while I was at it. (closes issue #15114) Reported by: chris-mac Patches: 15114.patch uploaded by mmichelson (license 60) Tested by: chris-mac ........ 2009-05-18 15:55 +0000 [r195209] Joshua Colp * main/frame.c, /: Merged revisions 195207 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r195207 | file | 2009-05-18 12:53:26 -0300 (Mon, 18 May 2009) | 14 lines Merged revisions 195206 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195206 | file | 2009-05-18 12:51:22 -0300 (Mon, 18 May 2009) | 7 lines Fix a typo which caused loss of audio when using G729 in some scenarios with a smoother present. (closes issue #15105) Reported by: bamby Patches: process-vad-correctly.diff uploaded by bamby (license 430) ........ ................ 2009-05-18 15:13 +0000 [r195167] Eliel C. Sardanons * apps/app_dial.c, main/pbx.c, /, apps/app_macro.c: Merged revisions 195162 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r195162 | eliel | 2009-05-18 10:45:23 -0400 (Mon, 18 May 2009) | 9 lines Warn about the use of the application WaitExten() within a Macro(). Update applications documentation to warn the user about the use of the WaitExten() application within a Macro(). Recommend the use of Read() instead. (closes issue #14444) Reported by: ewieling ........ 2009-05-18 13:58 +0000 [r195091-195098] Joshua Colp * main/rtp.c, /: Merged revisions 195096 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r195096 | file | 2009-05-18 10:56:16 -0300 (Mon, 18 May 2009) | 12 lines Merged revisions 195095 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195095 | file | 2009-05-18 10:53:39 -0300 (Mon, 18 May 2009) | 5 lines Fix a bug where the codecs of the called party leg were not properly sent back to the caller call leg when reinvited. (closes issue #13569) Reported by: bkw918 ........ ................ * /, channels/chan_sip.c: Merged revisions 195089 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r195089 | file | 2009-05-18 10:36:17 -0300 (Mon, 18 May 2009) | 5 lines Fix a bug where specifying an empty outboundproxy would cause packets to get sent to ourself. (closes issue #15106) Reported by: timeshell ........ 2009-05-18 13:07 +0000 [r195023] Russell Bryant * main/manager.c, /: Merged revisions 195021 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r195021 | russell | 2009-05-18 07:59:11 -0500 (Mon, 18 May 2009) | 12 lines Recorded merge of revisions 195020 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195020 | russell | 2009-05-18 07:57:46 -0500 (Mon, 18 May 2009) | 5 lines Don't try to unlock a bogus channel. (closes issue #15144) Reported by: cristiandimache ........ ................ 2009-05-15 22:46 +0000 [r194835-194876] David Vossel * channels/chan_iax2.c, /: Merged revisions 194874 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r194874 | dvossel | 2009-05-15 17:44:44 -0500 (Fri, 15 May 2009) | 23 lines Merged revisions 194873 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194873 | dvossel | 2009-05-15 17:43:13 -0500 (Fri, 15 May 2009) | 17 lines IAX2 REGAUTH loop IAX was not sending REGREJ to terminate invalid registrations. Instead it sent another REGAUTH if the authentication challenge failed. This caused a loop of REGREQ and REGAUTH frames. (Related to Security fix AST-2009-001) (closes issue #14867) Reported by: aragon Tested by: dvossel (closes issue #14717) Reported by: mobeck Patches: regauth_loop_update_patch.diff uploaded by dvossel (license 671) Tested by: dvossel ........ ................ * channels/chan_iax2.c, channels/iax2-parser.c, channels/iax2-parser.h, /, channels/iax2.h: Merged revisions 194833 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r194833 | dvossel | 2009-05-15 15:52:12 -0500 (Fri, 15 May 2009) | 24 lines Merged revisions 194557,194685 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194557 | dvossel | 2009-05-14 17:59:43 -0500 (Thu, 14 May 2009) | 10 lines IAX2 "Ghost" Channels There is a bug tracker issue where people are reporting "Ghost" channels in their 'iax2 show channels' output. The confusion is caused by channels being listed as "(NONE)" with format "unknown". These are not channels of coarse. They are usually just pending registration or poke requests, but it is confusing output. To help make sense of this I have added two columns to 'iax2 show channels'. One shows the first message which started the transaction, and the second shows the last message sent by either side of the call. This helps diagnose why the entry exists and why it may not go away. (closes issue #14207) Reported by: clive18 Review: https://reviewboard.asterisk.org/r/246/ ........ r194685 | dvossel | 2009-05-15 10:40:37 -0500 (Fri, 15 May 2009) | 6 lines Update to previous IAX2 "Ghost" Channels patch. Fixed some comments made on reviewboard for the previous patch. (issue #14207) ........ ................ 2009-05-15 18:44 +0000 [r194716-194767] Russell Bryant * configs/logger.conf.sample, /: Merged revisions 194765 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r194765 | russell | 2009-05-15 13:43:42 -0500 (Fri, 15 May 2009) | 10 lines Merged revisions 194764 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194764 | russell | 2009-05-15 13:43:18 -0500 (Fri, 15 May 2009) | 2 lines Fix some spelling fail. ........ ................ * /, codecs/g722/g722_encode.c, codecs/g722/g722_decode.c: Merged revisions 194722 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r194722 | russell | 2009-05-15 12:59:08 -0500 (Fri, 15 May 2009) | 4 lines Shuttle some bits around to address some gain issues with G.722. (closes AST-209) ........ * codecs/Makefile, codecs/g722/Makefile (removed), /: Merged revisions 194718 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r194718 | russell | 2009-05-15 12:37:12 -0500 (Fri, 15 May 2009) | 2 lines Further simplify codec_g722 build. ........ * codecs/Makefile, /: Merged revisions 194714 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r194714 | russell | 2009-05-15 12:24:39 -0500 (Fri, 15 May 2009) | 2 lines Actually force running make for g722. ........ 2009-05-14 22:30 +0000 [r194542] Kevin P. Fleming * /: Merged revisions 194520 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r194520 | kpfleming | 2009-05-14 17:26:02 -0500 (Thu, 14 May 2009) | 9 lines Merged revisions 194509 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194509 | kpfleming | 2009-05-14 17:23:49 -0500 (Thu, 14 May 2009) | 1 line Update URL to Reviewboard ........ ................ 2009-05-14 22:23 +0000 [r194507] Mark Michelson * /, channels/chan_sip.c: Merged revisions 194496 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r194496 | mmichelson | 2009-05-14 17:20:51 -0500 (Thu, 14 May 2009) | 30 lines Merged revisions 194484 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194484 | mmichelson | 2009-05-14 17:17:55 -0500 (Thu, 14 May 2009) | 24 lines Fix a race condition where a reinvite could trigger a 482 response. The loop detection/spiral detection code in chan_sip used the owner channel's state as a criterion for determining if the incoming INVITE is a looped request. The problem with this is that the INVITE-handling code happens in a different thread than the thread that marks the owner channel as being up. As a result, if a reinvite were to come in very quickly, say from another Asterisk on the same LAN, it was possible for the reinvite to arrive before the owner channel had been set to the up state. This patch corrects the problem by using the invitestate of the sip_pvt instead, since that can be guaranteed to be set correctly by the time the reinvite arrives. Since there is a switch statement further in the INVITE-handling code, the AST_STATE_RINGING state also checks the invitestate of the sip_pvt in case we should actually be treating the channel as if it were up already. (closes issue #12215) Reported by: jpyle Patches: 12215_confirmed.patch uploaded by mmichelson (license 60) Tested by: lmadsen ........ ................ 2009-05-14 17:07 +0000 [r194436] Joshua Colp * /, apps/app_meetme.c: Merged revisions 194434 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r194434 | file | 2009-05-14 14:05:33 -0300 (Thu, 14 May 2009) | 7 lines Fix a bug where the 'T' option to Meetme did not work. (closes issue #15031) Reported by: Stochastic (closes issue #13801) Reported by: justdave ........ 2009-05-13 13:41 +0000 [r194212] Joshua Colp * main/rtp.c, /: Merged revisions 194209 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r194209 | file | 2009-05-13 10:39:10 -0300 (Wed, 13 May 2009) | 18 lines Merged revisions 194208 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194208 | file | 2009-05-13 10:38:01 -0300 (Wed, 13 May 2009) | 11 lines Fix RFC2833 issues with DTMF getting duplicated and with duration wrapping over. (closes issue #14815) Reported by: geoff2010 Patches: v1-14815.patch uploaded by dimas (license 88) Tested by: geoff2010, file, dimas, ZX81, moliveras (closes issue #14460) Reported by: moliveras Tested by: moliveras ........ ................ 2009-05-13 00:54 +0000 [r194140] Tilghman Lesher * main/pbx.c, /: Merged revisions 194138 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r194138 | tilghman | 2009-05-12 19:52:49 -0500 (Tue, 12 May 2009) | 14 lines Merged revisions 194137 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194137 | tilghman | 2009-05-12 19:52:03 -0500 (Tue, 12 May 2009) | 7 lines Fix logic for how to proceed with a single digit extension. (closes issue #15091) Reported by: andrew Patches: 20090512__issue15091.diff.txt uploaded by tilghman (license 14) Tested by: andrew ........ ................ 2009-05-12 23:01 +0000 [r194062] Matthew Nicholson * apps/app_queue.c, /: Merged revisions 194057 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r194057 | mnicholson | 2009-05-12 17:32:13 -0500 (Tue, 12 May 2009) | 22 lines Merged revisions 194028 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194028 | mnicholson | 2009-05-12 17:15:45 -0500 (Tue, 12 May 2009) | 16 lines This change modifies app_queue to properly generate CDR records in failure situations. This involves setting a proper cdr disposition coresponding to the given failure condition and ensuring the proper information is stored in the cdr record. (closes issue #13691) Reported by: dferrer Tested by: mnicholson (closes issue #13637) Reported by: atis Tested by: atis ........ ................ 2009-05-12 20:51 +0000 [r193961] Mark Michelson * /, channels/chan_sip.c: Merged revisions 193954 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r193954 | mmichelson | 2009-05-12 15:28:13 -0500 (Tue, 12 May 2009) | 18 lines Update spiral support in trunk and 1.6.X to match what is in 1.4. In 1.4, a SIP spiral is treated the same way as a call forward. This works much better than what is currently in trunk and 1.6.X. The code in trunk and 1.6.X did not create a new call to the recipient of the spiral, instead trying to continue the same call. In addition to just being plain wrong, this also had the side effect of only being able to spiral calls to other SIP channels. With this in place, as long as call forwards are honored, SIP spirals will work properly. This means that it will work for outbound calls made by the Queue, Dial, and Page applications. For originated calls and spool calls, however, the spiral will not work properly until a generic call forward mechanism is introduced into Asterisk. (relates to issue #13630) ........ 2009-05-12 20:42 +0000 [r193822-193958] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 193956 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r193956 | tilghman | 2009-05-12 15:40:22 -0500 (Tue, 12 May 2009) | 13 lines Merged revisions 193955 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193955 | tilghman | 2009-05-12 15:39:21 -0500 (Tue, 12 May 2009) | 6 lines Avoid initializing routines if the authentication fails. Fixes a crash (RR) issue. (closes issue #14508) Reported by: tiziano Patches: 20090221_2_wrongmailbox.diff.txt uploaded by tiziano (license 377) ........ ................ * apps/app_voicemail.c, /: Merged revisions 193870 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r193870 | tilghman | 2009-05-12 12:29:33 -0500 (Tue, 12 May 2009) | 2 lines Convert a THREADSTORAGE object into a simple malloc'd object (as suggested by Russell on -dev) ........ * apps/app_voicemail.c, /: Recorded merge of revisions 193756 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r193756 | tilghman | 2009-05-11 17:50:47 -0500 (Mon, 11 May 2009) | 25 lines Recorded merge of revisions 193755 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193755 | tilghman | 2009-05-11 17:48:20 -0500 (Mon, 11 May 2009) | 18 lines Move 300 bytes around on the stack, to make more room for an extension buffer. This allows more concurrent extensions to be copied for a single voicemail, without creating a possibility of upsetting existing users, where a dialplan could run out of stack space where it had run fine before. Alternatively, we could have allocated off the heap, but that is a larger change and would have increased the chance for instability introduced by this change. This is really solved starting in 1.6.0.11, as the use of an ast_str buffer allows an unlimited number of extensions (up to available memory). We additionally create a new warning message when the buffer length is exceeded, permitting administrators to see an issue after the fact, whereas previously the list was silently truncated. (closes issue #14739) Reported by: p_lindheimer Patches: 20090417__bug14739.diff.txt uploaded by tilghman (license 14) Tested by: p_lindheimer ........ ................ 2009-05-11 19:16 +0000 [r193616] Richard Mudgett * channels/chan_misdn.c, /: Merged revisions 193614 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r193614 | rmudgett | 2009-05-11 14:11:29 -0500 (Mon, 11 May 2009) | 19 lines Merged revisions 193613 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193613 | rmudgett | 2009-05-11 14:09:00 -0500 (Mon, 11 May 2009) | 12 lines Sent wrong message to clear a call we started if the other end has not responed yet. In the state MISDN_CALLING (i.e. SETUP was sent but no answer has arrived yet), it is not allowed to clear the call with RELEASE_COMPLETE. It must be cleared with DISCONNECT. A RELEASE_COMPLETE is only allowed as an answer to a SETUP. (See Q.931 ch. 5.3.2, 5.3.2.a, 5.3.2.b) Patches: chan-misdn-ccstate7.patch uploaded by customer. JIRA ABE-1862 ........ ................ 2009-05-11 18:07 +0000 [r193547] Leif Madsen * /, funcs/func_channel.c: Recorded merge of revisions 193545 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r193545 | lmadsen | 2009-05-11 14:01:44 -0400 (Mon, 11 May 2009) | 14 lines Recorded merge of revisions 193544 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193544 | lmadsen | 2009-05-11 13:35:17 -0400 (Mon, 11 May 2009) | 7 lines Document CHANNEL(transfercapability) in CLI documentation. (issue #15073) Reported by: pkempgen Patches: 20090511__issue15073.diff.txt uploaded by tilghman (license 14) ........ ................ 2009-05-08 20:51 +0000 [r193389] David Vossel * /, channels/chan_sip.c: Merged revisions 193387 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r193387 | dvossel | 2009-05-08 15:32:51 -0500 (Fri, 08 May 2009) | 7 lines TCP not matching valid peer. find_peer() does not find a valid peer when using pvt->recv as the sockaddr_in argument. Because of the way TCP works, the port number in pvt->recv is not what we're looking for at all. There is currently only one place that find_peer searches for a peer using the sockaddr_in argument. If the peer is not found after using pvt->recv (works for UDP since the port number will be correct), a temp sockaddr_in struct is made using the Contact header in the sip_request. This has the correct port number in it. Review: http://reviewboard.digium.com/r/236/ ........ 2009-05-08 15:36 +0000 [r193335] Sean Bright * funcs/func_devstate.c, /: Merged revisions 193274 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r193274 | seanbright | 2009-05-08 11:18:40 -0400 (Fri, 08 May 2009) | 2 lines Fix the spelling of UNAVAILABLE in func_devstate CLI completion. ........ 2009-05-08 14:54 +0000 [r193265] David Vossel * channels/misdn_config.c, /: Merged revisions 193263 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r193263 | dvossel | 2009-05-08 09:52:19 -0500 (Fri, 08 May 2009) | 15 lines Merged revisions 193262 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193262 | dvossel | 2009-05-08 09:51:09 -0500 (Fri, 08 May 2009) | 9 lines "misdn show config" segfaults asterisk, if no MSN lists (closes issue #14976) Reported by: alecdavis Patches: misdn_config.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis, FabienToune ........ ................ 2009-05-08 14:10 +0000 [r193196] Kevin P. Fleming * configs/logger.conf.sample, /, main/logger.c: Merged revisions 193194 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r193194 | kpfleming | 2009-05-08 09:06:15 -0500 (Fri, 08 May 2009) | 13 lines Merged revisions 193193 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193193 | kpfleming | 2009-05-08 09:03:28 -0500 (Fri, 08 May 2009) | 7 lines Make absolute paths for logger channels work properly (Note: This is not a new feature, it was previously undocumented and broken.) The Asterisk logger has a feature to support absolute pathnames for logger channels, but the code implementing the feature was broken. This has been fixed, and the absolute path feature is now documented in the sample logger.conf. ........ ................ 2009-05-07 23:44 +0000 [r193122] Tilghman Lesher * main/pbx.c, /: Merged revisions 193120 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r193120 | tilghman | 2009-05-07 18:42:28 -0500 (Thu, 07 May 2009) | 26 lines Merged revisions 193119 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193119 | tilghman | 2009-05-07 18:41:11 -0500 (Thu, 07 May 2009) | 19 lines Fix Background within a Macro for FreePBX. If the single digit DTMF is an extension in the specified context, then go there and signal no DTMF. Otherwise, we should exit with that DTMF. If we're in Macro, we'll exit and seek that DTMF as the beginning of an extension in the Macro's calling context. If we're not in Macro, then we'll simply seek that extension in the calling context. Previously, someone complained about the behavior as it related to the interior of a Gosub routine, and the fix (#14011) inadvertently broke FreePBX (#14940). This change should fix both of these situations, but with the possible incompatibility that if a single digit extension does not exist (but a longer extension COULD have matched), it would have previously gone immediately to the "i" extension, but will now need to wait for a timeout. (closes issue #14940) Reported by: p_lindheimer Patches: 20090420__bug14940.diff.txt uploaded by tilghman (license 14) Tested by: p_lindheimer ........ ................ 2009-05-07 22:42 +0000 [r193079] Richard Mudgett * channels/chan_misdn.c, /: Merged revisions 193077 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r193077 | rmudgett | 2009-05-07 17:24:04 -0500 (Thu, 07 May 2009) | 12 lines Merged revisions 193050 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193050 | rmudgett | 2009-05-07 17:17:06 -0500 (Thu, 07 May 2009) | 5 lines Give a more helpful message when an incoming call's dialed extension does not match. Added the dialed extension and context to the chan_misdn messages warning that the dialed number cannot be matched in the dialplan. ........ ................ 2009-05-07 17:52 +0000 [r192935-193007] Tilghman Lesher * /, funcs/func_odbc.c: Merged revisions 193006 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r193006 | tilghman | 2009-05-07 12:51:13 -0500 (Thu, 07 May 2009) | 7 lines Second result should not contain data from the first result. (closes issue #15039) Reported by: jims Patches: 20090506__issue15039.diff.txt uploaded by tilghman (license 14) Tested by: jims ........ * channels/chan_unistim.c, /: Merged revisions 192938 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192938 | tilghman | 2009-05-07 12:13:36 -0500 (Thu, 07 May 2009) | 6 lines Send DTMF frame before playing back audio. (closes issue #14858) Reported by: barryf Patches: 20090507__bug14858.diff.txt uploaded by tilghman (license 14) ........ * /, channels/chan_sip.c: Merged revisions 192933 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r192933 | tilghman | 2009-05-07 11:43:56 -0500 (Thu, 07 May 2009) | 17 lines Merged revisions 192932 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192932 | tilghman | 2009-05-07 11:29:08 -0500 (Thu, 07 May 2009) | 10 lines Eliminate repetition of fullcontact during reconstruction. If the fullcontact field appears in both the sippeers and the sipregs table, then during reconstruction of the field, it will otherwise be doubled. (closes issue #14754) Reported by: Alexei Gradinari Patches: 20090506__bug14754.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen ........ ................ 2009-05-06 22:19 +0000 [r192869] Jeff Peeler * /, main/features.c: Merged revisions 192861 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r192861 | jpeeler | 2009-05-06 17:17:27 -0500 (Wed, 06 May 2009) | 17 lines Merged revisions 192858 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192858 | jpeeler | 2009-05-06 17:15:19 -0500 (Wed, 06 May 2009) | 10 lines Make ParkedCall application stop execution of the dialplan after hang up Just changed park_exec to always return non-zero. I really wasn't entirely sure at first if this was a bug. Decided it was since it would be surprising when not using ParkedCall in the dialplan to hang up and have dialplan execution continue. (closes issue #14555) Reported by: francesco_r ........ ................ 2009-05-06 17:53 +0000 [r192812] Matthew Fredrickson * channels/chan_dahdi.c, /: Merged revisions 190946 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r190946 | mattf | 2009-04-28 17:05:05 -0500 (Tue, 28 Apr 2009) | 1 line Make sure that we do not clear the down flag on the BRI during PTMP link transients. Also refix SS7 audio that the early media patch broke. ........ 2009-05-06 17:39 +0000 [r192636-192809] Joshua Colp * channels/chan_iax2.c, /: Merged revisions 192808 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192808 | file | 2009-05-06 14:38:51 -0300 (Wed, 06 May 2009) | 10 lines Fix a bug where a timer would be created but not acknowledged. This scenario crept up if chan_iax2 was loaded with no configuration file present. It would create a timer and tell it to go at an interval but the thread that normally acknowledges it would not be created because no configuration file was present. The timer will now be closed if no configuration file is present. (closes issue #15014) Reported by: madkins ........ * /, channels/chan_sip.c: Merged revisions 192634 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r192634 | file | 2009-05-06 10:34:35 -0300 (Wed, 06 May 2009) | 14 lines Merged revisions 192633 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192633 | file | 2009-05-06 10:30:51 -0300 (Wed, 06 May 2009) | 7 lines Update some old logic to stop both begin and end DTMF frames from reaching the core if rfc2833 is not enabled. (closes issue #15036) Reported by: dimas Patches: v1-15036.patch uploaded by dimas (license 88) ........ ................ 2009-05-05 20:02 +0000 [r192527] Sean Bright * /, static-http/astman.js: Merged revisions 192525 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r192525 | seanbright | 2009-05-05 15:57:49 -0400 (Tue, 05 May 2009) | 18 lines Merged revisions 192524 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192524 | seanbright | 2009-05-05 15:56:11 -0400 (Tue, 05 May 2009) | 11 lines Fix Javascript error when using astman.js in Internet Explorer. Internet Explorer (tested with 7.0) does not like trailing commas on constructs like object initializers, so get rid of them to avoid some errors. (closes issue #15026) Reported by: rajnishgiri Patches: bug15026.patch uploaded by seanbright (license 71) Tested by: seanbright ........ ................ 2009-05-05 18:26 +0000 [r192401-192473] Joshua Colp * /, main/features.c: Merged revisions 192462 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r192462 | file | 2009-05-05 15:23:58 -0300 (Tue, 05 May 2009) | 15 lines Merged revisions 192454 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192454 | file | 2009-05-05 15:22:27 -0300 (Tue, 05 May 2009) | 8 lines Fix an incorrect assumption that certain values on the channel will always exist when they may not. The CDR code involved with bridges wrongly assumed that the currently executing application and data values will always exist. It is possible for this to be false when call forwarding is involved. (closes issue #14984) Reported by: gincantalupo ........ ................ * apps/app_followme.c, /: Merged revisions 192430 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r192430 | file | 2009-05-05 14:46:51 -0300 (Tue, 05 May 2009) | 12 lines Merged revisions 192429 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192429 | file | 2009-05-05 14:43:30 -0300 (Tue, 05 May 2009) | 5 lines Fix a bug where the followme application would continue trying numbers after the caller hung up. (closes issue #13624) Reported by: sgenyuk ........ ................ * /, channels/chan_sip.c: Merged revisions 192387 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192387 | file | 2009-05-05 11:22:47 -0300 (Tue, 05 May 2009) | 10 lines Fix a bug with setting t38pt_udptl at the user or peer level. If an incoming call authenticated as a user or peer and t38pt_udptl was not set to yes in general then no UDPTL session would be present and any T38 related things would fail. This commit changes it so that if after authenticating T38 is enabled but no UDPTL session is present one will be created. (issue AST-215) ........ 2009-05-05 13:37 +0000 [r192281-192359] Kevin P. Fleming * main/astobj2.c, include/asterisk/stringfields.h, /, main/utils.c: Merged revisions 192357 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192357 | kpfleming | 2009-05-05 15:18:21 +0200 (Tue, 05 May 2009) | 5 lines Correct some flaws in the memory accounting code for stringfields and ao2 objects Under some conditions, the memory allocation for stringfields and ao2 objects would not have supplied valid file/function names for MALLOC_DEBUG tracking, so this commit corrects that. ........ * main/astobj2.c, main/datastore.c, main/channel.c, /, include/asterisk/astobj2.h, include/asterisk/datastore.h, include/asterisk/channel.h: Merged revisions 192318 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192318 | kpfleming | 2009-05-05 12:34:19 +0200 (Tue, 05 May 2009) | 5 lines Properly account for memory allocated for channels and datastores As in previous commits, when channels are allocated (with ast_channel_alloc) or datastores are allocated (with ast_datastore_alloc) properly account for the memory being owned by the caller, instead of the allocator function itself. ........ * include/asterisk/stringfields.h, /, main/utils.c: Merged revisions 192279 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192279 | kpfleming | 2009-05-05 10:51:06 +0200 (Tue, 05 May 2009) | 5 lines Ensure that string pools allocated to hold stringfields are properly accounted in MALLOC_DEBUG mode This commit modifies the stringfield pool allocator to remember the 'owner' of the stringfield manager the pool is being allocated for, and ensures that pools allocated in the future when fields are populated are owned by that file/function. ........ 2009-05-04 22:48 +0000 [r192216] David Vossel * channels/chan_iax2.c, /: Merged revisions 192214 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r192214 | dvossel | 2009-05-04 17:44:51 -0500 (Mon, 04 May 2009) | 17 lines Merged revisions 192213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192213 | dvossel | 2009-05-04 17:37:31 -0500 (Mon, 04 May 2009) | 11 lines global mohinterpret setting is ignored mohinterpret and mohsuggest global variables were not copied over during build_users and build_peers. (closes issue #14728) Reported by: dimas Patches: v1-14728.patch uploaded by dimas (license 88) Tested by: dimas, dvossel ........ ................ 2009-05-04 19:30 +0000 [r192172] Tilghman Lesher * /, configure, res/res_agi.c: Recorded merge of revisions 192171 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192171 | tilghman | 2009-05-04 14:29:13 -0500 (Mon, 04 May 2009) | 8 lines Restore 'asyncagi break' command to 1.6.1 and higher. (closes issue #14985) Reported by: nikkk Patches: 20090428__bug14985.diff.txt uploaded by tilghman (license 14) 20090429__bug14985__1.6.1.diff.txt uploaded by tilghman (license 14) Tested by: nikkk ........ 2009-05-04 19:20 +0000 [r192154] Kevin P. Fleming * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions 192059 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192059 | kpfleming | 2009-05-04 18:24:16 +0200 (Mon, 04 May 2009) | 5 lines Ensure that astobj2 memory allocations are properly accounted for when MALLOC_DEBUG is used This commit ensures that all astobj2 allocated objects are properly accounted for in MALLOC_DEBUG mode by passing down the file/function/line information from the module/function that actually called the astobj2 allocation function. ........ 2009-05-04 18:44 +0000 [r192134] Tilghman Lesher * autoconf/ast_ext_tool_check.m4, /: Merged revisions 192132 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192132 | tilghman | 2009-05-04 13:42:56 -0500 (Mon, 04 May 2009) | 6 lines Pass libraries in LIBS, not LDFLAGS. (closes issue #14671) Reported by: Chainsaw Patches: asterisk-1.6.0.6-toolcheck-libs-not-ldflags.patch uploaded by Chainsaw (license 723) ........ 2009-05-04 17:30 +0000 [r192094] Leif Madsen * apps/app_forkcdr.c: Resolve grammatical mistakes in the application description in app_forkcdr. (closes issue #14801) Reported by: festr 2009-05-04 10:00 +0000 [r191957] Kevin P. Fleming * /, configs/modules.conf.sample: Merged revisions 191955 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r191955 | kpfleming | 2009-05-04 11:57:36 +0200 (Mon, 04 May 2009) | 8 lines Ensure that by default only one console channel driver is loaded This configuration file was changed to ensure that only one console channel driver (chan_oss) is loaded by default, but the change would only work if chan_console was not built. Now it will work as expected; if chan_alsa or chan_console are built and installed, they will not be loaded unless explicity requested. ........ 2009-05-02 18:45 +0000 [r191777] Kevin P. Fleming * /, main/logger.c: Merged revisions 191775 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r191775 | kpfleming | 2009-05-02 20:39:48 +0200 (Sat, 02 May 2009) | 5 lines Fix an error in queue_log file rotation optimization code This code was copy-and-pasted without properly changing references to event_rotate into queue_rotate, so under some conditions the log rotation would rotate queue_log even though it was not necessary. ........ 2009-05-02 15:52 +0000 [r191702] Sean Bright * main/asterisk.c, /: Merged revisions 191700 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r191700 | seanbright | 2009-05-02 11:45:07 -0400 (Sat, 02 May 2009) | 1 line Update copyright year to 2009 ........ 2009-05-01 20:02 +0000 [r191553-191562] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 191560 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r191560 | tilghman | 2009-05-01 15:01:21 -0500 (Fri, 01 May 2009) | 13 lines Merged revisions 191559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r191559 | tilghman | 2009-05-01 15:00:23 -0500 (Fri, 01 May 2009) | 6 lines SIP Response 410 maps to cause code 22 (or 23), not 1. (closes issue #14993) Reported by: BigJimmy Patches: causepatch uploaded by BigJimmy (license 371) ........ ................ * channels/chan_iax2.c, /: Merged revisions 191494 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r191494 | tilghman | 2009-05-01 13:18:00 -0500 (Fri, 01 May 2009) | 4 lines Set debug message back to DEBUG level. (closes issue #15007) Reported by: hulber ........ 2009-05-01 18:20 +0000 [r191505] Jeff Peeler * main/channel.c, /: Merged revisions 191489 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r191489 | jpeeler | 2009-05-01 13:09:23 -0500 (Fri, 01 May 2009) | 15 lines Merged revisions 191488 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r191488 | jpeeler | 2009-05-01 12:40:46 -0500 (Fri, 01 May 2009) | 9 lines Fix DTMF not being sent to other side after a partial feature match This fixes a regression from commit 176701. The issue was that ast_generic_bridge never exited after the feature digit timeout had elapsed, which prevented the queued DTMF from being sent to the other side. This issue was reported to me directly. ........ ................ 2009-05-01 16:26 +0000 [r191454] Sean Bright * apps/app_queue.c: Fix a crash in app_queue with very long member lists. A user reported via #asterisk that with very long lists of members, a crash occurs in ast_strdupa, so just use a single buffer and ast_copy_string instead of stack allocating copys of each interface name. (Related to revision 191041 in branches/1.4) 2009-04-30 17:45 +0000 [r191223-191369] Tilghman Lesher * main/asterisk.c, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 191367 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r191367 | tilghman | 2009-04-30 12:40:58 -0500 (Thu, 30 Apr 2009) | 3 lines Detect eaccess (or euidaccess) before using it. Reported by Andrew Lindh via the -dev list. ........ * main/asterisk.c, /: Merged revisions 191283 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r191283 | tilghman | 2009-04-30 01:47:13 -0500 (Thu, 30 Apr 2009) | 11 lines Change working directory to / under certain conditions. If backgrounding and no core will be produced, then changing the directory won't break anything; likewise, if the CWD isn't accessible by the current user, then a core wasn't possible anyway. (closes issue #14831) Reported by: chris-mac Patches: 20090428__bug14831.diff.txt uploaded by tilghman (license 14) 20090430__bug14831.diff.txt uploaded by tilghman (license 14) Tested by: chris-mac ........ * /, channels/h323/ast_h323.cxx, channels/chan_h323.c: Merged revisions 191219 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r191219 | tilghman | 2009-04-29 18:06:56 -0500 (Wed, 29 Apr 2009) | 2 lines Make H.323 compile with FDLEAK detection code enabled ........ 2009-04-29 18:40 +0000 [r191138] David Brooks * pbx/pbx_config.c, /: Merged revisions 191136 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r191136 | dbrooks | 2009-04-29 13:32:58 -0500 (Wed, 29 Apr 2009) | 3 lines Removing crufty code that is no longer necessary. Code cleanup. ........ 2009-04-29 08:45 +0000 [r190988] TransNexus OSP Development * apps/app_osplookup.c, /: Merged revisions 190830 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r190830 | transnexus | 2009-04-28 17:10:42 +0800 (Tue, 28 Apr 2009) | 2 lines Updated for OSP Toolkit 3.5. ........ 2009-04-28 17:33 +0000 [r190906] Tilghman Lesher * doc/tex/cdrdriver.tex, /: Merged revisions 190904 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r190904 | tilghman | 2009-04-28 12:31:43 -0500 (Tue, 28 Apr 2009) | 2 lines UniqueID column has a maximum size of 150 ........ 2009-04-28 14:13 +0000 [r190731-190863] Kevin P. Fleming * /, Makefile.rules: Merged revisions 190861 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r190861 | kpfleming | 2009-04-28 09:12:09 -0500 (Tue, 28 Apr 2009) | 5 lines Remove Makefile rules for bison and flex sources We never, ever want these files to processed automatically, because we store the output files in Subversion and users should never need to rebuild them. ........ * /, configure, include/asterisk/autoconfig.h.in: Merged revisions 190725 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r190725 | kpfleming | 2009-04-27 14:30:54 -0500 (Mon, 27 Apr 2009) | 13 lines Merged revisions 190721 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r190721 | kpfleming | 2009-04-27 14:29:46 -0500 (Mon, 27 Apr 2009) | 7 lines Fix 'inconsistent line endings' when autoconf 2.63 is used Attempt to make configure script regeneration 'safe' using autoconf 2.63, which embeds a bare CR into the script, thus making Subversion complain about inconsistent line endings This commit changes the MIME type of the configure script to be 'binary' thus making Subversion no longer inspect line endings, and as a bonus 'svn diff' will no longer try to generate diff output for it, which is not generally useful anyway. ........ ................ 2009-04-27 19:36 +0000 [r190728] Tilghman Lesher * main/pbx.c, /: Merged revisions 190726 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r190726 | tilghman | 2009-04-27 14:34:48 -0500 (Mon, 27 Apr 2009) | 4 lines Don't warn on pipe in the System call. (closes issue #14979) Reported by: pj ........ 2009-08-10 Tilghman Lesher * Asterisk 1.6.1.4 released * AST-2009-005 2009-07-27 Leif Madsen * Asterisk 1.6.1.2 released * AST-2009-004 2009-06-05 Leif Madsen * Asterisk 1.6.1.1 released 2009-06-04 David Vossel * channels/chan_iax2.c: Additional updates for AST-2009-001 2009-06-04 David Vossel * channels/chan_iax2.c: REGAUTH loop fix related to AST-2009-001 2009-04-27 Leif Madsen * Create Asterisk 1.6.1.0 2009-04-20 Leif Madsen * Create Asterisk 1.6.1.0-rc5 2009-04-20 17:08 +0000 [r189352] Joshua Colp * /, channels/chan_sip.c: Merged revisions 189350 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r189350 | file | 2009-04-20 14:05:15 -0300 (Mon, 20 Apr 2009) | 10 lines Fix a bug with non-UDP connections that caused dialogs to not get freed. This issue crept up because of a reference count issue on non-UDP based dialogs. The dialog reference count was increased when transmitting a packet reliably but never decreased. This caused the dialog structure to hang around despite being unlinked from the dialogs container. (closes issue #14919) Reported by: vrban ........ 2009-04-20 14:06 +0000 [r189280] Mark Michelson * main/channel.c, /: Merged revisions 189278 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r189278 | mmichelson | 2009-04-20 09:05:27 -0500 (Mon, 20 Apr 2009) | 18 lines Merged revisions 189277 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r189277 | mmichelson | 2009-04-20 09:04:41 -0500 (Mon, 20 Apr 2009) | 12 lines Move the check for chan->fdno == -1 to after the zombie/hangup check. Many users were finding that their hung up channels were staying up and causing 100% CPU usage. (issue #14723) Reported by: seadweller Patches: 14723_1-4-tip.patch uploaded by mmichelson (license 60) Tested by: falves11, bamby ........ ................ 2009-04-18 01:38 +0000 [r189206] David Vossel * /, channels/chan_agent.c: Merged revisions 189204 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r189204 | dvossel | 2009-04-17 20:28:45 -0500 (Fri, 17 Apr 2009) | 18 lines Merged revisions 189203 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r189203 | dvossel | 2009-04-17 20:27:19 -0500 (Fri, 17 Apr 2009) | 12 lines Fixed autologoff in agents.conf not working when agent logs in via AgentLogin app An agent logs in by calling an extension that calls the AgentLogin app. In agents.conf ackcall=always is set, so when they get a call they have the choice to either acknowledge it or ignore it. autologoff=10 is set as well, so if the agent ignores the call over 10sec one may assume that the agent should be logged out (and in this case hungup on as well), but this was not happening. (closes issue #14091) Reported by: evandro Patches: autologoff.diff uploaded by dvossel (license 671) Review: http://reviewboard.digium.com/r/225/ ........ ................ 2009-04-17 21:55 +0000 [r189139] Richard Mudgett * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged revisions 189137 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r189137 | rmudgett | 2009-04-17 16:48:10 -0500 (Fri, 17 Apr 2009) | 17 lines Merged revisions 188833,189134 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188833 | rmudgett | 2009-04-16 16:37:58 -0500 (Thu, 16 Apr 2009) | 4 lines Only disable mISDN DSP if Asterisk DSP is enabled. Leave jitter setting alone. JIRA ABE-1835 ........ r189134 | rmudgett | 2009-04-17 16:27:55 -0500 (Fri, 17 Apr 2009) | 4 lines Modifed/added some debug messages. JIRA ABE-1835 ........ ................ 2009-04-17 20:21 +0000 [r189103] Mark Michelson * /, channels/chan_sip.c: Merged revisions 189097 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r189097 | mmichelson | 2009-04-17 15:20:23 -0500 (Fri, 17 Apr 2009) | 13 lines Prevent a crash when SIP blonde transferring an unbridged call. If one attempts to use the attended transfer button on a SIP phone to transfer an unbridged call (such as a call to an IVR) but hangs up while the target of the transfer is still ringing, we need to not crash. The problem was that ast_hangup was called from outside the channel thread. AST-211 ........ 2009-04-17 19:46 +0000 [r189080] Sean Bright * main/asterisk.c, /: Merged revisions 189077 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r189077 | seanbright | 2009-04-17 15:36:38 -0400 (Fri, 17 Apr 2009) | 1 line Fix copy/paste error with 'transmit silence' flag. ........ 2009-04-17 17:33 +0000 [r189069] Matthew Nicholson * main/pbx.c, /: Merged revisions 189010 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r189010 | mnicholson | 2009-04-17 10:44:18 -0500 (Fri, 17 Apr 2009) | 12 lines Merged revisions 189009 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r189009 | mnicholson | 2009-04-17 10:43:09 -0500 (Fri, 17 Apr 2009) | 5 lines Make Busy() application set the CDR disposition to BUSY. (closes issue #14306) Reported by: cristiandimache ........ ................ 2009-04-17 14:48 +0000 [r188940-188949] Joshua Colp * /, channels/chan_sip.c: Merged revisions 188947 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r188947 | file | 2009-04-17 11:44:56 -0300 (Fri, 17 Apr 2009) | 22 lines Merged revisions 188946 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188946 | file | 2009-04-17 11:41:25 -0300 (Fri, 17 Apr 2009) | 15 lines Fix a bug where a value used to create the channel name was bogus. This commit fixes the scenario where an incoming call is authenticated using a peer entry. Previously the channel name was created using either the username setting from the sip.conf entry or the IP address that the call came from. Now the channel name will be created using the peer name itself. This commit will not change the way the channel name is generated for users or friends. (closes issue #14256) Reported by: Nick_Lewis Patches: chan_sip.c-chname.patch uploaded by Nick (license 657) Tested by: Nick_Lewis, file ........ ................ * channels/chan_dahdi.c, /: Merged revisions 188938 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r188938 | file | 2009-04-17 11:26:53 -0300 (Fri, 17 Apr 2009) | 11 lines Merged revisions 188937 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188937 | file | 2009-04-17 11:25:57 -0300 (Fri, 17 Apr 2009) | 4 lines Fix a situation where the DAHDI channel private structure lock was not unlocked when it should have been. (issue AST-210) ........ ................ 2009-04-16 22:05 +0000 [r188776-188838] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 188836 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r188836 | tilghman | 2009-04-16 16:57:37 -0500 (Thu, 16 Apr 2009) | 14 lines Merged revisions 188835 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188835 | tilghman | 2009-04-16 16:41:13 -0500 (Thu, 16 Apr 2009) | 7 lines Only update realtime, if global option rtupdate != false (closes issue #14885) Reported by: deepesh Patches: 20090413__bug14885.diff.txt uploaded by tilghman (license 14) Tested by: deepesh ........ ................ * apps/app_voicemail.c, /: Merged revisions 188774 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r188774 | tilghman | 2009-04-16 16:03:31 -0500 (Thu, 16 Apr 2009) | 11 lines Merged revisions 188773 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188773 | tilghman | 2009-04-16 16:02:29 -0500 (Thu, 16 Apr 2009) | 4 lines Umask should not be exported into global namespace. (closes issue #14912) Reported by: jcapp ........ ................ 2009-04-15 22:12 +0000 [r188649] David Vossel * channels/chan_dahdi.c, /: Merged revisions 188647 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r188647 | dvossel | 2009-04-15 17:10:04 -0500 (Wed, 15 Apr 2009) | 18 lines Merged revisions 188646 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188646 | dvossel | 2009-04-15 17:08:40 -0500 (Wed, 15 Apr 2009) | 12 lines National prefix inserted even when caller ID not available When the caller ID is restricted, the expected behavior is for the caller id to be blank. In chan_dahdi, the national prefix is placed onto the callers number even if its restricted (empty) causing the caller id to be the national prefix rather than blank. (closes issue #13207) Reported by: shawkris Patches: national_prefix.diff uploaded by dvossel (license 671) Review: http://reviewboard.digium.com/r/220/ ........ ................ 2009-04-15 20:20 +0000 [r188473-188596] Mark Michelson * /, main/file.c: Merged revisions 188585 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r188585 | mmichelson | 2009-04-15 15:17:33 -0500 (Wed, 15 Apr 2009) | 13 lines Merged revisions 188582 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188582 | mmichelson | 2009-04-15 15:04:20 -0500 (Wed, 15 Apr 2009) | 7 lines Update ast_readvideo_callback to match ast_readaudio_callback. This fixes potential refcount errors that may occur on ast_filestreams. AST-208 ........ ................ * apps/app_queue.c, /: Merged revisions 188470 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188470 | mmichelson | 2009-04-14 18:28:13 -0500 (Tue, 14 Apr 2009) | 3 lines Fix a couple of queue member reference leaks. ........ 2009-04-14 17:43 +0000 [r188254-188415] Joshua Colp * main/rtp.c, /: Merged revisions 188413 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188413 | file | 2009-04-14 14:40:50 -0300 (Tue, 14 Apr 2009) | 5 lines Fix an incorrect clock rate when sending T140 text. (closes issue #14029) Reported by: epicac ........ * /, channels/chan_sip.c: Merged revisions 188247 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188247 | file | 2009-04-14 10:14:21 -0300 (Tue, 14 Apr 2009) | 7 lines Fix a bug with the change I made yesterday to outbound proxy support. Per discussion with oej on IRC we need the actual IP address, not the outbound proxy IP address, in the sa field. Upon further inspection this should make the behaviour of all other uses of the outbound proxy in the code. ........ 2009-04-14 05:46 +0000 [r188208-188212] Tilghman Lesher * main/pbx.c, /: Merged revisions 188210 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188210 | tilghman | 2009-04-14 00:45:13 -0500 (Tue, 14 Apr 2009) | 2 lines As suggested by Russell, warn users when their dialplan arguments contain pipes, but not commas. ........ * /, utils/smsq.c: Merged revisions 188206 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188206 | tilghman | 2009-04-14 00:27:53 -0500 (Tue, 14 Apr 2009) | 6 lines Application delimiter is ',', not '|'. (closes issue #14881) Reported by: stegro Patches: smsq.patch uploaded by stegro (license 752) ........ 2009-04-13 19:33 +0000 [r188104] Mark Michelson * /, res/res_musiconhold.c: Merged revisions 188102 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188102 | mmichelson | 2009-04-13 14:31:48 -0500 (Mon, 13 Apr 2009) | 5 lines Fix another crash related to cached realtime music on hold. This was another off-by-one problem caused by moh_register. ........ 2009-04-13 16:32 +0000 [r188069] Joshua Colp * /, channels/chan_sip.c: Merged revisions 188067 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188067 | file | 2009-04-13 13:28:06 -0300 (Mon, 13 Apr 2009) | 10 lines Fix a bug where using an outbound proxy would cause the local address to be 127.0.0.1. Copy the outbound proxy IP address into the SIP dialog structure as the IP address we will be sending to. This has to be done because the logic that determines what local IP address to use in the SIP messages is not aware of an outbound proxy being in place. It only knows what IP address we are sending to. (closes issue #12006) Reported by: mnicholson ........ 2009-04-13 14:20 +0000 [r188038] Mark Michelson * apps/app_queue.c, /: Merged revisions 188032 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188032 | mmichelson | 2009-04-13 09:17:56 -0500 (Mon, 13 Apr 2009) | 6 lines Set all queue variables on both the caller and member channels. This allows for the variables to be accessed if a member macro is run. Thanks to Grigoriy Puzankin for bringing this up on the -dev list. ........ 2009-04-10 20:28 +0000 [r187914] Jeff Peeler * channels/Makefile, /: Merged revisions 187906 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r187906 | jpeeler | 2009-04-10 15:26:46 -0500 (Fri, 10 Apr 2009) | 12 lines Fix module embedding for chan_h323. Include libchanh323.a in the modules.link file so that all the symbols can be resolved at link time. (closes issue #11966) Reported by: dome Patches: issue_11966.patch uploaded by kpfleming (license 421) Tested by: jpeeler ........ 2009-04-10 17:30 +0000 [r187767] Tilghman Lesher * contrib/scripts/sip-friends.sql, contrib/scripts/realtime_pgsql.sql, /: Merged revisions 187764 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r187764 | tilghman | 2009-04-10 12:29:34 -0500 (Fri, 10 Apr 2009) | 9 lines Merged revisions 187763 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187763 | tilghman | 2009-04-10 12:28:46 -0500 (Fri, 10 Apr 2009) | 2 lines Add lastms column to the contributed table designs ........ ................ 2009-04-10 16:54 +0000 [r187723] Kevin P. Fleming * /, build_tools/embed_modules.xml: Merged revisions 187721 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r187721 | kpfleming | 2009-04-10 11:51:44 -0500 (Fri, 10 Apr 2009) | 5 lines clean up some patterns for files to remove add embedding support for bridge and test modules ........ 2009-04-10 16:03 +0000 [r187678] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 187674 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r187674 | tilghman | 2009-04-10 10:59:40 -0500 (Fri, 10 Apr 2009) | 4 lines Ensure pvt is not NULL before dereferencing it. (closes issue #14784) Reported by: pj ........ 2009-04-10 16:00 +0000 [r187676] Russell Bryant * tests/test_heap.c, /: Merged revisions 187675 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r187675 | russell | 2009-04-10 11:00:29 -0500 (Fri, 10 Apr 2009) | 2 lines Disable test modules by default. ........ 2009-04-10 03:56 +0000 [r187600] Tilghman Lesher * main/channel.c, main/pbx.c, main/manager.c, /, include/asterisk/linkedlists.h, main/features.c, main/http.c, main/app.c, include/asterisk/lock.h, main/audiohook.c: Merged revisions 187599 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r187599 | tilghman | 2009-04-09 22:55:27 -0500 (Thu, 09 Apr 2009) | 2 lines Modify headers and macros, according to Russell's suggestions on the -dev list ........ 2009-04-09 19:14 +0000 [r187495] Mark Michelson * /, channels/chan_sip.c: Merged revisions 187488 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r187488 | mmichelson | 2009-04-09 13:58:41 -0500 (Thu, 09 Apr 2009) | 24 lines Merged revisions 187484 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187484 | mmichelson | 2009-04-09 13:51:20 -0500 (Thu, 09 Apr 2009) | 18 lines Handle a SIP race condition (reinvite before an ACK) properly. RFC 5047 explains the proper course of action to take if a reINVITE is received before the ACK from a previous invite transaction. What we are to do is to treat the reINVITE as if it were both an ACK and a reINVITE and process it normally. Later, when we receive the ACK we had been expecting, we will ignore it since its CSeq is less than the current iseqno of the sip_pvt representing this dialog. (closes issue #13849) Reported by: klaus3000 Patches: 13849_v2.patch uploaded by mmichelson (license 60) Tested by: mmichelson, klaus3000 ........ ................ 2009-04-09 18:54 +0000 [r187486] Tilghman Lesher * main/manager.c, /, include/asterisk/linkedlists.h, include/asterisk/lock.h: Merged revisions 187483 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r187483 | tilghman | 2009-04-09 13:40:01 -0500 (Thu, 09 Apr 2009) | 15 lines Merged revisions 187428 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187428 | tilghman | 2009-04-09 13:08:20 -0500 (Thu, 09 Apr 2009) | 8 lines Race condition between ast_cli_command() and 'module unload' could cause a deadlock. Add lock timeouts to avoid this potential deadlock. (closes issue #14705) Reported by: jamessan Patches: 20090320__bug14705.diff.txt uploaded by tilghman (license 14) Tested by: jamessan ........ ................ 2009-04-09 17:43 +0000 [r187427] Mark Michelson * /, res/res_musiconhold.c: Merged revisions 187421,187424 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r187421 | mmichelson | 2009-04-09 12:30:39 -0500 (Thu, 09 Apr 2009) | 21 lines Fix a crash in res_musiconhold when using cached realtime moh. The moh_register function links an mohclass and then immediately unrefs the class since the container now has a reference. The problem with using realtime music on hold is that the class is allocated, registered, and started in one fell swoop. The refcounting logic resulted in the count being off by one. The same problem did not happen when using a static config because the allocation and registration of an mohclass is a separate operation from starting moh. This also did not affect non-cached realtime moh because the classes are not registered at all. I also have modified res_musiconhold to use the _t_ variants of the ao2_ functions so that more info can be gleaned when attempting to trace the refcounts. I found this to be incredibly helpful for debugging this issue and there's no good reason to remove it. (closes issue #14661) Reported by: sum ........ r187424 | mmichelson | 2009-04-09 12:34:39 -0500 (Thu, 09 Apr 2009) | 3 lines Use safe macro practices even though they really aren't necessary. ........ 2009-04-09 17:22 +0000 [r187305-187388] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 187381 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r187381 | tilghman | 2009-04-09 12:20:49 -0500 (Thu, 09 Apr 2009) | 4 lines Allow '/' in username portion of register; this is a regression. (closes issue #14668) Reported by: Netview ........ * /, channels/chan_sip.c, apps/app_sendtext.c: Merged revisions 187363 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r187363 | tilghman | 2009-04-09 11:39:43 -0500 (Thu, 09 Apr 2009) | 10 lines Merged revisions 187362 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187362 | tilghman | 2009-04-09 11:38:37 -0500 (Thu, 09 Apr 2009) | 3 lines Permit zero-length text messages in SIP. (Related to an issue posted to the -users list, subject "AEL2, BASE64_DECODE and hexadecimal") ........ ................ * main/asterisk.c, agi/Makefile, build_tools/cflags.xml, utils/Makefile, include/asterisk.h, /, main/Makefile, main/file.c, main/astfd.c (added): Merged revisions 187302 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r187302 | tilghman | 2009-04-08 23:59:05 -0500 (Wed, 08 Apr 2009) | 14 lines Merged revisions 187300-187301 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187300 | tilghman | 2009-04-08 23:31:38 -0500 (Wed, 08 Apr 2009) | 3 lines Add debugging mode for diagnosing file descriptor leaks. (Related to issue #14625) ........ r187301 | tilghman | 2009-04-08 23:32:40 -0500 (Wed, 08 Apr 2009) | 2 lines Oops, missed this file in the last commit. ........ ................ 2009-04-08 16:53 +0000 [r186987-187048] Mark Michelson * /, res/res_musiconhold.c: Merged revisions 187046 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r187046 | mmichelson | 2009-04-08 11:52:20 -0500 (Wed, 08 Apr 2009) | 16 lines Merged revisions 187045 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187045 | mmichelson | 2009-04-08 11:52:03 -0500 (Wed, 08 Apr 2009) | 10 lines Fix a small logical error when loading moh classes. We were unconditionally incrementing the number of mohclasses registered. However, we should actually only increment if the call to moh_register was successful. While this probably has never caused problems, I noticed it and decided to fix it anyway. ........ ................ * main/channel.c, /: Merged revisions 186985 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186985 | mmichelson | 2009-04-08 10:27:41 -0500 (Wed, 08 Apr 2009) | 30 lines Merged revisions 186984 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186984 | mmichelson | 2009-04-08 10:26:46 -0500 (Wed, 08 Apr 2009) | 24 lines Make a couple of changes with regards to a new message printed in ast_read(). "ast_read() called with no recorded file descriptor" is a new message added after a bug was discovered. Unfortunately, it seems there are a bunch of places that potentially make such calls to ast_read() and trigger this error message to be displayed. This commit does two things to help to make this message appear less. First, the message has been downgraded to a debug level message if dev mode is not enabled. The message means a lot more to developers than it does to end users, and so developers should take an effort to be sure to call ast_read only when a channel is ready to be read from. However, since this doesn't actually cause an error in operation and is not something a user can easily fix, we should not spam their console with these messages. Second, the message has been moved to after the check for any pending masquerades. ast_read() being called with no recorded file descriptor should not interfere with a masquerade taking place. This could be seen as a simple way of resolving issue #14723. However, I still want to try to clear out the existing ways of triggering this message, since I feel that would be a better resolution for the issue. ........ ................ 2009-04-08 05:07 +0000 [r186900] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 186899 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r186899 | tilghman | 2009-04-08 00:06:22 -0500 (Wed, 08 Apr 2009) | 2 lines Add lastms to the require API call. ........ 2009-04-08 00:10 +0000 [r186835-186844] Mark Michelson * /, formats/format_wav.c, formats/format_wav_gsm.c: Merged revisions 186842 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186842 | mmichelson | 2009-04-07 19:09:28 -0500 (Tue, 07 Apr 2009) | 14 lines Merged revisions 186841 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186841 | mmichelson | 2009-04-07 19:09:04 -0500 (Tue, 07 Apr 2009) | 8 lines Fix a few typos of the word "frequency." (closes issue #14842) Reported by: jvandal Patches: frequency-typo.diff uploaded by jvandal (license 413) ........ ................ * /, channels/chan_sip.c: Merged revisions 186837 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r186837 | mmichelson | 2009-04-07 19:01:49 -0500 (Tue, 07 Apr 2009) | 7 lines Fix bad merge from fix for issue 13867. (closes issue #14686) Reported by: davidw ........ * main/channel.c, /: Merged revisions 186833 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186833 | mmichelson | 2009-04-07 18:50:56 -0500 (Tue, 07 Apr 2009) | 15 lines Merged revisions 186832 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186832 | mmichelson | 2009-04-07 18:49:49 -0500 (Tue, 07 Apr 2009) | 8 lines Set the AST_FEATURE_WARNING_ACTIVE flag when a p2p bridge returns AST_BRIDGE_RETRY. Without this flag set, warning sounds will not be properly played to either party of the bridge. (closes issue #14845) Reported by: adomjan ........ ................ 2009-04-07 22:33 +0000 [r186806] Tilghman Lesher * /, apps/app_macro.c: Merged revisions 186799 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186799 | tilghman | 2009-04-07 17:23:46 -0500 (Tue, 07 Apr 2009) | 10 lines Merged revisions 186775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186775 | tilghman | 2009-04-07 17:16:50 -0500 (Tue, 07 Apr 2009) | 3 lines Fix Macro documentation to match current (and intended) behavior. (See -dev mailing list) ........ ................ 2009-04-07 20:53 +0000 [r186722] Mark Michelson * main/manager.c, /: Merged revisions 186720 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186720 | mmichelson | 2009-04-07 15:46:18 -0500 (Tue, 07 Apr 2009) | 12 lines Merged revisions 186719 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186719 | mmichelson | 2009-04-07 15:43:49 -0500 (Tue, 07 Apr 2009) | 6 lines Ensure that \r\n is printed after the ActionID in an OriginateResponse. (closes issue #14847) Reported by: kobaz ........ ................ 2009-04-03 20:21 +0000 [r186466] Kevin P. Fleming * channels/chan_dahdi.c, /: Merged revisions 186461 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186461 | kpfleming | 2009-04-03 15:20:01 -0500 (Fri, 03 Apr 2009) | 11 lines Merged revisions 186458 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186458 | kpfleming | 2009-04-03 15:19:20 -0500 (Fri, 03 Apr 2009) | 5 lines Fix a bug where DAHDI/Zaptel channels would not properly switch formats when requested Don't offer AST_FORMAT_SLINEAR on DAHDI/Zaptel channels... while it could provide a slight performance benefit, the translation core in Asterisk has some flaws when a channel driver offers multiple raw formats. this fix is much simpler than fixing the translation core to solve that issue (although that will be done later). ........ ................ 2009-04-03 20:04 +0000 [r186448] Tilghman Lesher * apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged revisions 186444,186447 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186444 | tilghman | 2009-04-03 14:30:34 -0500 (Fri, 03 Apr 2009) | 14 lines Merged revisions 186415 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186415 | tilghman | 2009-04-03 14:06:58 -0500 (Fri, 03 Apr 2009) | 7 lines Distinguish in a sent email between simple sends and forwards. (closes issue #11678) Reported by: jamessan Patches: 20090330__bug11678.diff.txt uploaded by tilghman (license 14) Tested by: tilghman, lmadsen ........ ................ r186447 | tilghman | 2009-04-03 14:59:55 -0500 (Fri, 03 Apr 2009) | 9 lines Merged revisions 186445 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186445 | tilghman | 2009-04-03 14:56:48 -0500 (Fri, 03 Apr 2009) | 2 lines Found a conflict in the last commit, due to multiple targets ........ ................ 2009-04-03 16:38 +0000 [r186381] David Vossel * /, main/audiohook.c: Merged revisions 186379 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r186379 | dvossel | 2009-04-03 11:29:47 -0500 (Fri, 03 Apr 2009) | 4 lines audio_audiohook_write_list() did not correctly update sample size after ast_translate. audio_audiohook_write_list() did not take into account that the sample size may change after translation depending on if the original frame is is 8khz or 16khz. the sample size is now updated after translating to reflect this possibility. This caused the audio on the receiving end to sound terrible. Thanks to jcolp and mmichelson for helping me work this out. (issue AST-197) ........ 2009-04-03 Leif Madsen * Asterisk 1.6.1.0-rc4 released. 2009-04-03 15:54 +0000 [r186323] Joshua Colp * include/asterisk/crypto.h, /: Merged revisions 186321 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186321 | file | 2009-04-03 12:52:50 -0300 (Fri, 03 Apr 2009) | 12 lines Merged revisions 186320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186320 | file | 2009-04-03 12:48:56 -0300 (Fri, 03 Apr 2009) | 5 lines Fix a problem with the crypto variable definitions not actually being defined properly. (closes issue #14804) Reported by: jvandal ........ ................ 2009-04-03 14:33 +0000 [r186288] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 186286 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r186286 | mmichelson | 2009-04-03 09:32:05 -0500 (Fri, 03 Apr 2009) | 20 lines Fix the ability to retrieve voicemail messages from IMAP. A recent change made interactive vm_states no longer get added to the list of vm_states and instead get stored in thread-local storage. In trunk and all the 1.6.X branches, the problem is that when we search for messages in a voicemail box, we would attempt to update the appropriate vm_state struct by directly searching in the list of vm_states instead of using the get_vm_state_by_imap_user function. This meant we could not find the interactive vm_state that we wanted. (closes issue #14685) Reported by: BlargMaN Patches: 14685.patch uploaded by mmichelson (license 60) Tested by: BlargMaN, qualleyiv, mmichelson ........ 2009-04-03 02:06 +0000 [r186232] Russell Bryant * cdr/cdr_radius.c, /: Merged revisions 186230 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186230 | russell | 2009-04-02 21:03:48 -0500 (Thu, 02 Apr 2009) | 29 lines Merged revisions 186229 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186229 | russell | 2009-04-02 20:57:44 -0500 (Thu, 02 Apr 2009) | 21 lines Fix a memory leak in cdr_radius. I came across this while doing some testing of my ast_channel_ao2 branch. After running a test overnight that generated over 5 million calls, Asterisk had taken up about 1 GB of my system memory. So, I re-ran the test with MALLOC_DEBUG turned on. However, it showed no leaks in Asterisk during the test, even though Asterisk was still consuming it somehow. Instead, I turned to valgrind, which when run with --leak-check=full, told me exactly where the leak came from, which was from allocations inside the radiusclient-ng library. This explains why MALLOC_DEBUG did not report it. After a bit of analysis, I found that we were leaking a little bit of memory every time a CDR record was passed to cdr_radius. I don't actually have a radius server set up to receive CDR records. However, I always have my development systems compile and install all modules. In addition to making sure there are not build errors across modules, always loading modules helps find bugs like this, too, so it is strongly recommend for all developers. ........ ................ 2009-04-02 21:59 +0000 [r186177] Mark Michelson * configs/features.conf.sample, /: Merged revisions 186175 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186175 | mmichelson | 2009-04-02 16:56:21 -0500 (Thu, 02 Apr 2009) | 11 lines Merged revisions 186174 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186174 | mmichelson | 2009-04-02 16:55:34 -0500 (Thu, 02 Apr 2009) | 5 lines Fix instructions in one-step parking comment to make more sense. Changed a capital K to a lowercase k. ........ ................ 2009-04-02 17:27 +0000 [r186108] Kevin P. Fleming * channels/chan_dahdi.c, /: Merged revisions 186101 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186101 | kpfleming | 2009-04-02 12:26:07 -0500 (Thu, 02 Apr 2009) | 9 lines Merged revisions 186081 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186081 | kpfleming | 2009-04-02 12:21:29 -0500 (Thu, 02 Apr 2009) | 3 lines ensure that the buffer passed to DAHDI_SET_BUFINFO is fully initialized ........ ................ 2009-04-02 17:14 +0000 [r186062] Tilghman Lesher * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 186060 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186060 | tilghman | 2009-04-02 12:10:28 -0500 (Thu, 02 Apr 2009) | 16 lines Merged revisions 186059 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r186059 | tilghman | 2009-04-02 12:09:13 -0500 (Thu, 02 Apr 2009) | 9 lines Merged revisions 186056 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02 Apr 2009) | 2 lines Fix for AST-2009-003 ........ ................ ................ 2009-04-02 13:53 +0000 [r185956] Kevin P. Fleming * channels/chan_dahdi.c, /: Merged revisions 185953 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185953 | kpfleming | 2009-04-02 08:51:44 -0500 (Thu, 02 Apr 2009) | 11 lines Merged revisions 185952 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185952 | kpfleming | 2009-04-02 08:43:43 -0500 (Thu, 02 Apr 2009) | 5 lines the DAHDI_GETCONF, DAHDI_SETCONF and DAHDI_GET_PARAMS ioctls were recently corrected to show that they do, in fact, read data from userspace as part of their work. due to this fix, valgrind now reports a number of cases where chan_dahdi passed an uninitialized (or partially) buffer to these ioctls, which could lead to unexpected behavior. this patch corrects chan_dahdi to ensure that buffers passed to these ioctls are always fully initialized. ........ ................ 2009-04-01 19:06 +0000 [r185848] David Vossel * /, channels/chan_sip.c: Merged revisions 185846 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185846 | dvossel | 2009-04-01 14:03:32 -0500 (Wed, 01 Apr 2009) | 16 lines Merged revisions 185845 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185845 | dvossel | 2009-04-01 14:02:00 -0500 (Wed, 01 Apr 2009) | 10 lines Fixes issue with dropped calles due to re-Invite glare and re-Invites never executing after a 491 Acknowledgement for 491 responses were never being processed because it didn't match our pending invite's seqno. Since the ACK was never processed, the 491 frame would continue to be retransmitted until eventually the call was dropped due to max retries. Now during a pending invite, if we receive another invite, we send an 491 and hold on to that glare invite's seqno in the "glareinvite" variable for that sip_pvt struct. When ACK's are received, we first check to see if it is in response to our pending invite, if not we check to see if it is in response to a glare invite. In this case, it is in response to the glare invite and must be dealt with or the call is dropped. I've changed the wait time for resending the re-Invite after receving a 491 response to comply with RFC 3261. Before this patch the scheduled re-Invite would only change a flag indicating that the re-Invite should be sent out, now it actually sends it out as well. (closes issue #12013) Reported by: alx Review: http://reviewboard.digium.com/r/213/ ........ ................ 2009-04-01 13:50 +0000 [r185774] Russell Bryant * main/channel.c, /: Merged revisions 185772 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185772 | russell | 2009-04-01 08:48:26 -0500 (Wed, 01 Apr 2009) | 14 lines Merged revisions 185771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185771 | russell | 2009-04-01 08:47:30 -0500 (Wed, 01 Apr 2009) | 6 lines Fix a case where DTMF could bypass audiohooks. This change fixes a situation where an audiohook that wants DTMF would not actually get it. This is in the code path where we end DTMF digit length emulation while handling a NULL frame. ........ ................ 2009-03-31 22:38 +0000 [r185666] Kevin P. Fleming * utils, /: Merged revisions 185664 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r185664 | kpfleming | 2009-03-31 17:35:07 -0500 (Tue, 31 Mar 2009) | 1 line ignore copied (generated) file ........ 2009-03-31 22:05 +0000 [r185471-185602] Mark Michelson * apps/app_queue.c, /: Merged revisions 185600 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185600 | mmichelson | 2009-03-31 17:02:48 -0500 (Tue, 31 Mar 2009) | 12 lines Merged revisions 185599 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185599 | mmichelson | 2009-03-31 17:00:01 -0500 (Tue, 31 Mar 2009) | 6 lines Fix crash that would occur if an empty member was specified in queues.conf. (closes issue #14796) Reported by: pida ........ ................ * apps/app_voicemail.c, /: Merged revisions 185469 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185469 | mmichelson | 2009-03-31 14:46:18 -0500 (Tue, 31 Mar 2009) | 14 lines Merged revisions 185468 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185468 | mmichelson | 2009-03-31 14:45:30 -0500 (Tue, 31 Mar 2009) | 8 lines Fix Russian voicemail intro to say the word "messages" properly. (closes issue #14736) Reported by: chappell Patches: voicemail_no_messages.diff uploaded by chappell (license 8) ........ ................ 2009-03-31 17:48 +0000 [r185427] David Brooks * /, channels/chan_gtalk.c: Merged revisions 185363 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185363 | dbrooks | 2009-03-31 11:46:57 -0500 (Tue, 31 Mar 2009) | 44 lines Merged revisions 185362 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31 Mar 2009) | 35 lines Fix incorrect parsing in chan_gtalk when xmpp contains extra whitespaces To drill into the xmpp to find the capabilities between channels, chan_gtalk calls iks_child() and iks_next(). iks_child() and iks_next() are functions in the iksemel xml parsing library that traverse xml nodes. The bug here is that both iks_child() and iks_next() will return the next iks_struct node *regardless* of type. chan_gtalk expects the next node to be of type IKS_TAG, which in most cases, it is, but in this case (a call being made from the Empathy IM client), there exists iks_struct nodes which are not IKS_TAG data (they are extraneous whitespaces), and chan_gtalk doesn't handle that case, so capabilities don't match, and a call cannot be made. iks_first_tag() and iks_next_tag(), on the other hand, will not return the very next iks_struct, but will check to see if the next iks_struct is of type IKS_TAG. If it isn't, it will be skipped, and the next struct of type IKS_TAG it finds will be returned. This assures that chan_gtalk will find the iks_struct it is looking for. This fix simply changes all calls to iks_child() and iks_next() to become calls to iks_first_tag() and iks_next_tag(), which resolves the capability matching. The following is a payload listing from Empathy, which, due to the extraneous whitespace, will not be parsed correctly by iksemel: Review: http://reviewboard.digium.com/r/181/ ........ ................ 2009-03-31 14:57 +0000 [r185263] Russell Bryant * apps/app_queue.c, /: Merged revisions 185261 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r185261 | russell | 2009-03-31 09:53:45 -0500 (Tue, 31 Mar 2009) | 5 lines Don't free() an astobj2 object. (closes issue #14672) Reported by: makoto ........ 2009-03-31 14:10 +0000 [r185199] Joshua Colp * /, main/audiohook.c: Merged revisions 185197 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185197 | file | 2009-03-31 11:07:36 -0300 (Tue, 31 Mar 2009) | 15 lines Merged revisions 185196 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185196 | file | 2009-03-31 11:06:39 -0300 (Tue, 31 Mar 2009) | 8 lines Fix crash when moving audiohooks between channels. Handle the scenario where we are called to move audiohooks between channels and the source channel does not actually have any on it. (closes issue #14734) Reported by: corruptor ........ ................ 2009-03-30 20:50 +0000 [r185126-185127] Richard Mudgett * channels/misdn_config.c, /, configs/misdn.conf.sample: Merged revisions 185123 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185123 | rmudgett | 2009-03-30 15:42:14 -0500 (Mon, 30 Mar 2009) | 9 lines Merged revisions 185121 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185121 | rmudgett | 2009-03-30 15:40:11 -0500 (Mon, 30 Mar 2009) | 1 line Update the channel allocation method documentation. ........ ................ * channels/misdn/isdn_lib.c, /: Merged revisions 185122 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185122 | rmudgett | 2009-03-30 15:41:24 -0500 (Mon, 30 Mar 2009) | 26 lines Merged revisions 185120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185120 | rmudgett | 2009-03-30 15:38:11 -0500 (Mon, 30 Mar 2009) | 19 lines Make chan_misdn BRI TE side normally defer channel selection to the NT side. Channel allocation collisions are not handled by chan_misdn very well. This patch simply avoids the problem for BRI only. For PRI, allocation collisions are still possible but less likely since there are simply more channels available and each end could use a different allocation strategy. misdn.conf options available: te_choose_channel - Use to force the TE side to allocate channels. method - Specify the channel allocation strategy. (closes issue #13488) Reported by: Christian_Pinedo Patches: isdn_lib.patch.txt uploaded by crich Tested by: crich, siepkes, festr ........ ................ 2009-03-30 16:47 +0000 [r185088] Mark Michelson * apps/app_queue.c, /: Merged revisions 185072 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185072 | mmichelson | 2009-03-30 11:26:48 -0500 (Mon, 30 Mar 2009) | 45 lines Merged revisions 185031 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185031 | mmichelson | 2009-03-30 11:17:35 -0500 (Mon, 30 Mar 2009) | 39 lines Fix queue weight behavior so that calls in low-weight queues are not inappropriately blocked. (This is copied and pasted from the review request I made for this patch) Asterisk has some odd behavior when queue weights are used. The current logic used when potentially calling a queue member is: If the member we are going to call is part of another queue and _that other queue has any callers in it_ and has a higher weight than the queue we are calling from, then don't try to contact that member. The issue here is what I have marked with underscores. If the higher-weighted queue has any callers in it at all, then the queue member will be unreachable from the lower-weighted queue. This has the potential to be really really bad if using a queue strategy, such as leastrecent or fewestcalls, with the potential to call the same member repeatedly. The fix proposed by garychen on issue 13220 is very simple and, as far as I can see, works well for this situation. With this set of changes, the logic used becomes: If the member we are going to call is part of another queue, the other queue has a higher weight than the queue we are calling from, and the higher weight queue has at least as many callers as available members, then do not try to contact the queue member. If the higher weighted queue has fewer callers than available members, then there is no reason to deny the call to this member since the other queue can afford to spare a member. Since the fix involved writing a generic function for determining the number of available members in the queue, I also modified the is_our_turn function to make use of the new num_available_members function to determine if it is our turn to try calling a member. There is one small behavior change. Before writing this patch, if you had autofill disabled, then if you were the head caller in a queue, you would automatically be told that it was your turn to try calling a member. This did not take into account whether there were actually any queue members available to take the call. Now we actually make sure there is at least one member available to take the call if autofill is disabled. (closes issue #13220) Reported by: garychen Review: http://reviewboard.digium.com/r/202/ ........ ................ 2009-03-30 14:41 +0000 [r184950] Joshua Colp * /, channels/chan_sip.c: Merged revisions 184948 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r184948 | file | 2009-03-30 11:37:47 -0300 (Mon, 30 Mar 2009) | 21 lines Merged revisions 184947 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184947 | file | 2009-03-30 11:35:47 -0300 (Mon, 30 Mar 2009) | 14 lines Improve our handling of T38 in the initial INVITE from a device. We now answer with matching media streams to what is requested. If an INVITE is received with both a T38 and RTP media stream this means we answer with both. For any outgoing calls created as a result of this inbound one no T38 is requested in the initial INVITE. Instead if we start receiving udptl packets we trigger a reinvite on the outbound side. (closes issue #12437) Reported by: marsosa Tested by: pinga-fogo, okrief, file, afu Review: http://reviewboard.digium.com/r/208/ ........ ................ 2009-03-30 13:57 +0000 [r184912] Russell Bryant * channels/h323/Makefile.in, /: Merged revisions 184910 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184910 | russell | 2009-03-30 08:55:44 -0500 (Mon, 30 Mar 2009) | 4 lines Fix build error when chan_h323 is not being built. (reported by cai1982 in #asterisk-dev) ........ 2009-03-29 05:52 +0000 [r184840-184845] Russell Bryant * apps/app_followme.c, /: Merged revisions 184843 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r184843 | russell | 2009-03-29 00:52:20 -0500 (Sun, 29 Mar 2009) | 13 lines Merged revisions 184842 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184842 | russell | 2009-03-29 00:51:55 -0500 (Sun, 29 Mar 2009) | 5 lines Ensure targs variable is fully initialized. (closes issue #14758) Reported by: tim_ringenbach ........ ................ * channels/Makefile, /: Merged revisions 184838 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184838 | russell | 2009-03-29 00:32:04 -0500 (Sun, 29 Mar 2009) | 8 lines Simplify chan_h323 build to not require a second run of "make". (closes issue #14715) Reported by: jthurman Patches: h323-makefile-1.6.2.0-beta1.patch uploaded by jthurman (license 614) Tested by: tzafrir, russell ........ 2009-03-27 19:17 +0000 [r184765] Kevin P. Fleming * channels/chan_iax2.c, main/timing.c, main/channel.c, /, include/asterisk/timing.h, include/asterisk/channel.h: Merged revisions 184762 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184762 | kpfleming | 2009-03-27 14:10:32 -0500 (Fri, 27 Mar 2009) | 12 lines Improve timing interface to remember which provider provided a timer The ability to load/unload timing interfaces is nice, but it means that when a timer is allocated, it may come from provider A, but later provider B becomes the 'preferred' provider. If this happens, all timer API calls on the timer that was provided by provider A will actually be handed to provider B, which will say WTF and return an error. This patch changes the timer API to include a pointer to the provider of the timer handle so that future operations on the timer will be forwarded to the proper provider. (closes issue #14697) Reported by: moy Review: http://reviewboard.digium.com/r/211/ ........ 2009-03-27 18:09 +0000 [r184728] Russell Bryant * main/manager.c, /, apps/app_minivm.c: Merged revisions 184726 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184726 | russell | 2009-03-27 13:04:43 -0500 (Fri, 27 Mar 2009) | 2 lines Use ast_random() instead of rand() to ensure we use the best RNG available. ........ 2009-03-27 15:54 +0000 [r184675] Joshua Colp * /, res/res_agi.c: Merged revisions 184673 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184673 | file | 2009-03-27 12:46:46 -0300 (Fri, 27 Mar 2009) | 7 lines Fix speech structure leak in the AGI speech recognition integration. The AGI dialplan applications did not destroy the speech structure automatically if it was not destroyed by the running AGI script. They will now do this. (issue LUMENVOX-15) ........ 2009-03-27 14:04 +0000 [r184631] Russell Bryant * main/asterisk.c, include/asterisk/utils.h, main/pbx.c, /, res/ais/evt.c, main/event.c, pbx/pbx_dundi.c: Merged revisions 184630 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184630 | russell | 2009-03-27 09:00:18 -0500 (Fri, 27 Mar 2009) | 2 lines Change g_eid to ast_eid_default. ........ 2009-03-27 13:22 +0000 [r184587] Joshua Colp * /, channels/chan_sip.c: Merged revisions 184566 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r184566 | file | 2009-03-27 10:15:26 -0300 (Fri, 27 Mar 2009) | 16 lines Merged revisions 184565 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184565 | file | 2009-03-27 10:06:45 -0300 (Fri, 27 Mar 2009) | 9 lines Fix an issue where nat=yes would not always take effect for the RTP session on outgoing calls. If calls were placed using an IP address or hostname the global nat setting was copied over but was not set on the RTP session itself. This caused the RTP stack to not perform symmetric RTP actions. (closes issue #14546) Reported by: acunningham ........ ................ 2009-03-27 02:25 +0000 [r184513-184547] Russell Bryant * /, include/asterisk/lock.h: Merged revisions 184531 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184531 | russell | 2009-03-26 21:20:23 -0500 (Thu, 26 Mar 2009) | 20 lines Fix some issues with rwlock corruption that caused deadlock like symptoms. When dvossel and I were doing some load testing last week, we noticed that we could make Asterisk trunk lock up instantly when we started generating a bunch of calls. The backtraces of locked threads were bizarre, and many were stuck on an _unlock_ of an rwlock. The changes are: 1) Fix a number of places where a backtrace would be loaded into an invalid index of the backtrace array. It's an off by one error, which ends up writing over the rwlock itself. 2) Ensure that in the array of held locks, we NULL out an index once it is not being used so that it's not confusing when analyzing its contents. 3) Remove a bunch of logging referring to an rwlock operating being done with "deep reentrancy". It is normal for _many_ threads to hold a read lock on an rwlock. ........ * /, main/file.c: Merged revisions 184515 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184515 | russell | 2009-03-26 20:40:28 -0500 (Thu, 26 Mar 2009) | 2 lines Don't act surprised if we get a -1 indication. ........ * include/asterisk/heap.h, /, main/heap.c: Merged revisions 184512 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184512 | russell | 2009-03-26 20:35:56 -0500 (Thu, 26 Mar 2009) | 2 lines Pass more useful information through to lock tracking when DEBUG_THREADS is on. ........ 2009-03-26 22:19 +0000 [r184451] Kevin P. Fleming * sounds/Makefile, /: Merged revisions 184448 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r184448 | kpfleming | 2009-03-26 17:18:14 -0500 (Thu, 26 Mar 2009) | 9 lines Merged revisions 184447 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184447 | kpfleming | 2009-03-26 17:17:32 -0500 (Thu, 26 Mar 2009) | 3 lines use new, improved 8kHz prompts ........ ................ 2009-03-26 21:18 +0000 [r184394] David Vossel * /, apps/app_test.c: Merged revisions 184389 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r184389 | dvossel | 2009-03-26 16:09:37 -0500 (Thu, 26 Mar 2009) | 14 lines Merged revisions 184388 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184388 | dvossel | 2009-03-26 16:07:32 -0500 (Thu, 26 Mar 2009) | 8 lines pri loop TestClient/TestServer fails: server SEND DTMF 8 app_test was failing when sending the last DTMF digit, 8, because of the 100ms pause issued after DTMF is sent. During this pause the other side would hang up causing the test to look like it failed. Now the other side waits a second before hanging up. (closes issue #12442) Reported by: tzafrir ........ ................ 2009-03-25 22:13 +0000 [r184325-184345] Russell Bryant * /, main/event.c: Merged revisions 184344 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184344 | russell | 2009-03-25 17:11:35 -0500 (Wed, 25 Mar 2009) | 2 lines Remove unneeded AST_LIST_ENTRY() and comment on the purpose of ast_event_ref. ........ * channels/chan_iax2.c, channels/chan_dahdi.c, include/asterisk/event.h, channels/chan_skinny.c, res/ais/evt.c, main/event.c, include/asterisk/strings.h, main/asterisk.c, channels/chan_mgcp.c, apps/app_voicemail.c, channels/chan_unistim.c, include/asterisk/devicestate.h, /, channels/chan_sip.c, main/devicestate.c, include/asterisk/_private.h: Merged revisions 184339 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184339 | russell | 2009-03-25 16:57:19 -0500 (Wed, 25 Mar 2009) | 35 lines Improve performance of the ast_event cache functionality. This code comes from svn/asterisk/team/russell/event_performance/. Here is a summary of the changes that have been made, in order of both invasiveness and performance impact, from smallest to largest. 1) Asterisk 1.6.1 introduces some additional logic to be able to handle distributed device state. This functionality comes at a cost. One relatively minor change in this patch is that the extra processing required for distributed device state is now completely bypassed if it's not needed. 2) One of the things that I noticed when profiling this code was that a _lot_ of time was spent doing string comparisons. I changed the way strings are represented in an event to include a hash value at the front. So, before doing a string comparison, we do an integer comparison on the hash. 3) Finally, the code that handles the event cache has been re-written. I tried to do this in a such a way that it had minimal impact on the API. I did have to change one API call, though - ast_event_queue_and_cache(). However, the way it works now is nicer, IMO. Each type of event that can be cached (MWI, device state) has its own hash table and rules for hashing and comparing objects. This by far made the biggest impact on performance. For additional details regarding this code and how it was tested, please see the review request. (closes issue #14738) Reported by: russell Review: http://reviewboard.digium.com/r/205/ ........ * /: add reviewboard:url property. 2009-03-25 19:26 +0000 [r184282] Joshua Colp * /, channels/chan_sip.c: Merged revisions 184280 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184280 | file | 2009-03-25 16:22:06 -0300 (Wed, 25 Mar 2009) | 5 lines Fix issue with a T38 reinvite being sent even if not configured to do so. If we receive a T38 request negotiate control frame we should only attempt to do so if the option is enabled on the dialog. ........ 2009-03-25 15:12 +0000 [r184223] Eliel C. Sardanons * main/asterisk.c, /: Merged revisions 184220 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r184220 | eliel | 2009-03-25 10:38:19 -0400 (Wed, 25 Mar 2009) | 19 lines Merged revisions 184188 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184188 | eliel | 2009-03-25 10:12:54 -0400 (Wed, 25 Mar 2009) | 13 lines Avoid destroying the CLI line when moving the cursor backward and trying to autocomplete. When moving the cursor backward and pressing TAB to autocomplete, a NULL is put in the line and we are loosing what we have already wrote after the actual cursor position. (closes issue #14373) Reported by: eliel Patches: asterisk.c.patch uploaded by eliel (license 64) Tested by: lmadsen ........ ................ 2009-03-25 01:55 +0000 [r184149] Russell Bryant * main/timing.c, utils/Makefile, /, include/asterisk/compat.h: Merged revisions 184147 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184147 | russell | 2009-03-24 20:42:10 -0500 (Tue, 24 Mar 2009) | 5 lines Fix build issues on Mac OSX. (closes issue #14714) Reported by: ygor ........ 2009-03-24 22:42 +0000 [r184081] Mark Michelson * apps/app_senddtmf.c, /: Merged revisions 184079 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r184079 | mmichelson | 2009-03-24 17:40:39 -0500 (Tue, 24 Mar 2009) | 15 lines Merged revisions 184078 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184078 | mmichelson | 2009-03-24 17:34:45 -0500 (Tue, 24 Mar 2009) | 9 lines Change NULL pointer check to be ast_strlen_zero. The 'digit' variable is guaranteed to be non-NULL, so the if statement could never evaluate true. Changing to ast_strlen_zero makes the logic correct. This was found while reviewing ast_channel_ao2 code review. ........ ................ 2009-03-24 21:47 +0000 [r184039] Russell Bryant * channels/chan_iax2.c, /: Merged revisions 184037 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184037 | russell | 2009-03-24 16:40:44 -0500 (Tue, 24 Mar 2009) | 6 lines Exclude slin16, siren7, and siren14 from bandwidth=low and =medium The default codec configuration for chan_iax2 is bandwidth=low. I noticed slin16 being negotiated as the codec in some test calls, but that no longer happens after this change. ........ 2009-03-24 15:28 +0000 [r183867-183916] Tilghman Lesher * /, configs/voicemail.conf.sample: Merged revisions 183914 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183914 | tilghman | 2009-03-24 10:26:42 -0500 (Tue, 24 Mar 2009) | 10 lines Merged revisions 183913 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183913 | tilghman | 2009-03-24 10:25:42 -0500 (Tue, 24 Mar 2009) | 3 lines Additionally note that the operator option needs an 'o' extension. (Related to issue #14731) ........ ................ * /, main/http.c: Merged revisions 183865 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r183865 | tilghman | 2009-03-23 18:28:20 -0500 (Mon, 23 Mar 2009) | 2 lines Allow browsers to cache images and other static content. (This is a regression over 1.4) ........ 2009-03-23 18:59 +0000 [r183768] Mark Michelson * res/res_monitor.c, /: Merged revisions 183766 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183766 | mmichelson | 2009-03-23 13:58:03 -0500 (Mon, 23 Mar 2009) | 13 lines Merged revisions 183700 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183700 | mmichelson | 2009-03-23 12:59:28 -0500 (Mon, 23 Mar 2009) | 7 lines Fix a memory leak in res_monitor.c The only way that this leak would occur is if Monitor were started using the Manager interface and no File: header were given. Discovered while reviewing the ast_channel_ao2 review request. ........ ................ 2009-03-23 18:12 +0000 [r183703] Leif Madsen * channels/chan_dahdi.c, /: Merged revisions 183701 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r183701 | lmadsen | 2009-03-23 14:06:40 -0400 (Mon, 23 Mar 2009) | 7 lines Fixes a documentation error introduced during the CLI cleanup at AstriDevCon 2008. (closes issue #14655) Reported by: ulogic Patches: chan_dahdi.patch uploaded by ulogic (license 728) Tested by: lmadsen ........ 2009-03-20 17:08 +0000 [r183563] Russell Bryant * channels/chan_iax2.c, /: Merged revisions 183560 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183560 | russell | 2009-03-20 12:00:58 -0500 (Fri, 20 Mar 2009) | 10 lines Merged revisions 183559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183559 | russell | 2009-03-20 11:53:25 -0500 (Fri, 20 Mar 2009) | 2 lines Fix a crash in IAX2 registration handling found during load testing with dvossel. ........ ................ 2009-03-19 20:33 +0000 [r183438] David Vossel * include/asterisk/features.h, apps/app_dial.c, /, main/features.c: Merged revisions 183436 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183436 | dvossel | 2009-03-19 15:30:39 -0500 (Thu, 19 Mar 2009) | 13 lines Merged revisions 183386 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183386 | dvossel | 2009-03-19 14:40:07 -0500 (Thu, 19 Mar 2009) | 6 lines Cleaning up a few things in detect disconnect patch Initialized ast_call_feature in detect_disconnect to avoid accessing uninitialized memory. Cleaned up /param tags in features.h. No longer send dynamic features in ast_feature_detect. issue #11583 ........ ................ 2009-03-19 19:19 +0000 [r183333] Tilghman Lesher * channels/chan_dahdi.c, /: Merged revisions 183321 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183321 | tilghman | 2009-03-19 14:17:31 -0500 (Thu, 19 Mar 2009) | 15 lines Merged revisions 183319 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183319 | tilghman | 2009-03-19 14:15:33 -0500 (Thu, 19 Mar 2009) | 8 lines Delay signalling progress until a PRI channel really signals progress. (closes issue #13034) Reported by: klaus3000 Patches: 20090316__bug13034.diff.txt uploaded by tilghman (license 14) patch_trunk_183progress_klaus3000.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ ................ 2009-03-19 18:14 +0000 [r183249] Russell Bryant * main/loader.c, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 183242 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183242 | russell | 2009-03-19 13:00:15 -0500 (Thu, 19 Mar 2009) | 10 lines Merged revisions 183241 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183241 | russell | 2009-03-19 12:52:52 -0500 (Thu, 19 Mar 2009) | 2 lines Remove the use of RTLD_NOLOAD, as it is not behaving like expected. ........ ................ 2009-03-19 18:11 +0000 [r183246] Mark Michelson * apps/app_queue.c, /: Merged revisions 183244 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r183244 | mmichelson | 2009-03-19 13:10:34 -0500 (Thu, 19 Mar 2009) | 16 lines Fix a memory leak associated with queues. For every attempt that app_queue made to place an outbound call to a queue member, we would allocate a queue_end_bridge structure. When the bridge for the call had completed, we would free the structure. Unfortunately not all call attempts actually end up bridged to a member, so we need to be more selective of when to allocate the structure. With this change, the allocation occurs in an area where we can guarantee that the call will be bridged. (closes issue #14680) Reported by: caspy Patches: 14680.patch uploaded by mmichelson (license 60) Tested by: caspy ........ 2009-03-19 17:08 +0000 [r183198] David Vossel * include/asterisk/features.h, apps/app_dial.c, /, main/features.c: Merged revisions 183172 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183172 | dvossel | 2009-03-19 11:28:33 -0500 (Thu, 19 Mar 2009) | 20 lines Merged revisions 183126 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009) | 17 lines Allow disconnect feature before a call is bridged feature.conf has a disconnect option. By default this option is set to '*', but it could be anything. If a user wishes to disconnect a call before the other side answers, only '*' will work, regardless if the disconnect option is set to something else. This is because features are unavailable until bridging takes place. The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different. This patch allows features to be detected from outside of the bridge, but not operated on. In this case, the disconnect feature can be detected before briding and handled outside of features.c. (closes issue #11583) Reported by: sobomax Patches: patch-apps__app_dial.c uploaded by sobomax (license 359) 11583.latest-patch uploaded by murf (license 17) detect_disconnect.diff uploaded by dvossel (license 671) Tested by: sobomax, dvossel Review: http://reviewboard.digium.com/r/195/ ........ ................ 2009-03-19 16:09 +0000 [r183121] Mark Michelson * /, channels/chan_sip.c: Merged revisions 183117 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183117 | mmichelson | 2009-03-19 11:07:54 -0500 (Thu, 19 Mar 2009) | 20 lines Merged revisions 183115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183115 | mmichelson | 2009-03-19 11:04:02 -0500 (Thu, 19 Mar 2009) | 14 lines Fix an issue where cancelled outgoing SIP calls would erroneously report the device as "in use." A user was having an issue where if an outgoing SIP call was canceled, the SIP device would remain in use if we had not received any response to the initial INVITE we sent out. The SIP device would remain in use until the autocongestion timer was exhausted. I tracked down the cause of this to be the section of code I am removing here. I asked several people what the purpose of this code was meant to be, but no one could give me any sort of answer as to why this was here. The person who was having this issue has been using this patch for several months and it has stopped the problems they have had. AST-196 ........ ................ 2009-03-19 Leif Madsen * Release Asterisk 1.6.1.0-rc3 2009-03-19 15:43 +0000 [r183067-183110] Joshua Colp * /, channels/chan_sip.c: Merged revisions 183108 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r183108 | file | 2009-03-19 12:37:23 -0300 (Thu, 19 Mar 2009) | 11 lines Improve our triggering of a T38 switchover internally when triggered by a received reinvite. Previously we reached across the channel bridge to get the other party's SIP dialog structure in order to trigger an outgoing reinvite. This is extremely dangerous to do and only works if bridged to another SIP channel. This patch changes this to use the T38 control frame method of requesting a switchover. This change also causes the SIP channel driver to propogate back whether the switchover worked or not instead of blindly accepting the incoming T38 reinvite. Review: http://reviewboard.digium.com/r/200/ ........ * main/channel.c, /: Merged revisions 183057 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r183057 | file | 2009-03-18 19:22:56 -0300 (Wed, 18 Mar 2009) | 6 lines Fix an issue where a T38 control frame would get dropped. If two channels were bridged together using a generic bridge the T38 control frame would get passed up instead of being indicated on the other channel. ........ 2009-03-18 21:19 +0000 [r183030] Jeff Peeler * /, channels/h323/ast_h323.cxx: Merged revisions 183028 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r183028 | jpeeler | 2009-03-18 16:18:27 -0500 (Wed, 18 Mar 2009) | 4 lines Add some code removed by mistake from commit 182722 that works around a file descriptor leak in versions of PWLib prior to 1.12.0. ........ 2009-03-18 14:32 +0000 [r182946] Russell Bryant * main/poll.c, main/io.c, main/channel.c, channels/chan_skinny.c, configure, apps/app_mp3.c, res/res_agi.c, include/asterisk/poll-compat.h, channels/chan_alsa.c, main/asterisk.c, apps/app_nbscat.c, /, main/Makefile, include/asterisk/autoconfig.h.in, configure.ac, include/asterisk/io.h, main/utils.c, include/asterisk/channel.h: Merged revisions 182847 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r182847 | russell | 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009) | 52 lines Merged revisions 182810 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines Fix cases where the internal poll() was not being used when it needed to be. We have seen a number of problems caused by poll() not working properly on Mac OSX. If you search around, you'll find a number of references to using select() instead of poll() to work around these issues. In Asterisk, we've had poll.c which implements poll() using select() internally. However, we were still getting reports of problems. vadim investigated a bit and realized that at least on his system, even though we were compiling in poll.o, the system poll() was still being used. So, the primary purpose of this patch is to ensure that we're using the internal poll() when we want it to be used. The changes are: 1) Remove logic for when internal poll should be used from the Makefile. Instead, put it in the configure script. The logic in the configure script is the same as it was in the Makefile. Ideally, we would have a functionality test for the problem, but that's not actually possible, since we would have to be able to run an application on the _target_ system to test poll() behavior. 2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll() throughout the source tree to ast_poll(). I feel that it is good practice to give the API call a new name when we are changing its behavior and not using the system version directly in all cases. So, normally, ast_poll() is just redefined to poll(). On systems where AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll(). 4) Change poll() in main/poll.c to be ast_internal_poll(). It's worth noting that any code that still uses poll() directly will work fine (if they worked fine before). So, for example, out of tree modules that are using poll() will not stop working or anything. However, for modules to work properly on Mac OSX, ast_poll() needs to be used. (closes issue #13404) Reported by: agalbraith Tested by: russell, vadim http://reviewboard.digium.com/r/198/ ........ ................ 2009-03-17 20:52 +0000 [r182724] Jeff Peeler * channels/h323/chan_h323.h, channels/h323/compat_h323.cxx, /, channels/h323/ast_h323.cxx, configure, autoconf/ast_check_openh323.m4, channels/h323/compat_h323.h, channels/chan_h323.c, channels/h323/ast_h323.h: Merged revisions 182722 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182722 | jpeeler | 2009-03-17 15:47:31 -0500 (Tue, 17 Mar 2009) | 15 lines Allow H.323 Plus library to be used in addition to the OpenH323 library Chan_h323 can now be compiled against both the previously supported versions of OpenH323 as well as the current H.323 Plus (version 1.20.2). The configure script has been modified to look in the default install location of h323 to hopefully help avoid using the environment variables OPENH323DIR and PWLIBDIR. Also, the CLI command "h323 show version" has been added which indicates which version of h323 is in use. (closes issue #11261) Reported by: vhatz Patches: asterisk-1.6.0.6-h323plus.patch uploaded by jthurman (license 614) ........ 2009-03-17 15:31 +0000 [r182570] Russell Bryant * main/channel.c, /: Merged revisions 182553 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182553 | russell | 2009-03-17 10:22:12 -0500 (Tue, 17 Mar 2009) | 5 lines Tweak the handling of the frame list inside of ast_answer(). This does not change any behavior, but moves the frames from the local frame list back to the channel read queue using an O(n) algorithm instead of O(n^2). ........ 2009-03-17 15:00 +0000 [r182527-182533] Kevin P. Fleming * main/channel.c, /: Merged revisions 182530 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182530 | kpfleming | 2009-03-17 09:59:33 -0500 (Tue, 17 Mar 2009) | 2 lines correct logic flaw in ast_answer() changes in r182525 ........ * main/channel.c, /, main/features.c, include/asterisk/channel.h: Merged revisions 182525 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182525 | kpfleming | 2009-03-17 09:38:11 -0500 (Tue, 17 Mar 2009) | 11 lines Improve behavior of ast_answer() to not lose incoming frames ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations. When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames. This patch rectifies this situation, by storing all incoming frames during the delay period on a list, and then requeuing them onto the channel before returning to the caller. http://reviewboard.digium.com/r/196/ ........ 2009-03-17 05:54 +0000 [r182452] Tilghman Lesher * main/db.c, /: Merged revisions 182450 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r182450 | tilghman | 2009-03-17 00:51:54 -0500 (Tue, 17 Mar 2009) | 14 lines Merged revisions 182449 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182449 | tilghman | 2009-03-17 00:50:52 -0500 (Tue, 17 Mar 2009) | 7 lines Fix race in astdb The underlying db1 implementation does not fully isolate the pages retrieved from astdb, so the lock protecting accesses needs to be extended until the copy from the shared memory structure is done. (closes issue #14682) Reported by: makoto ........ ................ 2009-03-16 17:53 +0000 [r182284] David Vossel * channels/chan_iax2.c, /: Merged revisions 182282 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r182282 | dvossel | 2009-03-16 12:49:58 -0500 (Mon, 16 Mar 2009) | 13 lines Merged revisions 182281 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182281 | dvossel | 2009-03-16 12:47:42 -0500 (Mon, 16 Mar 2009) | 7 lines Randomize IAX2 encryption padding The 16-32 byte random padding at the beginning of an encrypted IAX2 frame turns out to not be all that random at all. This patch calls ast_random to fill the padding buffer with random data. The padding is randomized at the beginning of every encrypted call and for every encrypted retransmit frame. Review: http://reviewboard.digium.com/r/193/ ........ ................ 2009-03-16 17:38 +0000 [r182280] Tilghman Lesher * channels/chan_local.c, /, funcs/func_env.c: Merged revisions 182211,182278 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r182211 | tilghman | 2009-03-16 10:50:55 -0500 (Mon, 16 Mar 2009) | 14 lines Merged revisions 182208 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182208 | tilghman | 2009-03-16 10:39:15 -0500 (Mon, 16 Mar 2009) | 7 lines Fixup glare detection, to fix a memory leak of a local pvt structure. (closes issue #14656) Reported by: caspy Patches: 20090313__bug14656__2.diff.txt uploaded by tilghman (license 14) Tested by: caspy ........ ................ r182278 | tilghman | 2009-03-16 12:33:38 -0500 (Mon, 16 Mar 2009) | 7 lines Fix an off-by-one error in the FILE() function, and extend FILE()'s length parameter to work like variable substitution. Previously, FILE() returned one less character than specified, due to the terminating NULL. Both the offset and length parameters now behave identically to the way variable substitution offsets and lengths also work. (closes issue #14670) Reported by: BMC ................ 2009-03-16 14:00 +0000 [r182173] Joshua Colp * main/channel.c, /: Merged revisions 182171 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182171 | file | 2009-03-16 10:58:24 -0300 (Mon, 16 Mar 2009) | 7 lines Fix a memory leak in the ast_answer / __ast_answer API call. For a channel that is not yet answered this API call will wait until a voice frame is received on the channel before returning. It does this by waiting for frames on the channel and reading them in. The frames read in were not freed when they should have been. ........ 2009-03-13 21:27 +0000 [r182068-182123] Mark Michelson * apps/app_queue.c, /: Merged revisions 182121 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182121 | mmichelson | 2009-03-13 16:26:20 -0500 (Fri, 13 Mar 2009) | 6 lines Change faulty comparison used when announcing average hold minutes and seconds (closes issue #14227) Reported by: caspy ........ * /, main/features.c: Merged revisions 182029 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r182029 | mmichelson | 2009-03-13 12:26:43 -0500 (Fri, 13 Mar 2009) | 41 lines Merged revisions 181990 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181990 | mmichelson | 2009-03-13 12:12:32 -0500 (Fri, 13 Mar 2009) | 35 lines Check the DYNAMIC_FEATURES of both the chan and peer when interpreting DTMF. Dynamic features defined in the applicationmap section of features.conf allow one to specify whether the caller, callee, or both have the ability to use the feature. The documentation in the features.conf.sample file could be interpreted to mean that one only needs to set the DYNAMIC_FEATURES channel variable on the calling channel in order to allow for the callee to be able to use the features which he should have permission to use. However, the DYNAMIC_FEATURES variable would only be read from the channel of the participant that pressed the DTMF sequence to activate the feature. The result of this was that the callee was unable to use dynamic features unless the dialplan writer had taken measures to be sure that the DYNAMIC_FEATURES variable was set on the callee's channel. This commit changes the behavior of ast_feature_interpret to concatenate the values of DYNAMIC_FEATURES from both parties involved in the bridge. The features themselves determine who has permission to use them, so there is no reason to believe that one side of the bridge could gain the ability to perform an action that they should not have the ability to perform. Kevin Fleming pointed out on the asterisk-users list that the typical way that this was worked around in the past was by setting _DYNAMIC_FEATURES on the calling channel so that the value would be inherited by the called channel. While this works, the documentation alone is not enough to figure out why this is necessary for the callee to be able to use dynamic features. In this particular case, changing the code to match the documentation is safe, easy, and will generally make things easier for people for future installations. This bug was originally reported on the asterisk-users list by David Ruggles. (closes issue #14657) Reported by: mmichelson Patches: 14657.patch uploaded by mmichelson (license 60) ........ ................ 2009-03-13 17:29 +0000 [r182042] Joshua Colp * /, channels/chan_sip.c: Merged revisions 182022 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182022 | file | 2009-03-13 14:25:09 -0300 (Fri, 13 Mar 2009) | 7 lines Fix an issue with requesting a T38 reinvite before the call is answered. The code responsible for sending the T38 reinvite did not check if an INVITE was already being handled. This caused things to get confused and the call to fail. The code now defers sending the T38 reinvite until the current INVITE is done being handled. (issue AST-191) ........ 2009-03-13 16:58 +0000 [r181987] Kevin P. Fleming * /, channels/chan_sip.c: Merged revisions 181985 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r181985 | kpfleming | 2009-03-13 11:55:38 -0500 (Fri, 13 Mar 2009) | 1 line improve a bit of suboptimal code ........ 2009-03-12 21:45 +0000 [r181771-181849] Mark Michelson * apps/app_queue.c, /: Merged revisions 181846 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r181846 | mmichelson | 2009-03-12 16:43:51 -0500 (Thu, 12 Mar 2009) | 3 lines Run the macro on the queue member's channel when he answers, not the caller's channel. ........ * /, channels/chan_sip.c: Merged revisions 181769 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181769 | mmichelson | 2009-03-12 13:30:58 -0500 (Thu, 12 Mar 2009) | 28 lines Merged revisions 181768 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181768 | mmichelson | 2009-03-12 13:29:48 -0500 (Thu, 12 Mar 2009) | 22 lines Properly send a 487 on an INVITE we have not responded to if we receive a BYE. If we receive an INVITE from an endpoint and then later receive a BYE from that same endpoint before we have sent a final response for the INVITE, then we need to respond to the INVITE with a 487. There was logic in the code prior to this commit which seemed to exist solely to handle this situation, but there was one condition in an if statement which was incorrect. The only way we would send a 487 was if the sip_pvt had no owner channel. This made no sense since we created the owner channel when we received the INVITE, meaning that the majority of the time we would never send the 487. The 487 being sent should not rely on whether we have created a channel. Its delivery should be dependent on the current state of the initial INVITE transaction. With this commit, that logic is now correctly in place. (closes issue #14149) Reported by: legranjl Patches: 14149.patch uploaded by mmichelson (license 60) Tested by: legranjl ........ ................ 2009-03-12 18:07 +0000 [r181733] Tilghman Lesher * /, main/translate.c: Merged revisions 181731 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r181731 | tilghman | 2009-03-12 12:32:13 -0500 (Thu, 12 Mar 2009) | 9 lines Adjust translation table column widths based upon the translation times. Previously, only 5 columns were displayed, and if a translation time exceeded 99,999 useconds, it would be displayed as 0, instead of its actual time. (closes issue #14532) Reported by: pj Patches: 20090311__bug14532.diff.txt uploaded by tilghman (license 14) Tested by: pj ........ 2009-03-12 16:58 +0000 [r181614-181667] Joshua Colp * /, res/res_musiconhold.c: Merged revisions 181665 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181665 | file | 2009-03-12 13:56:58 -0300 (Thu, 12 Mar 2009) | 9 lines Merged revisions 181664 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181664 | file | 2009-03-12 13:56:20 -0300 (Thu, 12 Mar 2009) | 2 lines Fix incorrect usage of strncasecmp... I really meant to use strcasecmp. ........ ................ * /, res/res_musiconhold.c: Merged revisions 181661 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181661 | file | 2009-03-12 13:53:52 -0300 (Thu, 12 Mar 2009) | 19 lines Merged revisions 181659-181660 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181659 | file | 2009-03-12 13:50:37 -0300 (Thu, 12 Mar 2009) | 8 lines Fix another scenario where depending on configuration the stream would not get read. For custom commands we don't know whether the audio is coming from a stream or not so we are going to have to read the data despite no channels. (closes issue #14416) Reported by: caspy ........ r181660 | file | 2009-03-12 13:52:45 -0300 (Thu, 12 Mar 2009) | 2 lines Fix logic flaw in previous commit. ........ ................ * /, res/res_musiconhold.c: Merged revisions 181656 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181656 | file | 2009-03-12 13:32:20 -0300 (Thu, 12 Mar 2009) | 17 lines Merged revisions 181655 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181655 | file | 2009-03-12 13:29:19 -0300 (Thu, 12 Mar 2009) | 10 lines Fix issue with streaming MOH failing if nobody is listening. When a music class is setup to actually provide music on hold from a stream we need to constantly read audio from it since it will constantly be providing audio. This is now done despite there being no channels listening to it. (closes issue #14416) Reported by: caspy ........ ................ * apps/app_dial.c, /: Merged revisions 181612 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r181612 | file | 2009-03-12 10:24:12 -0300 (Thu, 12 Mar 2009) | 5 lines Fix crash when sleep and retries argument was not given to RetryDial application. (closes issue #14647) Reported by: sherpya ........ 2009-03-12 01:05 +0000 [r181544] Richard Mudgett * /, build_tools/make_version: Merged revisions 181542 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r181542 | rmudgett | 2009-03-11 20:00:29 -0500 (Wed, 11 Mar 2009) | 1 line Use the correct branch integrated property when generating the version string ........ 2009-03-11 23:21 +0000 [r181521] Michiel van Baak * /, configs/sip.conf.sample: Merged revisions 181499 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk Provide correct hint to debug SIP trouble in the default config (closes issue #14646) Reported by: strk 2009-03-11 22:27 +0000 [r181474] Russell Bryant * main/channel.c, /: Merged revisions 181465 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r181465 | russell | 2009-03-11 17:25:57 -0500 (Wed, 11 Mar 2009) | 2 lines Make handling of the BRIDGE_PLAY_SOUND variable thread-safe. ........ 2009-03-11 22:23 +0000 [r181457] Jason Parker * /, configure, configure.ac: Merged revisions 181444 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181444 | qwell | 2009-03-11 17:20:13 -0500 (Wed, 11 Mar 2009) | 11 lines Merged revisions 181436 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181436 | qwell | 2009-03-11 17:18:42 -0500 (Wed, 11 Mar 2009) | 4 lines Allow prefix to set localstatedir (when used and different from the default). This is similar to the /etc change that was made for the non-FreeBSD case. ........ ................ 2009-03-11 22:16 +0000 [r181426-181430] Russell Bryant * main/channel.c, /: Merged revisions 181428 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r181428 | russell | 2009-03-11 17:14:55 -0500 (Wed, 11 Mar 2009) | 2 lines Make handling of the BRIDGEPVTCALLID variable thread-safe. ........ * main/channel.c, /: Merged revisions 181424 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181424 | russell | 2009-03-11 16:49:29 -0500 (Wed, 11 Mar 2009) | 17 lines Merged revisions 181423 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181423 | russell | 2009-03-11 16:42:58 -0500 (Wed, 11 Mar 2009) | 9 lines Make code that updates BRIDGEPEER variable thread-safe. It is not safe to read the name field of an ast_channel without the channel locked. This patch fixes some places in channel.c where this was being done, and lead to crashes related to masquerades. (closes issue #14623) Reported by: guillecabeza ........ ................ 2009-03-11 17:40 +0000 [r181373] David Vossel * channels/chan_iax2.c, channels/iax2-parser.h, /: Merged revisions 181371 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181371 | dvossel | 2009-03-11 12:34:57 -0500 (Wed, 11 Mar 2009) | 17 lines Merged revisions 181340 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181340 | dvossel | 2009-03-11 12:25:31 -0500 (Wed, 11 Mar 2009) | 11 lines encrypted IAX2 during packet loss causes decryption to fail on retransmitted frames If an iax channel is encrypted, and a retransmit frame is sent, that packet's iseqno is updated while it is encrypted. This causes the entire frame to be corrupted. When the corrupted frame is sent, the other side decrypts it and sends a VNAK back because the decrypted frame doesn't make any sense. When we get the VNAK, we look through the sent queue and send the same corrupted frame causing a loop. To fix this, encrypted frames requiring retransmission are decrypted, updated, then re-encrypted. Since key-rotation may change the key held by the pvt struct, the keys used for encryption/decryption are held within the iax_frame to guarantee they remain correct. (closes issue #14607) Reported by: stevenla Tested by: dvossel Review: http://reviewboard.digium.com/r/192/ ........ ................ 2009-03-11 17:29 +0000 [r181298-181359] Joshua Colp * /, channels/chan_sip.c: Merged revisions 181345 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181345 | file | 2009-03-11 14:26:40 -0300 (Wed, 11 Mar 2009) | 21 lines Merged revisions 181328 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181328 | file | 2009-03-11 14:22:52 -0300 (Wed, 11 Mar 2009) | 14 lines Fix issue where an attended transfer could not be completed under a rare scenario. When completing an attended transfer chan_sip does a check to make sure the extension in the URI portion of the Refer-To header is a local valid extension. We don't actually need to check this since we know for sure the other channel is already up and talking to the extension. Some devices do not put the extension in the Refer-To header either, which can cause the extension check to fail. We now no longer do this check if it is an attended transfer. (closes issue #14628) Reported by: sverre Patches: 14628.diff uploaded by file (license 11) ........ ................ * /, channels/chan_sip.c: Merged revisions 181296 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181296 | file | 2009-03-11 13:40:48 -0300 (Wed, 11 Mar 2009) | 16 lines Merged revisions 181295 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181295 | file | 2009-03-11 13:36:50 -0300 (Wed, 11 Mar 2009) | 9 lines Fix a problem with inband DTMF detection on outgoing SIP calls when dtmfmode=auto. When dtmfmode was set to auto the inband DTMF detector was not setup on outgoing SIP calls. This caused inband DTMF detection to fail. The inband DTMF detector is now setup for both dtmfmode inband and auto. (closes issue #13713) Reported by: makoto ........ ................ 2009-03-11 15:54 +0000 [r181199-181283] Jeff Peeler * channels/h323/ast_h323.cxx: add missing header file * pbx/pbx_config.c, utils/Makefile, include/asterisk/utils.h, include/asterisk/astmm.h, /, channels/chan_sip.c, channels/h323/ast_h323.cxx, main/features.c, utils/extconf.c: Merged revisions 181135 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r181135 | jpeeler | 2009-03-10 23:06:44 -0500 (Tue, 10 Mar 2009) | 20 lines Fix malloc debug macros to work properly with h323. The main problem here was that cstdlib was undefining free thereby causing the proper debug macros to not be used. ast_h323.cxx has been changed to call ast_free instead to avoid the issue. A few other issues were addressed: - There were a few instances of functions improperly passing ast_free instead of ast_free_ptr. - Some clean up was done to avoid the debug macros intentionally being redefined. (copied below from Kevin's commit, appreciate the help) - disable astmm.h from doing anything when STANDALONE is defined, which is used by the tools in the utils/ directory that use parts of Asterisk header files in hackish ways; also ensure that utils/extconf.c and utils/conf2ael.c are compiled with STANDALONE defined. (closes issue #13593) Reported by: pj ........ 2009-03-11 01:04 +0000 [r181035] Mark Michelson * /, channels/chan_sip.c: Merged revisions 181032-181033 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r181032 | mmichelson | 2009-03-10 19:46:47 -0500 (Tue, 10 Mar 2009) | 19 lines Merged revisions 181029,181031 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181029 | mmichelson | 2009-03-10 19:30:26 -0500 (Tue, 10 Mar 2009) | 9 lines Fix incorrect tag checking on transfers when pedantic=yes is enabled. (closes issue #14611) Reported by: klaus3000 Patches: patch_chan_sip_attended_transfer_1.4.23.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ r181031 | mmichelson | 2009-03-10 19:32:40 -0500 (Tue, 10 Mar 2009) | 3 lines Remove unused variables. ........ ................ r181033 | mmichelson | 2009-03-10 19:49:00 -0500 (Tue, 10 Mar 2009) | 3 lines Add missing comment that quotes RFC 3891 ................ 2009-03-10 22:07 +0000 [r180947] Jason Parker * /, configure, configure.ac, autoconf/ast_prog_sed.m4, autoconf/ast_check_gnu_make.m4: Merged revisions 180944 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180944 | qwell | 2009-03-10 17:03:41 -0500 (Tue, 10 Mar 2009) | 9 lines Merged revisions 180941 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180941 | qwell | 2009-03-10 17:02:18 -0500 (Tue, 10 Mar 2009) | 1 line Make things happier when using autoconf 2.62+ ........ ................ 2009-03-10 14:42 +0000 [r180802] Joshua Colp * main/manager.c, /: Merged revisions 180800 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r180800 | file | 2009-03-10 11:40:38 -0300 (Tue, 10 Mar 2009) | 5 lines Reset the thread local string buffer when handling the UserEvent action. (closes issue #14593) Reported by: JimDickenson ........ 2009-03-09 21:22 +0000 [r180740] Jeff Peeler * include/asterisk/heap.h, include/asterisk/http.h, include/asterisk/logger.h, main/tcptls.c, include/asterisk/res_odbc.h, include/asterisk/doxyref.h, include/asterisk/event.h, include/asterisk/audiohook.h, include/asterisk/dsp.h, include/asterisk/lock.h, include/asterisk/udptl.h, include/asterisk/dnsmgr.h, include/asterisk/utils.h, include/asterisk/devicestate.h, /, include/asterisk/taskprocessor.h, include/asterisk/astobj2.h, include/asterisk/channel.h, include/asterisk/tcptls.h, include/asterisk/manager.h, main/enum.c, include/asterisk/callerid.h, include/asterisk/app.h, include/asterisk/linkedlists.h, include/asterisk/sched.h, include/asterisk/datastore.h, include/asterisk/timing.h, include/asterisk/dlinkedlists.h, include/asterisk/pbx.h, include/asterisk/enum.h, include/asterisk/config.h, include/asterisk/rtp.h, include/asterisk/extconf.h, main/devicestate.c: Merged revisions 180719 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r180719 | jpeeler | 2009-03-09 15:58:17 -0500 (Mon, 09 Mar 2009) | 16 lines Add Doxygen documentation for API changes from 1.6.0 to 1.6.1 Copied from my review board description: This is a continuation of the API changes documentation started for describing changes between releases. Most of the API changes were pretty simple needing only to be brought to attention via the new "Asterisk API Changes" list. However, if you see anything that needs further explanation feel free to supplement what is there. The current method of documenting is to add (in the header file): \version and then to add the function to the change list in doxyref.h on the AstAPIChanges page. I also made sure all the functions that were newly added were tagged with \since 1.6.1. I think this is a good habit to start both for the historical aspect as well as for the future ability to easily add a "New Asterisk API" page. Review: http://reviewboard.digium.com/r/190/ ........ 2009-03-06 18:26 +0000 [r180585] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 180579 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180579 | mmichelson | 2009-03-06 12:25:44 -0600 (Fri, 06 Mar 2009) | 9 lines Merged revisions 180567 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180567 | mmichelson | 2009-03-06 12:23:09 -0600 (Fri, 06 Mar 2009) | 2 lines Make compilation succeed in dev-mode when IMAP storage is enabled. ........ ................ 2009-03-06 17:35 +0000 [r180537] David Vossel * main/enum.c, /: Merged revisions 180534 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180534 | dvossel | 2009-03-06 11:26:38 -0600 (Fri, 06 Mar 2009) | 15 lines Merged revisions 180532 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180532 | dvossel | 2009-03-06 11:19:55 -0600 (Fri, 06 Mar 2009) | 9 lines Fix handling of backreferences for ENUM lookups enum.c did not handle regex backtraces correctly. The '\1' in the regex is a backreference that requires a pattern match to be inserted. The way the code used to work is that it would find the backreference and insert the entire input string minus the '+'. This is incorrect. The regexec() function takes in a variable called pmatch which is an array of structs containing the start and end indexes for each backreference substring. The original code actually passed the pmatch array pointer into regexec but never did anything with it. Now when a backtrace is found, the backtrace number is looked up in the pmatch array and the correct substring is inserted. (closes issue #14576) Reported by: chris-mac Review: http://reviewboard.digium.com/r/187/ ........ ................ 2009-03-05 23:28 +0000 [r180425-180467] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 180465 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180465 | mmichelson | 2009-03-05 17:26:58 -0600 (Thu, 05 Mar 2009) | 22 lines Merged revisions 180464 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180464 | mmichelson | 2009-03-05 17:26:11 -0600 (Thu, 05 Mar 2009) | 16 lines [IMAP] Fix message retrieval issues when identical mailbox names were defined in separate contexts. There was a fix put in a while back so that an X-Asterisk-VM-Context message header was added to stored IMAP voicemails. This would allow for us to differentiate if the same mailbox name was used in multiple contexts. The problem still left was that not all places where messages were retrieved actually attempted to use this header for information when retrieving messages. This commit fixes that so that MWI and message retrieval from VoiceMailMain work as expected. (closes issue #13853) Reported by: vicks1 Patches: 13853_v2.patch uploaded by mmichelson (license 60) Tested by: lmadsen ........ ................ * apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged revisions 180383 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180383 | mmichelson | 2009-03-05 13:14:14 -0600 (Thu, 05 Mar 2009) | 31 lines Merged revisions 180380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180380 | mmichelson | 2009-03-05 12:58:48 -0600 (Thu, 05 Mar 2009) | 25 lines Fix broken mailbox parsing when searchcontexts option is enabled. When using the searchcontexts option in voicemail.conf, the code made the assumption that all mailbox names defined were unique across all contexts. However, the code did nothing to actually enforce this assumption, nor did it do anything to alert a user that he may have created an ambiguity in his voicemail.conf file by defining the same mailbox name in multiple contexts. With this change, we now will issue a nice long warning if searchcontexts is on and we encounter the same mailbox name in multiple contexts and ignore any duplicates after the first box. Whether searchcontexts is enabled or not, if we come across a duplicate mailbox in the same context, then we will issue a warning and ignore the duplicated mailbox. I have also added a small note to voicemail.conf.sample in the explanation for searchcontexts explaining that you cannot define the same mailbox in multiple contexts if you have enabled the option. (closes issue #14599) Reported by: lmadsen Patches: 14599.patch uploaded by mmichelson (license 60) (with slight modification) Tested by: lmadsen ........ ................ 2009-03-05 18:40 +0000 [r180378] Kevin P. Fleming * include/asterisk/frame.h, main/rtp.c, main/frame.c, /: Merged revisions 180373 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180373 | kpfleming | 2009-03-05 12:29:38 -0600 (Thu, 05 Mar 2009) | 15 lines Merged revisions 180372 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar 2009) | 9 lines Fix problems when RTP packet frame size is changed During some code analysis, I found that calling ast_rtp_codec_setpref() on an ast_rtp session does not work as expected; it does not adjust the smoother that may on the RTP session, in fact it summarily drops it, even if it has data in it, even if the current format's framing size has not changed. This is not good. This patch changes this behavior, so that if the packetization size for the current format changes, any existing smoother is safely updated to use the new size, and if no smoother was present, one is created. A new API call for smoothers, ast_smoother_reconfigure(), was required to implement these changes. Review: http://reviewboard.digium.com/r/184/ ........ ................ 2009-03-04 Leif Madsen * Released Asterisk 1.6.1.0-rc2 2009-03-04 21:09 +0000 [r180263] Russell Bryant * /, channels/chan_sip.c: Merged revisions 180261 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r180261 | russell | 2009-03-04 15:01:05 -0600 (Wed, 04 Mar 2009) | 54 lines Resolve object matching issues related to the removal of the sip_user object. Previously, chan_sip had both sip_peer and sip_user objects in memory. A patch went in to remove sip_user to simplify the code, since everything could be done with just sip_peer. This patch resolves some regressions found that were introduced by those changes. This code comes from svn/asterisk/team/group/sip-object-matching/. Here is a list of the changes that have been made: 1) When doing a match by name with the find_peer() function, make it much easier to specify which objects should be matched by having a parameter that specifies exactly which object types should be considered. Also, update find_by_name() to handle this parameter. Finally, update all code to use the new option values. 2) When looking up an object for an outbound request by name, consider peers only. (create_addr()) 3) Only match peers on an incoming registration request. 4) When doing authentication (except for SUBSCRIBE), look up users by name, instead of all objects by name. 5) When doing authentication (except for SUBSCRIBE), after looking for a user by name, look for a peer by IP address, instead of all objects by IP address. 6) When handling the SIP qualify CLI command or manager action, look for a peer by name, instead of any object by name. 7) When handling the SIP unregister CLI command, look for a peer by name, instead of any object by name. 9) In sip_do_debug_peer(), search for a peer by name, instead of any object by name. 9) When handling the SIPPEER() dialplan function, search for a peer by name, instead of any object by name. 10) In the following session timer related functions, st_get_se(), st_get_refresher(), and st_get_mode(), when looking for an object for a given sip_pvt using pvt->peername, look for a peer by name, instead of any object by name. 11) Fix build_peer() to properly handle the case where separate type=peer and type=user entries were specified in sip.conf. (closes issue #14505) Reported by: lmadsen Review: http://reviewboard.digium.com/r/172/ ........ 2009-03-04 19:27 +0000 [r180122-180197] Joshua Colp * /, main/callerid.c: Merged revisions 180195 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180195 | file | 2009-03-04 15:24:59 -0400 (Wed, 04 Mar 2009) | 11 lines Merged revisions 180194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180194 | file | 2009-03-04 15:22:50 -0400 (Wed, 04 Mar 2009) | 4 lines Look for the number in a callerid string starting from the end. This way a value using <> can exist in the name portion. (issue #AST-194) ........ ................ * apps/app_dial.c, /: Merged revisions 180120 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r180120 | file | 2009-03-04 10:39:28 -0400 (Wed, 04 Mar 2009) | 7 lines Remove duplicate 'k' and 'K' Dial options. (closes issue #14601) Reported by: alecdavis Patches: app_dial.optionk.diff.txt uploaded by alecdavis (license 585) ........ 2009-03-03 23:39 +0000 [r180080] David Vossel * main/channel.c, include/asterisk/app.h, apps/app_read.c, /, main/app.c: Merged revisions 180032 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r180032 | dvossel | 2009-03-03 17:21:18 -0600 (Tue, 03 Mar 2009) | 14 lines app_read does not break from prompt loop with user terminated empty string In app.c, ast_app_getdata is called to stream the prompts and receive DTMF input. If ast_app_getdata() receives an empty string caused by the user inputing the end of string character, in this case '#', it should break from the prompt loop and return to app_read, but instead it cycles through all the prompts. I've added a return value for this special case in ast_readstring() which uses an enum I've delcared in apps.h. This enum is now used as a return value for ast_app_getdata(). (closes issue #14279) Reported by: Marquis Patches: fix_app_read.patch uploaded by Marquis (license 32) read-ampersanmd.patch2 uploaded by dvossel (license 671) Tested by: Marquis, dvossel Review: http://reviewboard.digium.com/r/177/ ........ 2009-03-03 23:31 +0000 [r180077] Steve Murphy * main/ast_expr2.fl, main/ast_expr2.c, utils/Makefile, utils/expr2.testinput, /, main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2f.c: Merged revisions 179973 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179973 | murf | 2009-03-03 15:12:02 -0700 (Tue, 03 Mar 2009) | 33 lines Merged revisions 179807 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 I had some work to do to port these changes to trunk; the check_expr stuff hasn't been updated here for quite some time, it appears. I added some more tests to the check_expr2 suite. I had to play around with the makefile a bit, etc. I added STANDALONE2 #ifdefs to ast_expr2.y so as not to conflict structure with aelparse. ........ r179807 | murf | 2009-03-03 11:11:34 -0700 (Tue, 03 Mar 2009) | 19 lines These changes allow AEL to better check ${} constructs within $[...], that are concatenated with text. I modified and added rules in ast_expr2.fl to better handle the concatenations. I added some default routines to ast_expr2.y so the standalone would compile. It also looks like I haven't run this thru bison since 2.1, so it's good to get this updated. The Makefile has comments added now for check_expr2 and check_expr to explain what they are for, and how to run them. The testexpr2s stuff has been removed, in favor of check_expr2. expr2.testinput has been updated to include the two expressions that inspired these changes (from mcnobody on #asterisk this morning) The regression has been run and all looks well. ........ ................ 2009-03-03 22:49 +0000 [r179939-180009] Mark Michelson * apps/app_queue.c, /, configs/queues.conf.sample: Merged revisions 180007 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180007 | mmichelson | 2009-03-03 16:49:07 -0600 (Tue, 03 Mar 2009) | 22 lines Merged revisions 180006 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180006 | mmichelson | 2009-03-03 16:48:18 -0600 (Tue, 03 Mar 2009) | 17 lines Clarify some documentation of queues.conf.sample It had always been possible to explicitly specify a "blank" value for a sound file in queues.conf and have no sound played back. The problem with this is that it would result in some ugly CLI warnings from file.c. This commit introduces a check when playing a file in app_queue to see if the name of the file is zero-length and return early if that is the case. Also, the ability to specify the blank sound files in queues.conf is now mentioned more clearly in queues.conf.sample (closes issue #14227) Reported by: caspy ........ ................ * doc/timing.txt (added), /, res/res_timing_dahdi.c: Merged revisions 179937 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r179937 | mmichelson | 2009-03-03 14:59:16 -0600 (Tue, 03 Mar 2009) | 20 lines Add documentation for timing modules used in Asterisk This document specifies the timing modules available in Asterisk beginning with Asterisk 1.6.1. The document goes into detail about the differences between each and gives a general overview of what timing is used for in Asterisk. There is also a section which can be used to help customize your setup or to troubleshoot timing issues you may have. I also added messages to the DAHDI timing test used in res_timing_dahdi.c that points to this new documentation if people experience problems. Big thanks to all who contributed comments on this. (closes issue #14490) Reported by: mmichelson Patches: timing.txt uploaded by mmichelson (license 60) Review: http://reviewboard.digium.com/r/164/ ........ 2009-03-03 20:09 +0000 [r179905] Russell Bryant * /, apps/app_directed_pickup.c: Merged revisions 179903 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r179903 | bmd | 2009-03-03 14:02:20 -0600 (Tue, 03 Mar 2009) | 1 line fix a leaked channel lock (and future deadlock) when we try to pick up our own channel ........ 2009-03-03 18:30 +0000 [r179843] Joshua Colp * /, main/features.c: Merged revisions 179841 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179841 | file | 2009-03-03 14:28:46 -0400 (Tue, 03 Mar 2009) | 16 lines Merged revisions 179840 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179840 | file | 2009-03-03 14:27:09 -0400 (Tue, 03 Mar 2009) | 9 lines Do not assume that the bridge_cdr is still attached to the channel when the 'h' exten is finished executing. It is possible for a masquerade operation to occur when the 'h' exten is operating. This operation moves the CDR records around causing the bridge_cdr to no longer exist on the channel where it is expected to. We can not safely modify it afterwards because of this, so don't even try. (closes issue #14564) Reported by: meric ........ ................ 2009-03-03 16:48 +0000 [r179744] Russell Bryant * main/channel.c, /: Merged revisions 179742 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179742 | russell | 2009-03-03 10:47:28 -0600 (Tue, 03 Mar 2009) | 14 lines Merged revisions 179741 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179741 | russell | 2009-03-03 10:45:46 -0600 (Tue, 03 Mar 2009) | 6 lines Ensure chan->fdno always gets reset to -1 after handling a channel fd event. Since setting fdno to -1 had to be moved, a couple of other code paths that do process an fd event return early and do not pass through the code path where it was moved to. So, set it to -1 in a few other places, too. ........ ................ 2009-03-03 14:41 +0000 [r179674] Joshua Colp * main/channel.c, /: Merged revisions 179672 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179672 | file | 2009-03-03 10:40:04 -0400 (Tue, 03 Mar 2009) | 10 lines Merged revisions 179671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179671 | file | 2009-03-03 10:38:09 -0400 (Tue, 03 Mar 2009) | 3 lines Move where fdno is set to the default value to *after* the read callback of the channel driver is called. We have to do this as the underlying channel driver may need the fdno value to determine what to read. ........ ................ 2009-03-03 13:56 +0000 [r179611] Russell Bryant * main/channel.c, /: Merged revisions 179609 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179609 | russell | 2009-03-03 07:54:41 -0600 (Tue, 03 Mar 2009) | 17 lines Merged revisions 179608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179608 | russell | 2009-03-03 07:53:52 -0600 (Tue, 03 Mar 2009) | 9 lines Make it easier to detect an improper call to ast_read(). When you call ast_waitfor() on a channel, the index into the channel fds array that holds the file descriptor that poll() determines has input available is stored in fdno. This patch clears out this value after a call to ast_read() and also reports errors if ast_read() is called without an fdno set. From a discussion on the asterisk-dev list. ........ ................ 2009-03-03 00:04 +0000 [r179539] Jeff Peeler * main/channel.c, /: Merged revisions 179537 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179537 | jpeeler | 2009-03-02 18:01:51 -0600 (Mon, 02 Mar 2009) | 21 lines Merged revisions 179536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179536 | jpeeler | 2009-03-02 17:54:39 -0600 (Mon, 02 Mar 2009) | 15 lines Fix bridging regression from commit 176701 This fixes a bad regression where the bridge would exit after an attended transfer was made. The problem was due to nexteventts getting set after the masquerade which caused the bridge to return AST_BRIDGE_COMPLETE. (closes issue #14315) Reported by: tim_ringenbach ........ ................ 2009-03-02 23:39 +0000 [r179535] Russell Bryant * /, apps/app_meetme.c: Merged revisions 179533 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179533 | russell | 2009-03-02 17:36:38 -0600 (Mon, 02 Mar 2009) | 48 lines Merged revisions 179532 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179532 | russell | 2009-03-02 17:34:13 -0600 (Mon, 02 Mar 2009) | 40 lines Move ast_waitfor() down to avoid the results of the API call becoming stale. This call to ast_waitfor() was being done way too soon in this section of code. Specifically, there was code in between the call to waitfor and the code that uses the result that puts the channel in autoservice. By putting the channel in autoservice, the previous results of ast_waitfor() become meaningless, as the autoservice thread will do it's own ast_waitfor() and ast_read() on the channel. So, when we came back out of autoservice and eventually hit the block of code that calls ast_read() on the channel, there may not actually be any input on the channel available. Even though the previous call to ast_waitfor() in app_meetme said there was input, the autoservice thread has since serviced the channel for some period of time. This bug manifested itself while dvossel was doing some testing of MeetMe in Asterisk trunk. He was using the timerfd timing module. When the code hit ast_read() erroneously, it determined that it must have been called because of input on the timer fd, as chan->fdno was set to AST_TIMING_FD, since that was the cause of the last legitimate call to ast_read() done by autoservice. In this test, an IAX2 channel was calling into the MeetMe conference. It was _much_ more likely to be seen with an IAX2 channel because of the way audio is handled. Every audio frame that comes in results in a call to ast_queue_frame(), which then uses ast_timer_enable_continuous() to notify the channel thread that a frame is waiting to be handled. So, the chances of ast_waitfor() indicating that a channel needs servicing due to a timer event on an IAX2 event is very high. Finally, it is interesting to note that if a different timing interface was being used, this bug would probably not be noticed. When ast_read() is called and erroneously thinks that there is a timer event to handle, it calls the ast_timer_ack() function. The pthread and dahdi timing modules handle the ack() function being called when there is no event by simply ignoring it. In the case of the timerfd module, it results in a read() on the timer fd that will block forever, as there is no data to read. This caused Asterisk to lock up very quickly. Thanks to dvossel and mmichelson for the fun debugging session. :-) ........ ................ 2009-03-02 23:12 +0000 [r179471] Tilghman Lesher * /, main/app.c: Merged revisions 179469 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179469 | tilghman | 2009-03-02 17:10:18 -0600 (Mon, 02 Mar 2009) | 17 lines Merged revisions 179468 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179468 | tilghman | 2009-03-02 17:09:01 -0600 (Mon, 02 Mar 2009) | 10 lines When ending a recording with silence detection, remember to reduce the duration. The end of the recording is correspondingly trimmed, but the duration was not trimmed by the number of seconds trimmed, so the saved duration was necessarily longer than the actual soundfile duration. (closes issue #14406) Reported by: sasargen Patches: 20090226__bug14406.diff.txt uploaded by tilghman (license 14) Tested by: sasargen ........ ................ 2009-03-02 23:04 +0000 [r179464] Russell Bryant * main/channel.c, /: Merged revisions 179462 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179462 | russell | 2009-03-02 17:00:30 -0600 (Mon, 02 Mar 2009) | 16 lines Merged revisions 179461 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179461 | russell | 2009-03-02 16:58:18 -0600 (Mon, 02 Mar 2009) | 8 lines Ensure that only one thread is calling ast_settimeout() on a channel at a time. For example, with an IAX2 channel, you can have both the channel thread and the chan_iax2 processing threads calling this function, and doing so twice at the same time is a bad thing. (Found in a debugging session with dvossel and mmichelson) ........ ................ 2009-03-02 20:18 +0000 [r179407] Jason Parker * /, main/editline/configure, main/editline/np/unvis.c, main/editline/sys.h, main/editline/configure.in: Merged revisions 179396 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r179396 | qwell | 2009-03-02 14:16:51 -0600 (Mon, 02 Mar 2009) | 9 lines Merged revisions 179395 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179395 | qwell | 2009-03-02 14:14:57 -0600 (Mon, 02 Mar 2009) | 1 line Remove several silly warnings in editline. One about a broken preprocessor directive, and another about strlcpy/strlcat. (closes issue #14264) Reported by: dimas ........ ................ 2009-03-02 17:19 +0000 [r179362] Tilghman Lesher * cdr/cdr_sqlite3_custom.c, /: Merged revisions 179361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r179361 | tilghman | 2009-03-02 11:18:48 -0600 (Mon, 02 Mar 2009) | 2 lines Backport 1.6.0 fix to trunk (failsafe if db is not loaded) ........ 2009-03-02 14:14 +0000 [r179293] Joshua Colp * /, main/audiohook.c: Merged revisions 179291 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r179291 | file | 2009-03-02 10:13:45 -0400 (Mon, 02 Mar 2009) | 7 lines Fix issue where changing the volume of both directions of audio did not work. (closes issue #14574) Reported by: KNK Patches: audiohook_volume_fix.diff uploaded by KNK (license 545) ........ 2009-03-01 23:28 +0000 [r179221-179256] Mark Michelson * apps/app_speech_utils.c, /: Merged revisions 179254 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r179254 | mmichelson | 2009-03-01 17:25:23 -0600 (Sun, 01 Mar 2009) | 5 lines Swap reversed timevals. This was pointed out by ScribbleJ in #asterisk-dev. Thanks very much, ScribbleJ! ........ * /, channels/chan_sip.c: Merged revisions 179219 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r179219 | mmichelson | 2009-03-01 15:45:08 -0600 (Sun, 01 Mar 2009) | 18 lines Properly free memory and remove scheduler entries when a transmission failure occurs. Previously, only the "data" field of the sip_pkt created during __sip_reliable_xmit was freed when XMIT_ERROR was returned by __sip_xmit. When retrans_pkt was called, this inevitably resulted in the reading and writing of freed memory. XMIT_ERROR is a condition meaning that we don't want to attempt resending the packet at all. The proper action to take is to remove the scheduler entry we just created, free the packet's data as well as the packet itself, and unlink it from the list of packets on the sip_pvt structure. (closes issue #14455) Reported by: Nick_Lewis Patches: 14455.patch uploaded by mmichelson (license 60) Tested by: Nick_Lewis ........ 2009-02-27 21:48 +0000 [r179166] Russell Bryant * configs/ais.conf.sample, res/res_ais.c, /, doc/distributed_devstate.txt: Merged revisions 179164 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r179164 | russell | 2009-02-27 15:47:18 -0600 (Fri, 27 Feb 2009) | 2 lines Mark res_ais as experimental, as the binary event format is subject to change. ........ 2009-02-27 21:34 +0000 [r179163] Tilghman Lesher * cdr/cdr_sqlite3_custom.c, /: Merged revisions 179161 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r179161 | tilghman | 2009-02-27 15:32:13 -0600 (Fri, 27 Feb 2009) | 3 lines If config file is blank, don't load module. (Closes issue #14563) ........ 2009-02-27 21:25 +0000 [r179160] Russell Bryant * /, UPGRADE.txt: Merged revisions 179154 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r179154 | russell | 2009-02-27 15:23:12 -0600 (Fri, 27 Feb 2009) | 2 lines Add a note about the ordering of entries in sip.conf in 1.6.1. ........ 2009-02-27 19:06 +0000 [r179059] Jason Parker * /, doc/tex/channelvariables.tex: Merged revisions 179057 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r179057 | qwell | 2009-02-27 13:04:57 -0600 (Fri, 27 Feb 2009) | 8 lines Update documentation for DIALEDTIME and ANSWEREDTIME variables. (closes issue #14566) Reported by: klaus3000 Patches: ANSWEREDTIME-1.4-patch.txt uploaded by klaus3000 (license 65) ANSWEREDTIME-trunk-patch.txt uploaded by klaus3000 (license 65) ........ 2009-02-27 03:56 +0000 [r178988] Steve Murphy * configs/features.conf.sample, /, main/features.c: Merged revisions 178986 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r178986 | murf | 2009-02-26 20:45:58 -0700 (Thu, 26 Feb 2009) | 26 lines Merged revisions 178956 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 In this case, it's just a matter of reducing the default timeouts from 2000 to 1000 msec, as the max def feature digit timeout is no longer halved. ........ r178956 | murf | 2009-02-26 14:27:32 -0700 (Thu, 26 Feb 2009) | 18 lines This change moves the default feature digit timeout to 1000 ms from the previous default of 500. As per bug 14515, a dev discussion arrived at a "mediated concensus" of a default feature digit timeout of 1.0 sec. Some voted for 1300; ctooley thought 1500 for distracted phone users in phone booths; kpfleming put his foot down at 1.0 sec. Users who found the previous default max delay of 250 msec perfect, are welcome to override the new default. Notice that I said that 250 msec was the default; wait a minute, you might say, the config file said it was 500 msec!; well, because of the bug fix for 14515, we found that 500 msec was actually enforcing a max of 250. The bug fix would restore 500 msec, but we felt even that was a bit tight for most users... 2000 msec was pushed earlier by mmichelson, so that reduces to 1000 msec after the bug fix. Enjoy! ........ ................ 2009-02-26 17:50 +0000 [r178875] David Vossel * channels/chan_iax2.c, /: Merged revisions 178871 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r178871 | dvossel | 2009-02-26 11:46:12 -0600 (Thu, 26 Feb 2009) | 6 lines IAX2 prune realtime, minor tweak to last fix A return statement was missing which caused unexpected cli output. issue #14479 ........ 2009-02-26 17:38 +0000 [r178869] Steve Murphy * /, main/features.c: Merged revisions 178828 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r178828 | murf | 2009-02-26 10:22:11 -0700 (Thu, 26 Feb 2009) | 34 lines Merged revisions 178804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178804 | murf | 2009-02-26 10:09:03 -0700 (Thu, 26 Feb 2009) | 28 lines This patch prevents the feature detection timeout from being cut in half. Because the ast_channel_bridge() call will return 0 and pass a frame pointer for both DTMF_BEGIN and DTMF_END, the feature_timer field in hte config struct is getting decremented twice, which effectively cuts the digittimeout in half. I added conditions to the if statement to only let DTMF_END frames to flow thru, which solved the problem. Also, when the frame pointer is null, let control flow thru-- this usually happens on timeouts. I added a comment to the code to explain what's going on and why. Many thanks to sodom for reporting this problem. Personnally, it always seemed like something was wrong with the featuredigittimeout, but I never could quite decide what... and was too busy to investigate. This bug forced the issue, and now we know. Sodom had other issues in 14515, but I couldn't reproduce them. If he still has problems, and wants to get them solved, he is welcome to reopen 14515. (closes issue #14515) Reported by: sodom Patches: 14515.patch uploaded by murf (license 17) Tested by: murf, sodom ........ ................ 2009-02-26 16:44 +0000 [r178803] Joshua Colp * /, main/file.c: Merged revisions 178801 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r178801 | file | 2009-02-26 12:42:36 -0400 (Thu, 26 Feb 2009) | 5 lines Fix an issue where the timer for file playback would not be stopped if DAHDI was not installed. (closes issue #14541) Reported by: grant ........ 2009-02-26 16:07 +0000 [r178769] David Vossel * channels/chan_iax2.c, /: Merged revisions 178767 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r178767 | dvossel | 2009-02-26 09:50:22 -0600 (Thu, 26 Feb 2009) | 8 lines IAX2 prune realtime fix Iax2 prune realtime had issues. If "iax2 prune realtime all" was called, it would appear like the command was successful, but in reality nothing happened. This is because the reload that was supposed to take place checks the config files, sees no changes, and does nothing. If there had been a change in the the config file, the realtime users would have been marked for deletion and everything would have been fine. Now prune_users() and prune_peers() are called instead of reload_config() to prune all users/peers that are realtime. These functions remove all users/peers with the rtfriend and delme flags set. iax2_prune_realtime() also lacked the code to properly delete a single friend. For example. if iax2 prune realtime was called, only the peer instance would be removed. The user would still remain. (closes issue #14479) Reported by: mousepad99 Review: http://reviewboard.digium.com/r/176/ ........ 2009-02-25 12:46 +0000 [r178511] Russell Bryant * main/asterisk.c, /: Merged revisions 178509 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r178509 | russell | 2009-02-25 06:45:30 -0600 (Wed, 25 Feb 2009) | 10 lines Merged revisions 178508 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178508 | russell | 2009-02-25 06:43:36 -0600 (Wed, 25 Feb 2009) | 2 lines Update the copyright year for the main page of the doxygen documentation. ........ ................ 2009-02-24 23:28 +0000 [r178383-178448] Tilghman Lesher * configs/extensions.conf.sample, /: Merged revisions 178446 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r178446 | tilghman | 2009-02-24 17:27:23 -0600 (Tue, 24 Feb 2009) | 12 lines Merged revisions 178445 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178445 | tilghman | 2009-02-24 17:25:24 -0600 (Tue, 24 Feb 2009) | 5 lines Add section about the #exec command in configuration files. (closes issue #14540) Reported by: jtodd Patch by: jtodd, with additional notes by tilghman (license 14) ........ ................ * main/asterisk.c, /: Merged revisions 178381 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r178381 | tilghman | 2009-02-24 14:52:44 -0600 (Tue, 24 Feb 2009) | 2 lines Apparently, a void cast doesn't override warn_unused_result. ........ 2009-02-24 20:44 +0000 [r178379-178380] Russell Bryant * Makefile: revert accidental Makefile change. * main/rtp.c, Makefile, /: Merged revisions 178374 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r178374 | russell | 2009-02-24 14:39:57 -0600 (Tue, 24 Feb 2009) | 14 lines Merged revisions 178373 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178373 | russell | 2009-02-24 14:36:19 -0600 (Tue, 24 Feb 2009) | 6 lines Only set dtmfcount on BEGIN, and ensure it gets reset to 0 properly. (issue #14460) Reported by: moliveras Tested by: russell ........ ................ 2009-02-24 20:41 +0000 [r178305-178377] Tilghman Lesher * main/asterisk.c, /: Merged revisions 178375 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r178375 | tilghman | 2009-02-24 14:40:02 -0600 (Tue, 24 Feb 2009) | 2 lines The 3 possible errors with pipe(2) are all impossible in this situation. ........ * main/asterisk.c, /, utils/astcanary.c: Merged revisions 178342 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r178342 | tilghman | 2009-02-24 14:06:48 -0600 (Tue, 24 Feb 2009) | 2 lines Use a SIGPIPE to kill the process, instead of depending upon the astcanary process being inherited by init. ........ * /, utils/astcanary.c: Merged revisions 178303 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r178303 | tilghman | 2009-02-24 11:51:36 -0600 (Tue, 24 Feb 2009) | 7 lines Cause astcanary to exit if Asterisk exits abnormally and doesn't kill astcanary. Also, add some documentation supporting the use of astcanary. (closes issue #14538) Reported by: KNK Patches: asterisk-1.6.x-astcanary.diff uploaded by KNK (license 545) ........ 2009-02-24 15:22 +0000 [r178232] Joshua Colp * /, channels/chan_sip.c: Merged revisions 178213 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r178213 | file | 2009-02-24 11:18:38 -0400 (Tue, 24 Feb 2009) | 16 lines Merged revisions 178205 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178205 | file | 2009-02-24 11:16:07 -0400 (Tue, 24 Feb 2009) | 9 lines Skip check for extension when subscribing for MWI. Since the remote side is not actually subscribing to a specific extension when subscribing for MWI just skip the check to see if the extension exists. They can't use it to specify the mailbox either since we require configuration of that in sip.conf (closes issue #14531) Reported by: festr ........ ................ 2009-02-23 23:22 +0000 [r178172] Russell Bryant * main/rtp.c, /: Merged revisions 178142 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r178142 | russell | 2009-02-23 17:11:37 -0600 (Mon, 23 Feb 2009) | 22 lines Merged revisions 178141 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178141 | russell | 2009-02-23 17:09:01 -0600 (Mon, 23 Feb 2009) | 14 lines Fix infinite DTMF when a BEGIN is received without an END. This commit is related to rev 175124 of 1.4 where a previous attempt was made to fix this problem. The problem with the previous patch was that the inserted code needed to go _before_ setting the lastrxts to the current timestamp. Because those were the same, the dtmfcount variable was never decremented, and so the END was never sent. In passing, I removed the dtmfsamples variable which was completed unused. I also removed a redundant setting of the lastrxts variable. (closes issue #14460) Reported by: moliveras ........ ................ 2009-02-21 16:04 +0000 [r177945] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 177944 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177944 | tilghman | 2009-02-21 09:59:49 -0600 (Sat, 21 Feb 2009) | 2 lines On update, test against the existence of sipregs. ........ 2009-02-21 12:51 +0000 [r177851] Michiel van Baak * /, channels/chan_sip.c: Merged revisions 177849 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177849 | mvanbaak | 2009-02-21 13:22:32 +0100 (Sat, 21 Feb 2009) | 2 lines make chan_sip.c compile on OpenBSD again. ........ 2009-02-20 23:05 +0000 [r177789] Tilghman Lesher * main/pbx.c, /: Merged revisions 177787 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r177787 | tilghman | 2009-02-20 17:02:35 -0600 (Fri, 20 Feb 2009) | 16 lines Merged revisions 177786 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177786 | tilghman | 2009-02-20 16:59:52 -0600 (Fri, 20 Feb 2009) | 9 lines Don't print the CR-NL combination when we aren't outputting to the manager. An embedded CR-NL in a CLI command screws up several AMI parsers that don't expect to see that combination in the middle of output. (Closes issue #14305) Reported by: martins Patch by: tilghman ........ ................ 2009-02-20 22:27 +0000 [r177785] Dwayne M. Hubbard * /, apps/app_fax.c: Merged revisions 177699 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177699 | dhubbard | 2009-02-20 14:29:00 -0600 (Fri, 20 Feb 2009) | 9 lines Make app_fax compatible with spandsp-0.0.6pre4 Prior to spandsp-0.0.6pre4 the t30_stats_t structure used a pages_transferred integer to indicate the number of pages transferred (so far) during the fax session. The spandsp-0.0.6pre4 release removed the pages_transferred integer and replaced it with two different integers - pages_tx and pages_rx. This revision uses the new integers for spandsp-0.0.6pre4 while maintaining backwards compatibility for previous spandsp releases. ........ 2009-02-20 22:15 +0000 [r177760-177764] Tilghman Lesher * include/asterisk/strings.h: Oops, last merge broke 1.6.1 branch * apps/app_system.c, include/asterisk/app.h, /, main/app.c: Merged revisions 177664 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177664 | tilghman | 2009-02-20 11:29:51 -0600 (Fri, 20 Feb 2009) | 8 lines Allow semicolons to be escaped, when passing arguments to the System command. (closes issue #14231) Reported by: jcovert Patches: 20090113__bug14231__2.diff.txt uploaded by Corydon76 (license 14) corrected_20090113__bug14231__2.diff.txt uploaded by jcovert (license 551) Tested by: jcovert ........ * include/asterisk/threadstorage.h, /: Merged revisions 177732 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r177732 | tilghman | 2009-02-20 15:25:37 -0600 (Fri, 20 Feb 2009) | 10 lines Merged revisions 177701 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177701 | tilghman | 2009-02-20 15:15:01 -0600 (Fri, 20 Feb 2009) | 3 lines This exception does not appear to still be true for Solaris 10, and OpenSolaris definitely needs it to be removed. Fixed for snuff-home on -dev channel. ........ ................ 2009-02-20 20:34 +0000 [r177700] David Vossel * channels/chan_iax2.c, include/asterisk/frame.h: Fixes issue with undefined audio codecs in chan_iax2 During iax2 call negotiation, supported codecs are passed in an Information Element containing a 2 byte field where each bit correlates to a specific codec. In 1.6 only audio codec bits 0-12 are defined, leaving bits 13-14 undefined. By default all bits are enabled unless specified otherwise. Since its a 2 byte field and 13-14 are not defined, these bits are never turned off. In trunk, bits 13-14 are defined, which means 1.6 is advertising support for codecs it does not have when talking to trunk. I fixed this by adding #define for undefined audio codec bits. These bits are then removed from iax2's full bandwidth capabilities. (closes issue #14283) Reported by: jcovert 2009-02-20 17:28 +0000 [r177663] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 177661 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177661 | tilghman | 2009-02-20 11:22:19 -0600 (Fri, 20 Feb 2009) | 2 lines Oops, merge broke trunk ........ 2009-02-20 00:38 +0000 [r177626] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 177624 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177624 | jpeeler | 2009-02-19 18:35:53 -0600 (Thu, 19 Feb 2009) | 7 lines Set sip_request ast_str data to NULL so ast_str_copy allocates space properly in copy_request (issue #14478) Reported by: erik_dedecker ........ 2009-02-20 00:26 +0000 [r177623] Steve Murphy * /, main/Makefile, main/ast_expr2f.c: Merged revisions 177595 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r177595 | murf | 2009-02-19 16:56:50 -0700 (Thu, 19 Feb 2009) | 32 lines Merged revisions 177540 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 Trunk was already pretty 8-bit clean; but I'm still removing the --full from the flex command so everything is uniform. ........ r177540 | murf | 2009-02-19 15:51:37 -0700 (Thu, 19 Feb 2009) | 21 lines This patch fixes a problem with 8-bit input to the ast_expr2 scanner. The real culprit was the --full argument to flex in the Makefile! This causes a 7-bit scanner to be generated. I reviewed the rules and found one rule where I needed to specifically include 8-bit chars for a token. I tested against the text supplied by ibercom, and all looks very well. This has been there a surprisingly long time! (closes issue #14498) Reported by: ibercom Patches: 14498.patch uploaded by murf (license 17) Tested by: murf ........ ................ 2009-02-19 22:35 +0000 [r177539] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 177537 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r177537 | tilghman | 2009-02-19 16:33:00 -0600 (Thu, 19 Feb 2009) | 14 lines Merged revisions 177536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177536 | tilghman | 2009-02-19 16:26:01 -0600 (Thu, 19 Feb 2009) | 7 lines Fix up potential crashes, by reducing the sharing between interactive and non-interactive threads. (closes issue #14253) Reported by: Skavin Patches: 20090219__bug14253.diff.txt uploaded by Corydon76 (license 14) Tested by: Skavin ........ ................ 2009-02-19 16:46 +0000 [r177389] Jeff Peeler * /, include/asterisk/channel.h: Merged revisions 177387 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177387 | jpeeler | 2009-02-19 10:45:02 -0600 (Thu, 19 Feb 2009) | 3 lines Fix another merge error from 176708 ........ 2009-02-19 16:40 +0000 [r177386] Joshua Colp * apps/app_speech_utils.c, /: Merged revisions 177384 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r177384 | file | 2009-02-19 12:38:41 -0400 (Thu, 19 Feb 2009) | 10 lines Merged revisions 177383 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177383 | file | 2009-02-19 12:37:25 -0400 (Thu, 19 Feb 2009) | 3 lines If we are able to create a speech structure unset the ERROR variable in case it was previously set. (issue #LUMENVOX-13) ........ ................ 2009-02-19 15:57 +0000 [r177358] Jeff Peeler * /, main/features.c: Merged revisions 177356 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177356 | jpeeler | 2009-02-19 09:56:31 -0600 (Thu, 19 Feb 2009) | 4 lines Fix mismerge from revision 176708 pointed out by Kaloyan Kovachev on the asterisk-dev mailing list. Thanks! ........ 2009-02-19 00:17 +0000 [r177294] Steve Murphy * /, res/ael/ael.tab.c, res/ael/ael.y: Merged revisions 177286 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r177286 | murf | 2009-02-18 16:50:57 -0700 (Wed, 18 Feb 2009) | 39 lines Merged revisions 177225 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177225 | murf | 2009-02-18 15:43:14 -0700 (Wed, 18 Feb 2009) | 34 lines This patch fixes a regression of sorts that was introduced in rev 24425. It basically fixes AST-190/ABE-1782. What was wrong: the user has 6000 extensions in one context; and then 6000 contexts, one per extension. The parser could only handle about 4893 of the 6000 extens in the single context. This was due to the regression I mentioned. To get rid of shift/reduce conflicts, Luigi set up right-recursive lists for globals, context elements, switch lists, and statements. Right recursive lists got rid of the warnings, but instead, they use up a tremendous amount of stack space when the lists are long. I saw this a few years back, and resolved not to fix it until someone complained. That day has arrived! After the changes were made, I ran the regression test suite, and there were no problems. I took the test case the user provided, and added 100,000 extensions to the single context, that already had 6,000 extens in it. (I'll see your 6, and raise you 100!) It takes a few minutes to read it all in, check it and generate code for it, but no problems. So, I think I can say that fundamentally, there are no longer any limits on the number of items you can place in contexts, statement blocks, switches, or globals, beyond your virt mem constraints. ........ ................ 2009-02-18 23:15 +0000 [r177230] Kevin P. Fleming * main/frame.c, /: Merged revisions 177229 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177229 | kpfleming | 2009-02-18 17:09:58 -0600 (Wed, 18 Feb 2009) | 3 lines fix two very minor bugs: if anyone ever uses SLINEAR16 as a format in RTP, ensure that the samples are byte-swapped to network order if needed. also, when a smoother is operating on a format that has a sample rate other than 8000 samples per second, use the proper sample rate for computing delivery timestamps. ........ 2009-02-18 23:03 +0000 [r177228] David Vossel * /, main/features.c: Merged revisions 177226 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177226 | dvossel | 2009-02-18 16:51:38 -0600 (Wed, 18 Feb 2009) | 9 lines Locking issue in action_bridge and bridge_exec action_bridge() and bridge_exec() both search for the channels to bridge to, and then immediately drop the lock. Instead, they should hold the lock until the masquerade is complete. This will guarantee the channel remains and prevent any other weirdness from occurring. In action_bridge() some more weirdness comes into play. Both channels are needlessly locked at the same time and perform the exact same logic. It makes sense from a coding organizational standpoint, but could cause a theoretical deadlock so I split the code up. There is an issue associated with this, but since its a rather complicated thing to reproduce I'm not certain this alone will close it. issue# 14296 Review: http://reviewboard.digium.com/r/167/ ........ 2009-02-18 20:16 +0000 [r177164] Jeff Peeler * channels/h323/chan_h323.h, channels/h323/cisco-h225.cxx, channels/h323/compat_h323.cxx, autoconf/ast_check_pwlib.m4, channels/h323/cisco-h225.h, /, channels/h323/caps_h323.cxx, channels/h323/ast_ptlib.h (added), channels/h323/ast_h323.cxx, configure, channels/h323/compat_h323.h, configure.ac, channels/h323/caps_h323.h, autoconf/ast_prog_sed.m4, channels/h323/ast_h323.h: Merged revisions 177162 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177162 | jpeeler | 2009-02-18 14:11:57 -0600 (Wed, 18 Feb 2009) | 14 lines Modify h323 to build against PTLib as well as the older PWLib Several changes in PTLib have occurred requiring build time detection. Changes accounted for include the library name change, config option change, install location change, and a boolean type change which is handled by ast_ptlib.h. Also, the sed check has been modified to properly work with autoconf >= 2.62. (closes issue #14224) Reported by: bergolth Patches: asterisk-autoconf-sed.patch uploaded by bergolth (license 661) asterisk-pwlib-v3.patch uploaded by bergolth (license 661) Tested by: jpeeler ........ 2009-02-18 19:30 +0000 [r177158] Russell Bryant * /, apps/app_meetme.c: Merged revisions 177101 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177101 | russell | 2009-02-18 13:12:49 -0600 (Wed, 18 Feb 2009) | 8 lines Re-add 'o' option to MeetMe, reverting rev 62297. Enabling this option by default proved to be a bad idea, as the talker detection is not very reliable. So, make it optional again, and off by default. (issue #13801) Reported by: justdave ........ 2009-02-18 19:09 +0000 [r177100] Tilghman Lesher * /, include/asterisk/config.h: Merged revisions 177098 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r177098 | tilghman | 2009-02-18 13:05:15 -0600 (Wed, 18 Feb 2009) | 9 lines Merged revisions 177096 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177096 | tilghman | 2009-02-18 12:30:38 -0600 (Wed, 18 Feb 2009) | 2 lines Document the return value of the update method (as requested on -dev list) ........ ................ 2009-02-18 17:26 +0000 [r177037] Doug Bailey * /, main/utils.c: Merged revisions 177035 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177035 | dbailey | 2009-02-18 11:24:07 -0600 (Wed, 18 Feb 2009) | 2 lines Fixed error where a check for an zero length, terminated string was needed. ........ 2009-02-18 17:14 +0000 [r177007] Joshua Colp * /, channels/chan_sip.c: Merged revisions 177005 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177005 | file | 2009-02-18 13:11:52 -0400 (Wed, 18 Feb 2009) | 6 lines Fix ordering of output for a ChannelUpdate manager event. (closes issue #14497) Reported by: vinsik Patches: chan_update_fix-chan_sip.c.diff uploaded by vinsik (license 623) ........ 2009-02-18 16:20 +0000 [r176962] Doug Bailey * /, main/utils.c: Merged revisions 176948 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176948 | dbailey | 2009-02-18 10:09:12 -0600 (Wed, 18 Feb 2009) | 2 lines Need to take into account the \0 terminator of the old string to determine the amount available. ........ 2009-02-18 15:59 +0000 [r176946] Steve Murphy * main/pbx.c, /: Merged revisions 176943 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176943 | murf | 2009-02-18 08:35:26 -0700 (Wed, 18 Feb 2009) | 45 lines This patch fixes merge_contexts_and_delete so it does not deadlock when hints are present. Reason: when I re-engineered the merge_and_delete func to reduce its lock time, I failed to notice that the functions it calls still also do locking as before. This leads to deadlocks on dialplan reloads, when there are actually living, subscribed hints registered in the system. While the reporter come across this problem while using AEL, I might note that these deadlocks should also happen if extensions.conf were used. Here I added these routines to pbx.c: ast_add_extension_nolock add_pri_lockopt ast_add_extension2_lockopt find_context add_hint_nolock All of the above routines are static and restricted to be used only within pbx.c, and more specifically within the merge_contexts_and_delete routine. They are pretty much the same as their counterparts except they don't lock contexts or hints. Most of them now do the real work of their name-alike, with optional locking via extra arguments, and are called by their name-alike. The goal was to have the original functions so they would behave exactly as before. Both PJ and I tested these fixes, and the deadlocking problem is no longer encountered. (closes issue #14357) Reported by: pj Patches: 14357.diff uploaded by murf (license 17) Tested by: pj, murf ........ 2009-02-18 06:15 +0000 [r176903-176906] Russell Bryant * include/asterisk/heap.h, /: Merged revisions 176904 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176904 | russell | 2009-02-18 00:14:47 -0600 (Wed, 18 Feb 2009) | 2 lines Add example code for a heap traversal. ........ * main/pbx.c, /: Merged revisions 176901 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176901 | russell | 2009-02-18 00:00:40 -0600 (Wed, 18 Feb 2009) | 9 lines Fix a number of incorrect uses of strncpy(). The big problem here is that the 3rd argument provided in these uses of strncpy() did not reserve a byte for the null terminator, leaving the potential for writing one byte past the end of the buffer. Aside from this, there were coding guidelines violations with regards to spacing, as well as hard coded lengths being used instead of sizeof(). ........ 2009-02-18 00:23 +0000 [r176809] Shaun Ruffell * /, codecs/codec_dahdi.c: Merged revisions 176760 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176760 | sruffell | 2009-02-17 16:28:41 -0600 (Tue, 17 Feb 2009) | 10 lines Several changes to codec_dahdi to play nice with G723. This commit brings in the changes that were living out on the svn/asterisk/team/sruffell/asterisk-trunk-transcoder branch. codec_dahdi.c now always uses signed linear as the simple codec so that a soft g729 codec will not end up being preferred to the hardware codec. There are also changes to allow codec_dahdi.c to feed packets to the hardware in the native sample size of the codec. This solves problems with choppy audio when using G723. ........ 2009-02-17 22:21 +0000 [r176731] Dwayne M. Hubbard * /, channels/chan_sip.c: Merged revisions 176705 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176705 | dhubbard | 2009-02-17 15:59:38 -0600 (Tue, 17 Feb 2009) | 11 lines create a UDPTL structure in create_addr_from_peer() if it does not already exist for T38 This is required to create a UDPTL structure in create_addr_from_peer() to handle the scenario where 't38pt_udptl=yes' is not defined in the [general] section of sip.conf but is defined the peer's context. I tested this patch by enabling t38pt_udptl in the [general] section on one system and only enabling t38pt_udptl in a peer's context on the system sending a fax. Without the patch, the sending system will fail to initiate T38 negotiation with the warning message, "No way to add SDP without an UDPTL structure". When this patch is applied the sending side will successfully initiate T38 negotiation. ........ 2009-02-17 22:15 +0000 [r176711] Jeff Peeler * main/channel.c, /, main/features.c, include/asterisk/channel.h: Merged revisions 176708 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r176708 | jpeeler | 2009-02-17 16:08:00 -0600 (Tue, 17 Feb 2009) | 23 lines Merged revisions 176701 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176701 | jpeeler | 2009-02-17 15:54:34 -0600 (Tue, 17 Feb 2009) | 17 lines Modify bridging to properly evaluate DTMF after first warning is played The main problem is currently if the Dial flag L is used with a warning sound, DTMF is not evaluated after the first warning sound. To fix this, a flag has been added in ast_generic_bridge for playing the warning which ensures that if a scheduled warning is missed, multiple warrnings are not played back (due to a feature evaluation or waiting for digits). ast_channel_bridge was modified to store the nexteventts in the ast_bridge_config structure as that information was lost every time ast_channel_bridge was reentered, causing a hangup due to incorrect time calculations. (closes issue #14315) Reported by: tim_ringenbach Reviewed on reviewboard: http://reviewboard.digium.com/r/163/ ........ ................ 2009-02-17 21:41 +0000 [r176699] Mark Michelson * include/asterisk/frame.h, /: Merged revisions 176697 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176697 | mmichelson | 2009-02-17 15:40:09 -0600 (Tue, 17 Feb 2009) | 3 lines Clear up documentation of AST_FRIENDLY_OFFSET in frame.h ........ 2009-02-17 21:24 +0000 [r176675] Russell Bryant * main/timing.c, main/channel.c, /, res/res_timing_pthread.c, res/res_timing_dahdi.c, include/asterisk/timing.h: Merged revisions 176666 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176666 | russell | 2009-02-17 15:22:40 -0600 (Tue, 17 Feb 2009) | 16 lines Update the timing API to have better support for multiple timing interfaces. 1) Add module use count handling so that timing modules can be unloaded. 2) Implement unload_module() functions for the timing interface modules. 3) Allow multiple timing modules to be loaded, and use the one with the highest priority value. 4) Report which timing module is being use in the "timing test" CLI command. (closes issue #14489) Reported by: russell Review: http://reviewboard.digium.com/r/162/ ........ 2009-02-17 21:16 +0000 [r176644] Tilghman Lesher * res/res_odbc.c, channels/chan_local.c, /: Merged revisions 176592,176642 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176592 | tilghman | 2009-02-17 12:49:20 -0600 (Tue, 17 Feb 2009) | 4 lines Add assertions in the quest to track down a refcount leak. (closes issue #14485) Reported by: davevg ........ r176642 | tilghman | 2009-02-17 15:14:18 -0600 (Tue, 17 Feb 2009) | 8 lines Prior to masquerade, move the group definitions to the channel performing the masq, so that the group count lingers past the bridge. (closes issue #14275) Reported by: kowalma Patches: 20090216__bug14275.diff.txt uploaded by Corydon76 (license 14) Tested by: kowalma ........ 2009-02-17 20:57 +0000 [r176559-176637] Russell Bryant * tests/test_heap.c (added), /: Merged revisions 176635 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176635 | russell | 2009-02-17 14:56:26 -0600 (Tue, 17 Feb 2009) | 4 lines Add a test module for the heap implementation. Review: http://reviewboard.digium.com/r/160/ ........ * include/asterisk/heap.h (added), /, main/Makefile, main/heap.c (added): Merged revisions 176632 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176632 | russell | 2009-02-17 14:51:10 -0600 (Tue, 17 Feb 2009) | 8 lines Add an implementation of the heap data structure. A heap is a convenient data structure for implementing a priority queue. Code from svn/asterisk/team/russell/heap/. Review: http://reviewboard.digium.com/r/160/ ........ * apps/app_queue.c, main/pbx.c, /: Merged revisions 176557 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176557 | russell | 2009-02-17 11:33:38 -0600 (Tue, 17 Feb 2009) | 12 lines Fix a race condition that caused device states to become incorrect for hints. The problem here is that the hint processing code was subscribed to the wrong event type. So, it started processing state for a hint too soon, before the device state cache had been updated. Also, fix a similar bug in app_queue, as it was also subscribed to the wrong event type. (closes issue #14461) Reported by: alecdavis ........ 2009-02-17 14:48 +0000 [r176461-176503] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 176501 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176501 | tilghman | 2009-02-17 08:39:36 -0600 (Tue, 17 Feb 2009) | 3 lines In this version, we can combine the queries, because we support dropping nonexistent columns. ........ * /, channels/chan_sip.c: Merged revisions 176459 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r176459 | tilghman | 2009-02-16 19:58:39 -0600 (Mon, 16 Feb 2009) | 17 lines Merged revisions 176426 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176426 | tilghman | 2009-02-16 18:49:22 -0600 (Mon, 16 Feb 2009) | 10 lines After a 'sip reload', qualifies for realtime peers weren't immediately restarted, instead waiting until the next registration. We're now caching the qualify across a reload/restart and starting the qualify immediately upon loading the peer. (closes issue #14196) Reported by: pdf Patches: 20090120__bug14196_1.4.diff.txt uploaded by pdf (license 663) Tested by: pdf ........ ................ 2009-02-16 23:57 +0000 [r176362] David Vossel * channels/chan_iax2.c, /: Merged revisions 176355 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r176355 | dvossel | 2009-02-16 17:33:55 -0600 (Mon, 16 Feb 2009) | 13 lines Merged revisions 176354 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176354 | dvossel | 2009-02-16 17:30:52 -0600 (Mon, 16 Feb 2009) | 8 lines Fixes issue with AST_CONTROL_SRCUPDATE not being relayed correctly during bridging This should have been committed with rev176247, but I missed it. srcupdate frames no longer break out of the native bridge, but are not being sent to the other call leg either. This fixs that. issue #13749 ........ ................ 2009-02-16 23:17 +0000 [r176321] Tilghman Lesher * /, channels/chan_skinny.c: Merged revisions 176320 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176320 | tilghman | 2009-02-16 17:14:08 -0600 (Mon, 16 Feb 2009) | 7 lines Use the correct list macros for deleting an item from the middle of a list. (issue #13777) Reported by: pj Patches: 20090203__bug13777.diff.txt uploaded by Corydon76 (license 14) Tested by: pj ........ 2009-02-16 22:00 +0000 [r176259] Kevin P. Fleming * include/asterisk/stringfields.h, /, main/utils.c: Merged revisions 176255 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r176255 | kpfleming | 2009-02-16 15:45:54 -0600 (Mon, 16 Feb 2009) | 13 lines Merged revisions 176216 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176216 | kpfleming | 2009-02-16 15:10:38 -0600 (Mon, 16 Feb 2009) | 3 lines fix a flaw in the ast_string_field_build() family of API calls; these functions made no attempt to reuse the space already allocated to a field, so every time the field was written it would allocate new space, leading to what appeared to be a memory leak. ........ r176254 | kpfleming | 2009-02-16 15:41:46 -0600 (Mon, 16 Feb 2009) | 3 lines correct a logic error in the last stringfields commit... don't mark additional space as allocated if the string was built using already-allocated space ........ ................ 2009-02-16 21:50 +0000 [r176257] Mark Michelson * /, apps/app_meetme.c: Merged revisions 176253 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r176253 | mmichelson | 2009-02-16 15:40:40 -0600 (Mon, 16 Feb 2009) | 24 lines Merged revisions 176249,176252 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176249 | mmichelson | 2009-02-16 15:34:27 -0600 (Mon, 16 Feb 2009) | 14 lines Open the DAHDI pseudo device and set it to be nonblocking atomically Apparently on FreeBSD, attempting to set the O_NONBLOCKING flag separately from opening the file was causing an "inappropriate ioctl for device" error. While I cannot fathom why this would be happening, I certainly am not opposed to making the code a bit more compact/efficient if it also fixes a bug. (closes issue #14482) Reported by: ys Patches: meetme.patch uploaded by ys (license 281) Tested by: ys ........ r176252 | mmichelson | 2009-02-16 15:39:21 -0600 (Mon, 16 Feb 2009) | 3 lines Remove unused variable and make dev-mode compilation happy ........ ................ 2009-02-16 21:36 +0000 [r176251] David Vossel * channels/chan_iax2.c, /: Merged revisions 176248 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r176248 | dvossel | 2009-02-16 15:30:17 -0600 (Mon, 16 Feb 2009) | 11 lines Merged revisions 175597 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175597 | dvossel | 2009-02-13 14:11:55 -0600 (Fri, 13 Feb 2009) | 4 lines Fixed iax2 key rotation backwards compatibility Turns key rotation back on by default. Added bit into encryption IE to indicate whether or not key rotation is supported or not. If it is not supported then it is not enabled, which insures backwards compatibility. This eliminates the need for the keyrotate option in iax.conf, so it has been removed. ........ ................ 2009-02-16 18:38 +0000 [r176176] Mark Michelson * /, main/logger.c: Merged revisions 176174 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176174 | mmichelson | 2009-02-16 12:25:57 -0600 (Mon, 16 Feb 2009) | 11 lines Assist proper thread synchronization when stopping the logger thread. I was finding that on my dev box, occasionally attempting to "stop now" in trunk would cause Asterisk to hang. I traced this to the fact that the logger thread was waiting on a condition which had already been signalled. The logger thread also need to be sure to check the value of the close_logger_thread variable. The close_logger_thread variable is only checked when the list of logmessages is empty. This allows for the logger thread to print and free any pending messages before exiting. ........ 2009-02-16 17:10 +0000 [r176102] Russell Bryant * /, channels/chan_features.c (removed): Merged revisions 176100 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176100 | russell | 2009-02-16 11:09:24 -0600 (Mon, 16 Feb 2009) | 4 lines Remove chan_features. Review: http://reviewboard.digium.com/r/161/ ........ 2009-02-16 17:07 +0000 [r176099] Tilghman Lesher * configs/func_odbc.conf.sample: Eliminate mention of a variable which exists only in trunk. (Thanks, jsmith) 2009-02-16 15:38 +0000 [r176032] Joshua Colp * /, channels/chan_sip.c: Merged revisions 176030 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r176030 | file | 2009-02-16 11:36:19 -0400 (Mon, 16 Feb 2009) | 16 lines Merged revisions 176029 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176029 | file | 2009-02-16 11:33:53 -0400 (Mon, 16 Feb 2009) | 9 lines Don't have the Via header stored as a stringfield as it can change often during the lifetime of a dialog. This issue crept up with subscriptions on the AA50. When an outgoing NOTIFY is sent a new branch value is created and the Via header is changed to reflect it. Since this was a stringfield a new spot in the pool was used for the value while the old was left untouched/unused. If the current pool was full a new pool was created. This would cause memory usage to increase steadily. (issue #AA50-2332) ........ ................ 2009-02-16 09:42 +0000 [r176023] Michiel van Baak * include/asterisk/manager.h, doc/unistim.txt, channels/chan_unistim.c, /, channels/chan_sip.c: Merged revisions 175952 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r175952 | mvanbaak | 2009-02-16 01:26:59 +0100 (Mon, 16 Feb 2009) | 10 lines Merged revisions 175921 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175921 | mvanbaak | 2009-02-16 00:37:03 +0100 (Mon, 16 Feb 2009) | 3 lines fix mis-spelling of the word registered. Reported by De_Mon on #asterisk-dev. ........ ................ 2009-02-15 21:28 +0000 [r175831-175890] Russell Bryant * main/sched.c, /, include/asterisk/sched.h: Merged revisions 175882 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175882 | russell | 2009-02-15 15:27:33 -0600 (Sun, 15 Feb 2009) | 2 lines Make ast_sched_report() and ast_sched_dump() thread safe. ........ * main/sched.c, /, channels/chan_sip.c, include/asterisk/sched.h: Merged revisions 175829 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175829 | russell | 2009-02-15 14:56:27 -0600 (Sun, 15 Feb 2009) | 14 lines Fix a number of problems with ast_sched_report(). 1) It had numerous coding guidelines violations with regards to formatting. 2) It allocated memory using ast_calloc() that was never freed. 3) It didn't check for failure from the allocation. 4) It used sprintf() and strcat() to build the result, doing zero checking to prevent writing past the end of the provided buffer. The function also lacks API documentation, but that has not been addressed in this commit. ........ 2009-02-13 20:48 +0000 [r175662] David Vossel * channels/chan_iax2.c, configs/iax.conf.sample, channels/iax2.h: Merged revisions 175597 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175597 | dvossel | 2009-02-13 14:11:55 -0600 (Fri, 13 Feb 2009) | 4 lines Fixed iax2 key rotation backwards compatibility Turns key rotation back on by default. Added bit into encryption IE to indicate whether or not key rotation is supported or not. If it is not supported then it is not enabled, which insures backwards compatibility. This eliminates the need for the keyrotate option in iax.conf, so it has been removed. ........ 2009-02-13 19:52 +0000 [r175593] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 175591 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r175591 | mmichelson | 2009-02-13 13:49:38 -0600 (Fri, 13 Feb 2009) | 22 lines Merged revisions 175590 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175590 | mmichelson | 2009-02-13 13:47:48 -0600 (Fri, 13 Feb 2009) | 16 lines Fix a potential crash situation when using IMAP voicemail If calling into VoiceMailMain when using IMAP storage, it was possible to crash Asterisk by hanging up the phone when prompted for a voicemail mailbox. This patch fixes the issue. While it may appear that this patch is superficial, it allows code execution to continue to the failure case just below the IMAP_STORAGE code block where this patch has been applied (closes issue #14473) Reported by: dwpaul Patches: voicemail_imap_crash_no_mailbox.patch uploaded by dwpaul (license 689) ........ ................ 2009-02-13 16:44 +0000 [r175551] Joshua Colp * /, apps/app_record.c: Merged revisions 175549 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175549 | file | 2009-02-13 12:41:15 -0400 (Fri, 13 Feb 2009) | 4 lines Add an option to keep the recorded file upon hangup. (closes issue #14341) Reported by: fnordian ........ 2009-02-12 21:41 +0000 [r175370] Russell Bryant * /, channels/chan_sip.c: Merged revisions 175368 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175368 | russell | 2009-02-12 15:41:01 -0600 (Thu, 12 Feb 2009) | 2 lines Remove useless string copy, and make sscanf safe again ........ 2009-02-12 21:27 +0000 [r175342] Tilghman Lesher * main/udptl.c, /: Merged revisions 175334 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r175334 | tilghman | 2009-02-12 15:25:14 -0600 (Thu, 12 Feb 2009) | 16 lines Merged revisions 175311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175311 | tilghman | 2009-02-12 15:19:40 -0600 (Thu, 12 Feb 2009) | 9 lines Fix crashes when receiving certain T.38 packets. Also, increase the maximum size of T.38 packets and warn users when they try to set the limits above those maximums. (closes issue #13050) Reported by: schern Patches: 20090212__bug13050.diff.txt uploaded by Corydon76 (license 14) Tested by: schern ........ ................ 2009-02-12 20:51 +0000 [r175300] Jeff Peeler * /, main/features.c: Merged revisions 175298 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r175298 | jpeeler | 2009-02-12 14:48:56 -0600 (Thu, 12 Feb 2009) | 15 lines Merged revisions 175294 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175294 | jpeeler | 2009-02-12 14:34:36 -0600 (Thu, 12 Feb 2009) | 9 lines Fix ParkedCall event information for From field in the case of a blind transfer If the parker information can not be obtained from the peer, try and see if the BLINDTRANSFER channel variable has been set. Previously, a blind transfer to the ParkAndAnnounce app would return nothing for the From. Closes AST-189 ........ ................ 2009-02-12 20:48 +0000 [r175257-175297] Russell Bryant * /, channels/chan_sip.c: Merged revisions 175295 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175295 | russell | 2009-02-12 14:45:47 -0600 (Thu, 12 Feb 2009) | 2 lines Avoid using ast_strdupa() in a loop. ........ * build_tools/cflags.xml, /: Merged revisions 175255 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175255 | russell | 2009-02-12 13:11:08 -0600 (Thu, 12 Feb 2009) | 4 lines Don't enable something by default that has a dependency on something _not_ enabled by default. menuselect was not happy with this. ........ 2009-02-12 18:50 +0000 [r175251] Kevin P. Fleming * channels/chan_iax2.c, /: Merged revisions 175250 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175250 | kpfleming | 2009-02-12 12:48:52 -0600 (Thu, 12 Feb 2009) | 1 line correct warning message to not refer specifically to DAHDI ........ 2009-02-12 18:01 +0000 [r175190] Jeff Peeler * /, main/features.c: Merged revisions 175188 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r175188 | jpeeler | 2009-02-12 12:00:11 -0600 (Thu, 12 Feb 2009) | 12 lines Merged revisions 175187 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175187 | jpeeler | 2009-02-12 11:57:10 -0600 (Thu, 12 Feb 2009) | 6 lines Fix crash in event of failed attempt to transfer to parking The peer may not necessarily exist, such as in the case of a transfer to ParkAndAnnounce. In this case don't try to play a sound to it. ........ ................ 2009-02-12 17:09 +0000 [r175130] David Vossel * channels/chan_iax2.c, /: Merged revisions 175127 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175127 | dvossel | 2009-02-12 11:07:17 -0600 (Thu, 12 Feb 2009) | 4 lines Setting key rotation to be off by default Key rotation breaks compatibility between (trunk/1.6.1) and (1.2/1.4/1.6.0). As a follow up to this, I am investigating possible ways to allow key rotation to be on by default and not affect the other branches, but for now it must be turned off. ........ 2009-02-12 17:08 +0000 [r175129] Russell Bryant * main/rtp.c, /: Merged revisions 175125 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r175125 | russell | 2009-02-12 10:57:25 -0600 (Thu, 12 Feb 2009) | 35 lines Merged revisions 175124 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175124 | russell | 2009-02-12 10:51:13 -0600 (Thu, 12 Feb 2009) | 27 lines Don't send DTMF for infinite time if we do not receive an END event. I thought that this was going to end up being a pretty gnarly fix, but it turns out that there was actually already a configuration option in rtp.conf, dtmftimeout, that was intended to handle this situation. However, in between Asterisk 1.2 and Asterisk 1.4, the code that processed the option got lost. So, this commit brings it back to life. The default timeout is 3 seconds. However, it is worth noting that having this be configurable at all is not really the recommended behavior in RFC 2833. From Section 3.5 of RFC 2833: Limiting the time period of extending the tone is necessary to avoid that a tone "gets stuck". Regardless of the algorithm used, the tone SHOULD NOT be extended by more than three packet interarrival times. A slight extension of tone durations and shortening of pauses is generally harmless. Three seconds will pretty much _always_ be far more than three packet interarrival times. However, that behavior is not required, so I'm going to leave it with our legacy behavior for now. Code from svn/asterisk/team/russell/issue_14460 (closes issue #14460) Reported by: moliveras ........ ................ 2009-02-12 16:35 +0000 [r174947-175123] Mark Michelson * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions 175121 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175121 | mmichelson | 2009-02-12 10:28:06 -0600 (Thu, 12 Feb 2009) | 11 lines Make lock information for ao2_trylock be more useful and gnarly Core show locks information involving an ao2_trylock did not show the function that called ao2_trylock, but would instead show ao2_trylock as the source of the lock. This is not useful when trying to debug locking issues. One bizarre note is that this logic is already in 1.4 but somehow did not get merged to trunk or the 1.6.X branches. ........ * apps/app_queue.c, /: Merged revisions 174951 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174951 | mmichelson | 2009-02-11 17:12:57 -0600 (Wed, 11 Feb 2009) | 3 lines Fix a bit of odd logic for announcing position. Sync with 1.6.0's logic ........ * apps/app_queue.c, /: Merged revisions 174948 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174948 | mmichelson | 2009-02-11 17:03:08 -0600 (Wed, 11 Feb 2009) | 20 lines Fix odd "thank you" sound playing behavior in app_queue.c If someone has configured the queue to play an position or holdtime announcement, then it is odd and potentially unexpected to hear a "Thank you for your patience" sound when no position or holdtime was actually announced. This fixes the announcement so that the "thanks" sound is only played in the case that a position or holdtime was actually announced. There is a way that the "thank you" sound can be played without a position or holdtime, and that is to set announce-frequency to a value but keep announce-position and announce-holdtime both turned off. (closes issue #14227) Reported by: caspy Patches: 14227_v3.patch uploaded by putnopvut (license 60) Tested by: caspy ........ * apps/app_dial.c, main/channel.c, main/pbx.c, /, apps/app_dictate.c, apps/app_waitforsilence.c, include/asterisk/channel.h: Merged revisions 174945 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174945 | mmichelson | 2009-02-11 16:41:01 -0600 (Wed, 11 Feb 2009) | 29 lines Fix 'd' option for app_dial and add new option to Answer application The 'd' option would not work for channel types which use RTP to transport DTMF digits. The only way to allow for this to work was to answer the channel if we saw that this option was enabled. I realized that this may cause issues with CDRs, specifically with giving false dispositions and answer times. I therefore modified ast_answer to take another parameter which would tell if the CDR should be marked answered. I also extended this to the Answer application so that the channel may be answered but not CDRified if desired. I also modified app_dictate and app_waitforsilence to only answer the channel if it is not already up, to help not allow for faulty CDR answer times. All of these changes are going into Asterisk trunk. For 1.6.0 and 1.6.1, however, all the changes except for the change to the Answer application will go in since we do not introduce new features into stable branches (closes issue #14164) Reported by: DennisD Patches: 14164.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/145 ........ 2009-02-11 14:46 +0000 [r174846] Joshua Colp * main/channel.c, /: Merged revisions 174844 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174844 | file | 2009-02-11 10:44:47 -0400 (Wed, 11 Feb 2009) | 10 lines Tell the device state core a change happened when a channel is freed but not a specific state. We need to do this because while we know that the freeing of the channel may cause something to become not in use we do not know this for sure. There may be another channel that is still up which would cause it to be in use. (closes issue #13238) Reported by: kowalma Patches: 20090121__bug13238.diff.txt uploaded by Corydon76 (license 14) Tested by: alecdavis ........ 2009-02-10 23:21 +0000 [r174769-174823] Mark Michelson * apps/app_chanspy.c, /: Merged revisions 174805 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174805 | mmichelson | 2009-02-10 17:17:03 -0600 (Tue, 10 Feb 2009) | 11 lines Fix potential for stack overflows in app_chanspy.c When using the 'g' or 'e' options, the stack allocations that were used could cause a stack overflow if a spyer stayed on the line long enough without actually successfully spying on anyone. The problem has been corrected by using static buffers and copying the contents of the appropriate strings into them instead of using functions like alloca or ast_strdupa ........ * main/manager.c, /: Merged revisions 174764 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174764 | mmichelson | 2009-02-10 15:45:14 -0600 (Tue, 10 Feb 2009) | 21 lines Fix an fd leak that would occur in HTTP AMI sessions The explanation behind this fix is a bit complicated, and I've already typed it up in the code as a huge comment inside of manager.c, so I'll give the abridged version here. We needed a way to separate action-specific data from session-specific data. Unfortunately, the only way to maintain API compatibility and to not have to change every single manager action was to rename the current mansession structure and wrap it inside a new mansession structure which actually contains action- specific data. (closes issue #14364) Reported by: awk Patches: 14364_better.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/148/ ........ 2009-02-10 20:17 +0000 [r174714] Joshua Colp * /, channels/chan_sip.c: Merged revisions 174710 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174710 | file | 2009-02-10 16:15:43 -0400 (Tue, 10 Feb 2009) | 4 lines Only decrease inringing count if above zero. (issue #13238) Reported by: kowalma ........ 2009-02-10 18:18 +0000 [r174590] Matthew Nicholson * /, main/jitterbuf.c: Merged revisions 174584 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r174584 | mnicholson | 2009-02-10 12:16:31 -0600 (Tue, 10 Feb 2009) | 25 lines Merged revisions 174583 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174583 | mnicholson | 2009-02-10 11:52:42 -0600 (Tue, 10 Feb 2009) | 18 lines Improve behavior of jitterbuffer when maxjitterbuffer is set. This change improves the way the jitterbuffer handles maxjitterbuffer and dramatically reduces the number of frames dropped when maxjitterbuffer is exceeded. In the previous jitterbuffer, when maxjitterbuffer was exceeded, all new frames were dropped until the jitterbuffer is empty. This change modifies the code to only drop frames until maxjitterbuffer is no longer exceeded. Also, previously when maxjitterbuffer was exceeded, dropped frames were not tracked causing stats for dropped frames to be incorrect, this change also addresses that problem. (closes issue #14044) Patches: bug14044-1.diff uploaded by mnicholson (license 96) Tested by: mnicholson Review: http://reviewboard.digium.com/r/144/ ........ ................ 2009-02-10 17:49 +0000 [r174545-174582] Joshua Colp * /, channels/chan_sip.c: Merged revisions 174580 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174580 | file | 2009-02-10 13:48:29 -0400 (Tue, 10 Feb 2009) | 4 lines Set the type for the peer structure to be a peer as the default. (closes issue #14447) Reported by: triccyx ........ * /, channels/chan_sip.c: Merged revisions 174543 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174543 | file | 2009-02-10 11:37:07 -0400 (Tue, 10 Feb 2009) | 6 lines Make the logic for inuse and inringing manipluation match that of 1.4. The old broken logic would reset the values back to 0 during certain scenarios causing the wrong state to be reported. (closes issue #14399) Reported by: caspy (issue #13238) Reported by: kowalma ........ 2009-02-10 07:07 +0000 [r174471-174504] Tilghman Lesher * apps/app_stack.c, apps/app_voicemail.c, /: Merged revisions 174503 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174503 | tilghman | 2009-02-10 01:06:29 -0600 (Tue, 10 Feb 2009) | 2 lines Fix0ring build ........ * apps/app_stack.c, /: Merged revisions 174470 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174470 | tilghman | 2009-02-09 23:39:33 -0600 (Mon, 09 Feb 2009) | 2 lines Remove the usage of the KeepAlive app, as it no longer exists. ........ 2009-02-10 05:13 +0000 [r174428-174440] Steve Murphy * apps/app_osplookup.c: This patch corrects warnings which seem to appear only on 64-bit compilers, gcc-4.3.2. * apps/app_rpt.c: One final fix in the 1.6.1 release only; some variables the compiler worries "may not be initialized". * apps/app_rpt.c, /: Merged revisions 174435 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174435 | murf | 2009-02-09 21:49:02 -0700 (Mon, 09 Feb 2009) | 8 lines This patch removes the use of AST_PBX_KEEPALIVE from app_rpt.c. (closes issue #14435) Reported by: D_McNaul ........ * apps/app_rpt.c, /: Merged revisions 174432 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174432 | murf | 2009-02-09 21:36:22 -0700 (Mon, 09 Feb 2009) | 3 lines More intptr_t work. ........ * apps/app_rpt.c, /: Merged revisions 174370 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r174370 | murf | 2009-02-09 19:45:56 -0700 (Mon, 09 Feb 2009) | 10 lines Merged revisions 174369 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174369 | murf | 2009-02-09 19:27:40 -0700 (Mon, 09 Feb 2009) | 5 lines This patch solves some compiler complaints in both 32 and 64-bit environments. ........ ................ 2009-02-09 17:47 +0000 [r174330] David Vossel * /, apps/app_externalivr.c: Merged revisions 174325 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174325 | dvossel | 2009-02-09 11:26:02 -0600 (Mon, 09 Feb 2009) | 9 lines Fixes issue with hangups not being sent and external process never terminating. The ignore_hangup, run_dead, and noanswer flags were never initilized to zero causing hangups to never be issued. If the external script expects to be notified of a hangup and never receives one, it runs indefinitely. (closes issue #14251) Reported by: chris-mac Tested by: dvossel ........ 2009-02-09 17:30 +0000 [r174326-174329] Mark Michelson * /, channels/chan_sip.c: Merged revisions 174327 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174327 | mmichelson | 2009-02-09 11:27:32 -0600 (Mon, 09 Feb 2009) | 3 lines Fix something I messed up in the merge I just did ........ * /, channels/chan_sip.c: Merged revisions 174301 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r174301 | mmichelson | 2009-02-09 11:20:55 -0600 (Mon, 09 Feb 2009) | 20 lines Merged revisions 174282 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174282 | mmichelson | 2009-02-09 11:11:05 -0600 (Mon, 09 Feb 2009) | 12 lines Don't do an SRV lookup if a port is specified RFC 3263 says to do A record lookups on a hostname if a port has been specified, so that's what we're going to do. See section 4.2. (closes issue #14419) Reported by: klaus3000 Patches: patch_chan_sip_nosrvifport_1.4.23.txt uploaded by klaus3000 (license 65) ........ ................ 2009-02-09 14:50 +0000 [r174221] Joshua Colp * /, res/res_musiconhold.c: Merged revisions 174219 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r174219 | file | 2009-02-09 10:49:24 -0400 (Mon, 09 Feb 2009) | 11 lines Merged revisions 174218 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174218 | file | 2009-02-09 10:48:21 -0400 (Mon, 09 Feb 2009) | 4 lines Don't overwrite our pointer to the music class when music on hold stops. We will use this if it starts again to see if we can resume the music where it left off. (closes issue #14407) Reported by: mostyn ........ ................ 2009-02-07 16:18 +0000 [r174154] Russell Bryant * /, res/snmp/agent.c: Merged revisions 174149 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r174149 | russell | 2009-02-07 10:16:50 -0600 (Sat, 07 Feb 2009) | 10 lines Merged revisions 174148 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174148 | russell | 2009-02-07 10:15:07 -0600 (Sat, 07 Feb 2009) | 2 lines Fix a race condition that could cause a crash. ........ ................ 2009-02-07 00:09 +0000 [r174086] Dwayne M. Hubbard * /, channels/chan_sip.c: Merged revisions 174084 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r174084 | dhubbard | 2009-02-06 17:51:56 -0600 (Fri, 06 Feb 2009) | 13 lines Merged revisions 174082 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174082 | dhubbard | 2009-02-06 17:36:03 -0600 (Fri, 06 Feb 2009) | 5 lines check ast_strlen_zero() before calling ast_strdupa() in sip_uri_headers_cmp() and sip_uri_params_cmp() The reporter didn't actually upload a properly-formed patch, instead a modified chan_sip.c file was uploaded. I created a patch to determine the changes, then modified the suggested changes to create a proper fix. The summary above is a complete description of the changes. (closes issue #13547) Reported by: tecnoxarxa Patches: chan_sip.c.gz uploaded by tecnoxarxa (license 258) Tested by: tecnoxarxa ........ ................ 2009-02-06 19:30 +0000 [r173994-174043] Joshua Colp * channels/chan_dahdi.c, /: Merged revisions 174041 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174041 | file | 2009-02-06 15:28:53 -0400 (Fri, 06 Feb 2009) | 4 lines Don't subscribe to a mailbox on pseudo channels. It is futile. This solves an issue where duplicated pseudo channels would cause a crash because the first one would unsubscribe and the next one would also try to unsubscribe the same subscription. (closes issue #14322) Reported by: amessina ........ * /, channels/chan_sip.c: Merged revisions 173974 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173974 | file | 2009-02-06 13:18:35 -0400 (Fri, 06 Feb 2009) | 15 lines Merged revisions 173967-173968 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173967 | file | 2009-02-06 13:14:15 -0400 (Fri, 06 Feb 2009) | 4 lines Some clients do not put the call-id for replaces at the beginning, so support it being anywhere in the string. (closes issue #14350) Reported by: fhackenberger ........ r173968 | file | 2009-02-06 13:15:01 -0400 (Fri, 06 Feb 2009) | 2 lines Remove a debug message I put in by accident. ........ ................ 2009-02-06 17:05 +0000 [r173964-173966] Matthew Nicholson * /, channels/chan_sip.c: Merged revisions 173952 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173952 | mnicholson | 2009-02-06 10:28:19 -0600 (Fri, 06 Feb 2009) | 14 lines Merged revisions 173917 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173917 | mnicholson | 2009-02-06 10:20:23 -0600 (Fri, 06 Feb 2009) | 7 lines Limit the addition of the Contact header in SIP responses according to various SIP RFCs. (closes issue #13602) Reported by: hjourdain Tested by: mnicholson ........ ................ * main/ast_expr2.c, /, channels/chan_sip.c, main/ast_expr2.h: revert revision 173964 * main/ast_expr2.c, /, channels/chan_sip.c, main/ast_expr2.h: Merged revisions 173952 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173952 | mnicholson | 2009-02-06 10:28:19 -0600 (Fri, 06 Feb 2009) | 14 lines Merged revisions 173917 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173917 | mnicholson | 2009-02-06 10:20:23 -0600 (Fri, 06 Feb 2009) | 7 lines Limit the addition of the Contact header in SIP responses according to various SIP RFCs. (closes issue #13602) Reported by: hjourdain Tested by: mnicholson ........ ................ 2009-02-06 16:01 +0000 [r173904] Joshua Colp * apps/app_chanspy.c, /, main/audiohook.c: Merged revisions 173902 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r173902 | file | 2009-02-06 11:59:17 -0400 (Fri, 06 Feb 2009) | 4 lines Always detach and destroy the whisper and barge audiohooks. Additionally also allow an audiohook to be detached if it has not been attached. (closes issue #14414) Reported by: bluecrow76 ........ 2009-02-06 10:26 +0000 [r173850] Russell Bryant * main/manager.c, /: Merged revisions 173848 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r173848 | russell | 2009-02-06 04:25:09 -0600 (Fri, 06 Feb 2009) | 2 lines Resolve a memory leak that would occur on an invalid channel given to Action: Status ........ 2009-02-05 23:53 +0000 [r173779] Mark Michelson * configs/extensions.conf.sample, /: Merged revisions 173776 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r173776 | mmichelson | 2009-02-05 17:48:48 -0600 (Thu, 05 Feb 2009) | 14 lines Update extensions.conf.sample to be correct. In trunk, the only necessary change pointed out was that the call to ChanIsAvail uses an option that has been removed. For the 1.6.1 branch, however, it appears that the sample file is badly in need of updating since there are |'s used all over the place there. My tentative plan is just to copy trunk's sample config file to those branches since the info there is most up-to-date and should be correct for use in 1.6.1 Thanks to macli in #asterisk-dev for bringing this up ........ 2009-02-05 23:51 +0000 [r173778] Tilghman Lesher * res/res_config_sqlite.c: Oops, merge from trunk broke 1.6.1 2009-02-05 23:31 +0000 [r173775] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 173773 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r173773 | mmichelson | 2009-02-05 17:28:19 -0600 (Thu, 05 Feb 2009) | 7 lines Properly set "seen" and "unseen" flags when moving messages from the new to the old folder when using IMAP for voicemail storage (closes issue #13905) Reported by: jaroth Patches: foldermove_v2.patch uploaded by jaroth (license 50) ........ 2009-02-05 21:06 +0000 [r173699] Jeff Peeler * apps/app_voicemail.c, /: Merged revisions 173697 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173697 | jpeeler | 2009-02-05 15:00:26 -0600 (Thu, 05 Feb 2009) | 18 lines Merged revisions 173696 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173696 | jpeeler | 2009-02-05 14:47:51 -0600 (Thu, 05 Feb 2009) | 12 lines Add new configuration option to make shared IMAP mailboxes function as expected. The new option is "imapvmshareid" which is an ID to tag multiple mailboxes using the same IMAP storage location to function as one mailbox. This allows all messages to be retrieved for any user in the group. The patch alters the 'X-Asterisk-VM-Extension' header that is responsible for matching voicemails for a given user. (closes issue #13673) Reported by: howardwilkinson ........ ................ 2009-02-05 20:35 +0000 [r173695] Mark Michelson * apps/app_queue.c, /: Merged revisions 173693 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173693 | mmichelson | 2009-02-05 14:30:45 -0600 (Thu, 05 Feb 2009) | 20 lines Merged revisions 173692 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173692 | mmichelson | 2009-02-05 14:29:09 -0600 (Thu, 05 Feb 2009) | 12 lines Fix situations where queue members could be autopaused unexpectedly Specifically, this patch prevents us from autopausing members when we receive a busy or congestion frame from them. (closes issue #14376) Reported by: fiddur Patches: 14376.patch uploaded by putnopvut (license 60) Tested by: fiddur ........ ................ 2009-02-05 19:37 +0000 [r173658] Tilghman Lesher * res/res_config_sqlite.c, /: Merged revisions 173657 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r173657 | tilghman | 2009-02-05 13:36:29 -0600 (Thu, 05 Feb 2009) | 2 lines Change the first field, or we don't get the necessary field separation. ........ 2009-02-05 18:50 +0000 [r173541-173595] Mark Michelson * apps/app_mixmonitor.c, /: Merged revisions 173593 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173593 | mmichelson | 2009-02-05 12:48:55 -0600 (Thu, 05 Feb 2009) | 11 lines Merged revisions 173592 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173592 | mmichelson | 2009-02-05 12:47:24 -0600 (Thu, 05 Feb 2009) | 3 lines Add some missing cleanup to app_mixmonitor ........ ................ * apps/app_mixmonitor.c, /: Merged revisions 173589 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173589 | mmichelson | 2009-02-05 12:34:06 -0600 (Thu, 05 Feb 2009) | 33 lines Merged revisions 173559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173559 | mmichelson | 2009-02-05 11:34:33 -0600 (Thu, 05 Feb 2009) | 25 lines Fix a problem where a channel pointer becomes invalid due to masquerading or hanging up. app_mixmonitor runs its own thread to monitor the channel's activity and write the mixed audio to a file. Since this thread runs independently of the channel, it is possible that the mixmonitor thread's channel pointer will point to freed memory when the channel either is masqueraded or hangs up (technically, both cases are hangups, but we need to handle the cases slightly differently). The solution for this is to employ a datastore, which has the nice benefit of allowing us to hook into channel masquerades and hangups and update our pointer as necessary. If this looks familiar, this same technique is employed in app_chanspy. app_chanspy is a bit more involved since it does a lot more operations on the channel that is being spied upon. app_mixmonitor does have an extra touch that app_chanspy doesn't have, though. Since there is a thread race between the channel's thread and the mixmonitor thread on a hangup, we em- ploy a condition-and-boolean combination to ensure that the channel thread finishes with our structure before the mixmonitor thread attempts to free it. No crashes! (closes issue #14374) Reported by: aragon Patches: 14374.patch uploaded by putnopvut (license 60) Tested by: aragon, putnopvut ........ ................ * apps/app_queue.c, /: Merged revisions 173507 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r173507 | mmichelson | 2009-02-04 16:16:19 -0600 (Wed, 04 Feb 2009) | 7 lines Fix some areas where the incorrect interface was passed to ast_device_state I swear it feels like I already did this once... (closes issue #14359) Reported by: francesco_r ........ 2009-02-04 21:32 +0000 [r173506] David Vossel * channels/chan_iax2.c, channels/iax2-parser.h, /: Merged revisions 173502 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r173502 | dvossel | 2009-02-04 15:25:14 -0600 (Wed, 04 Feb 2009) | 9 lines Fixes issue with IAX2 transfer not handing off calls. Reverts changes in 116884 Fixes issue with IAX2 transfers not taking place. As it was, a call that was being transfered would never be handed off correctly to the call ends because of how call numbers were stored in a hash table. The hash table, "iax_peercallno_pvt", storing all the current call numbers did not take into account the complications associated with transferring a call, so a separate hash table was required. This second hash table "iax_transfercallno_pvt" handles calls being transfered, once the call transfer is complete the call is removed from the transfer hash table and added to the peer hash table resuming normal operations. Addition functions were created to handle storing, removing, and comparing items in the iax_transfercallno_pvt table. The changes reverted in 116884 caused backwards compatibility issues involving iax2 transfer with 1.6.0, 1.4, and 1.2. (closes issue #13468) Reported by: nicox Tested by: dvossel ........ 2009-02-04 21:28 +0000 [r173505] Jeff Peeler * include/asterisk/features.h, /, main/features.c: Merged revisions 173500 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173500 | jpeeler | 2009-02-04 15:17:53 -0600 (Wed, 04 Feb 2009) | 23 lines Merged revisions 173211 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173211 | jpeeler | 2009-02-03 15:57:01 -0600 (Tue, 03 Feb 2009) | 17 lines Parking attempts made to one end of a bridge no longer will hang up due to a parking failure. Parking attempts made using either one-touch, or doing either a blind or assisted transfer to the parking extension now keep up the bridge instead of hanging up the attempted parked party. Normal causes for the parking attempt to fail includes the specific specified extension (via PARKINGEXTEN) not being available or if all the parking spaces are currently in use. To avoid having to reverse a masquerade park_space_reserve was made to provide foresight if a parking attempt will succeed and if so reserve the parking space. (closes issue #13494) Reported by: mdu113 Reviewed by Russell: http://reviewboard.digium.com/r/133/ ........ ................ 2009-02-04 18:52 +0000 [r173459] Tilghman Lesher * main/tcptls.c, /: Merged revisions 173458 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r173458 | tilghman | 2009-02-04 12:48:06 -0600 (Wed, 04 Feb 2009) | 9 lines When using a socket as a FILE *, the stdio functions will sometimes try to do an fseek() on the stream, which is an invalid operation for a socket. Turning off buffering explicitly lets the stdio functions know they cannot do this, thus avoiding a potential error. (closes issue #14400) Reported by: fnordian Patches: tcptls.patch uploaded by fnordian (license 110) ........ 2009-02-04 17:46 +0000 [r173356-173399] Mark Michelson * apps/app_chanspy.c, /: Merged revisions 173397 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173397 | mmichelson | 2009-02-04 11:45:14 -0600 (Wed, 04 Feb 2009) | 11 lines Merged revisions 173396 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173396 | mmichelson | 2009-02-04 11:44:48 -0600 (Wed, 04 Feb 2009) | 3 lines Revert my previous change because it was stupid ........ ................ * apps/app_chanspy.c, /: Merged revisions 173393 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173393 | mmichelson | 2009-02-04 11:41:02 -0600 (Wed, 04 Feb 2009) | 11 lines Merged revisions 173392 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173392 | mmichelson | 2009-02-04 11:40:29 -0600 (Wed, 04 Feb 2009) | 3 lines Add a missing unlock. Extremely unlikely to ever matter, but it's needed. ........ ................ * /, main/file.c: Merged revisions 173354 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r173354 | mmichelson | 2009-02-04 09:30:12 -0600 (Wed, 04 Feb 2009) | 30 lines Fix a problem where file playback would cause fds to remain open forever The problem came from the fact that a frame read from a format interpreter was not freed. Adding a call to ast_frfree fixed this. The explanation for why this caused the problem is a bit complex, but here goes: There was a problem in all versions of Asterisk where the embedded frame of a filestream structure was referenced after the filestream was freed. This was fixed by adding reference counting to the filestream structure. The refcount would increase every time that a filestream's frame pointer was pointing to an actual frame of data. When the frame was freed, the refcount would decrease. Once the refcount reached 0, the filestream was freed, and as part of the operation, the open files were closed as well. Thus it becomes more clear why a missing ast_frfree would cause a reference leak and cause the files to not be closed. You may ask then if there was a frame leak before this patch. The answer to that is actually no! The filestream code was "smart" enough to know that since the frame we received came from a format interpreter, the frame had no malloced data and thus didn't need to be freed. Now, however, there is cleanup that needs to be done when we finish with the frame, so we do need to call ast_frfree on the frame to be sure that the refcount for the filestream is decremented appropriately. (closes issue #14384) Reported by: fiddur Patches: 14384.patch uploaded by putnopvut (license 60) Tested by: fiddur, putnopvut ........ 2009-02-04 00:46 +0000 [r173313] Tilghman Lesher * main/pbx.c, /: Merged revisions 173311 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r173311 | tilghman | 2009-02-03 18:43:52 -0600 (Tue, 03 Feb 2009) | 10 lines Ensure that commas placed in the middle of extension character classes do not interfere with correct parsing of the extension. Also, if an unterminated character class DOES make its way into the pbx core (through some other method), ensure that it does not crash Asterisk. (closes issue #14362) Reported by: Nick_Lewis Patches: 20090129__bug14362.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ 2009-02-03 00:26 +0000 [r173115] Tilghman Lesher * configs/extensions.conf.sample, /: Merged revisions 173104 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173104 | tilghman | 2009-02-02 18:24:52 -0600 (Mon, 02 Feb 2009) | 12 lines Merged revisions 173070 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173070 | tilghman | 2009-02-02 18:15:59 -0600 (Mon, 02 Feb 2009) | 5 lines Add warning to standard config, that globals may be overridden by other dialplan configuration files. (closes issue #14388) Reported by: macli ........ ................ 2009-02-03 00:01 +0000 [r173069] Terry Wilson * /, main/features.c: Merged revisions 173067 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r173067 | twilson | 2009-02-02 17:57:25 -0600 (Mon, 02 Feb 2009) | 9 lines Merged revisions 173066 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173066 | twilson | 2009-02-02 17:48:06 -0600 (Mon, 02 Feb 2009) | 2 lines Fix a feature inheritance bug I added after code review ........ ................ 2009-02-02 18:15 +0000 [r172895] Leif Madsen * /, configs/res_ldap.conf.sample: Merged revisions 172894 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r172894 | lmadsen | 2009-02-02 13:13:40 -0500 (Mon, 02 Feb 2009) | 7 lines Update the res_ldap.conf file with a better working example. (closes issue #13861) Reported by: scramatte Patches: __20080110-res_ldap.conf-2.patch uploaded by blitzrage (license 10) Tested by: jcovert ........ 2009-02-01 02:45 +0000 [r172708-172743] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 172741 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r172741 | tilghman | 2009-01-31 20:44:23 -0600 (Sat, 31 Jan 2009) | 4 lines Blank argument crashes Asterisk (closes issue #14377) Reported by: amorsen ........ * /, funcs/func_strings.c: Merged revisions 172706 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r172706 | tilghman | 2009-01-31 10:40:59 -0600 (Sat, 31 Jan 2009) | 7 lines Don't increment the loop, now that incrementing is taken care of by the decoder function. (closes issue #14363) Reported by: andrew53 Patches: func_strings_filter.patch uploaded by andrew53 (license 519) ........ 2009-01-31 00:07 +0000 [r172636-172638] Terry Wilson * configs/features.conf.sample, /: Merged revisions 172581 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r172581 | twilson | 2009-01-30 15:50:03 -0600 (Fri, 30 Jan 2009) | 2 lines Remove incorret line from sample config ........ * CHANGES, configs/features.conf.sample, apps/app_dial.c, main/global_datastores.c, /, main/features.c, include/asterisk/global_datastores.h: Merged revisions 172580 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r172580 | twilson | 2009-01-30 15:29:12 -0600 (Fri, 30 Jan 2009) | 44 lines Merged revisions 172517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) | 37 lines Fix feature inheritance with builtin features When using builtin features like parking and transfers, the AST_FEATURE_* flags would not be set correctly for all instances when either performing a builtin attended transfer, or parking a call and getting the timeout callback. Also, there was no way on a per-call basis to specify what features someone should have on picking up a parked call (since that doesn't involve the Dial() command). There was a global option for setting whether or not all users who pickup a parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT, AUTOMON, or PARKCALL. This patch: 1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the dialplan or with setvar in channels that support it. This variable can be set to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the equivalent dial options), to set what features should be activated on this channel. The patch moves the setting of the features datastores into the bridging code instead of app_dial to help facilitate this. 2) adds global options parkedcallparking, parkedcallhangup, and parkedcallrecording to be similar to the parkedcalltransfers option for globally setting features. 3) has builtin_atxfer call builtin_parkcall if being transfered to the parking extension since tracking everything through multiple masquerades, etc. is difficult and error-prone 4) attempts to fix all cases of return calls from parking and completed builtin transfers not having the correct permissions (closes issue #14274) Reported by: aragon Patches: fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396) Tested by: aragon, otherwiseguy Review http://reviewboard.digium.com/r/138/ ........ ................ 2009-01-30 22:24 +0000 [r172609] Mark Michelson * /, include/asterisk/channel.h: Merged revisions 172598 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r172598 | mmichelson | 2009-01-30 16:22:04 -0600 (Fri, 30 Jan 2009) | 3 lines Fix redefinition of flag in channel.h ........ 2009-01-30 08:27 +0000 [r172509] Olle Johansson * CHANGES: Remove an extra "the" and restructure a bit 2009-01-29 23:53 +0000 [r172504] Tilghman Lesher * apps/app_rpt.c, main/asterisk.c, /, autoconf/ast_func_fork.m4, configure, main/app.c: Merged revisions 172441 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r172441 | tilghman | 2009-01-29 17:15:40 -0600 (Thu, 29 Jan 2009) | 16 lines Merged revisions 172438 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172438 | tilghman | 2009-01-29 16:54:29 -0600 (Thu, 29 Jan 2009) | 9 lines Lose the CAP_NET_ADMIN at every fork, instead of at startup. Otherwise, if Asterisk runs as a non-root user and the administrator does a 'restart now', Asterisk loses the ability to set QOS on packets. (closes issue #14004) Reported by: nemo Patches: 20090105__bug14004.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ ................ 2009-01-29 22:05 +0000 [r172435] Richard Mudgett * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged revisions 172400 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r172400 | rmudgett | 2009-01-29 14:38:34 -0600 (Thu, 29 Jan 2009) | 12 lines channels/chan_dahdi.c * Added doxygen comments to the major dahdi structures. * Fixed PRI and SS7 using an incorrect string value if the extension delimiter is not present in the Dial() function. * Fixed SS7 not checking if the dialed extension is at least as long as the stripmsd option. * Fixed PRI not handling unknown TON/NPI prefix letters correctly. * Fixed some uninitialized string variables on FXS ports. configs/chan_dahdi.conf.sample * Updated some documentation. ........ 2009-01-29 20:54 +0000 [r172317-172402] Tilghman Lesher * utils/muted.c, /: Merged revisions 146514 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk (closes issue #14360) Reported by: oej ........ r146514 | russell | 2008-10-05 17:11:30 -0500 (Sun, 05 Oct 2008) | 2 lines Make this build on my mac. ........ * configs/func_odbc.conf.sample, /: Merged revisions 172315 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r172315 | tilghman | 2009-01-29 10:48:25 -0600 (Thu, 29 Jan 2009) | 2 lines Better document mode=multirow, based upon a conversation with Jared. ........ 2009-01-29 13:50 +0000 [r172272] Leif Madsen * contrib/scripts/realtime_pgsql.sql, /: Merged revisions 172271 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r172271 | lmadsen | 2009-01-29 08:47:27 -0500 (Thu, 29 Jan 2009) | 5 lines The realtime_pgsql.sql script is missing a couple of fields. closes issue #14339) Reported by: fiddur Patches: realtime_pgsql.sql.diff uploaded by fiddur (license 678) ........ 2009-01-29 11:24 +0000 [r172218-172235] Olle Johansson * /, channels/chan_sip.c: Merged revisions 172234 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r172234 | oej | 2009-01-29 12:19:29 +0100 (Tor, 29 Jan 2009) | 7 lines Make sure register= line supports both port and expiry at the same time. (closes issue #14185) Reported by: Nick_Lewis Patches: chan_sip.c-expiryrequest6.patch uploaded by Nick (license 657) Tested by: Nick_Lewis ........ * /, channels/chan_sip.c: Merged revisions 172173 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r172173 | oej | 2009-01-29 10:18:01 +0100 (Tor, 29 Jan 2009) | 24 lines Merged revisions 172169 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172169 | oej | 2009-01-29 09:48:18 +0100 (Tor, 29 Jan 2009) | 16 lines Make sure that we always add the hangupcause headers. In some cases, the owner was disconnected before we checked for the cause. This patch implements a temporary storage in the pvt and use that instead. The code is based on ideas from code from Adomjan in issue #13385 (Add support for Reason: header) Thanks to Klaus Darillion for testing! (closes issue #14294) related to issue #13385 Reported by: klaus3000 and adomjan Patches: bug14294b.diff uploaded by oej (license 306) Based on 20080829_chan_sip.c-q850reason_header.patch uploaded by adomjan (license 487) Tested by: oej, klaus3000 ........ ................ * /, configs/sip.conf.sample: Merged revisions 171880 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r171880 | oej | 2009-01-28 14:26:31 +0100 (Ons, 28 Jan 2009) | 2 lines Add some more notes about device matching. ........ 2009-01-28 Leif Madsen * Asterisk 1.6.1-rc1 released 2009-01-28 22:52 +0000 [r172133] Tilghman Lesher * res/res_config_odbc.c, /: Merged revisions 172131 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r172131 | tilghman | 2009-01-28 16:48:01 -0600 (Wed, 28 Jan 2009) | 7 lines Fix how we skip fields (to avoid fields which don't exist) when doing an UPDATE. (closes issue #14205) Reported by: maxgo Patches: 20090128__bug14205__5.diff.txt uploaded by Corydon76 (license 14) Tested by: blitzrage ........ 2009-01-28 20:56 +0000 [r172067] Steve Murphy * apps/app_channelredirect.c, main/pbx.c, main/manager.c, /, main/features.c, include/asterisk/channel.h: Merged revisions 172063 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r172063 | murf | 2009-01-28 13:31:06 -0700 (Wed, 28 Jan 2009) | 52 lines Merged revisions 172030 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172030 | murf | 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) | 46 lines This patch fixes h-exten running misbehavior in manager-redirected situations. What it does: 1. A new Flag value is defined in include/asterisk/channel.h, AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the bridge hangup exten code not to run the h-exten there (nor publish the bridge cdr there). It will done at the pbx-loop level instead. 2. In the manager Redirect code, I set this flag on the channel if the channel has a non-null pbx pointer. I did the same for the second (chan2) channel, which gets run if name2 is set... and the first succeeds. 3. I restored the ending of the cdr for the pbx loop h-exten running code. Don't know why it was removed in the first place. 4. The first attempt at the fix for this bug was to place code directly in the async_goto routine, which was called from a large number of places, and could affect a large number of cases, so I tested that fix against a fair number of transfer scenarios, both with and without the patch. In the process, I saw that putting the fix in async_goto seemed not to affect any of the blind or attended scenarios, but still, I was was highly concerned that some other scenarios I had not tested might be negatively impacted, so I refined the patch to its current scope, and jmls tested both. In the process, tho, I saw that blind xfers in one situation, when the one-touch blind-xfer feature is used by the peer, we got strange h-exten behavior. So, I inserted code to swap CDRs and to set the HANGUP_DONT field, to get uniform behavior. 5. I added code to the bridge to obey the HANGUP_DONT flag, skipping both publishing the bridge CDR, and running the h-exten; they will be done at the pbx-loop (higher) level instead. 6. I removed all the debug logs from the patch before committing. 7. I moved the AUTOLOOP set/reset in the h-exten code in res_features so it's only done if the h-exten is going to be run. A very minor performance improvement, but technically correct. (closes issue #14241) Reported by: jmls Patches: 14241_redirect_no_bridgeCDR_or_h_exten_via_transfer uploaded by murf (license 17) Tested by: murf, jmls ........ ................ 2009-01-28 17:29 +0000 [r171966] Tilghman Lesher * channels/chan_dahdi.c, /: Merged revisions 171964 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r171964 | tilghman | 2009-01-28 11:27:40 -0600 (Wed, 28 Jan 2009) | 9 lines Merged revisions 171963 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171963 | tilghman | 2009-01-28 11:25:18 -0600 (Wed, 28 Jan 2009) | 2 lines Clarify log message (suggested by manxpower on #asterisk-dev) ........ ................ 2009-01-28 13:21 +0000 [r171857] Olle Johansson * configs/sip.conf.sample: Merged revisions 171838 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r171838 | oej | 2009-01-28 14:11:44 +0100 (Ons, 28 Jan 2009) | 10 lines Merged revisions 171837 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171837 | oej | 2009-01-28 14:07:27 +0100 (Ons, 28 Jan 2009) | 2 lines Add a better explanation of the difference between the device namespace and the dialplan for newbies. ........ ................ 2009-01-27 22:01 +0000 [r171620-171693] Mark Michelson * /, channels/chan_agent.c: Merged revisions 171691 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r171691 | mmichelson | 2009-01-27 15:58:39 -0600 (Tue, 27 Jan 2009) | 47 lines Merged revisions 171689 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171689 | mmichelson | 2009-01-27 15:55:08 -0600 (Tue, 27 Jan 2009) | 39 lines Fix devicestate problems for "always-on" agent channels A revision to chan_agent attempted to "inherit" the device state of the underlying channel in order to report the device state of an agent channel more accurately. The problem with the logic here is that it makes no sense to use this for always-on agents. If the agent is logged in, then to the underlying channel, the agent will always appear to be "in use," no matter if the agent is on a call or not. The reason is that to the underlying channel, the channel is currently in use on a call to the AgentLogin application. The most common cause that I found for this issue to occur was for a SIP channel to be the underlying channel type for an Agent channel. If the SIP phone re-registers, then the registration will cause the device state core to query the device state of the SIP channel. Since the SIP channel is in use, the Agent channel would also inherit this status. Once the agent channel was set to "in use" there was no way that the device state could change on that channel unless the agent logged out. The solution for this problem is a bit different in 1.4 than it is in the other branches. In 1.4, there will be a one-line fix to make sure that only callback agents will inherit device state from their underlying channel type. For the other branches of Asterisk, since callback support has been removed, there is also no need for device state inheritance in chan_agent, so I will simply be removing it from the code. In addition, the 1.4 source is getting a new comment to help the next person who edits chan_agent.c. I'm adding a comment that a agent_pvt's loginchan field may be used to determine if the agent is a callback agent or not. (closes issue #14173) Reported by: nathan Patches: 14173.patch uploaded by putnopvut (license 60) Tested by: nathan, aramirez ........ ................ * /, main/slinfactory.c: Merged revisions 171622 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r171622 | mmichelson | 2009-01-27 14:11:30 -0600 (Tue, 27 Jan 2009) | 26 lines Merged revisions 171621 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171621 | mmichelson | 2009-01-27 14:06:01 -0600 (Tue, 27 Jan 2009) | 18 lines Prevent a crash from occurring when a jitter buffer interpolated frame is removed from a slinfactory slinfactory used the "samples" field of an ast_frame in order to determine the amount of data contained within the frame. In certain cases, such as jitter buffer interpolated frames, the frame would have a non-zero value for "samples" but have NULL "data" This caused a problem when a memcpy call in ast_slinfactory_read would attempt to access invalid memory. The solution in use here is to never feed frames into the slinfactory if they have NULL "data" (closes issue #13116) Reported by: aragon Patches: 13116.diff uploaded by putnopvut (license 60) ........ ................ * apps/app_queue.c, /: Merged revisions 171618 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r171618 | mmichelson | 2009-01-27 13:30:54 -0600 (Tue, 27 Jan 2009) | 24 lines Fix queue crashes that would occur after the calling channel was masqueraded. The data passed to the end_bridge_callback was assumed to be data which was still stack'd. The problem was that with some call features, attended transfers in particular, a new bridge thread is started once the feature completes, meaning that when the end_bridge_callback is called, the end_bridge_callback_data was invalid. To fix this problem, there are two measures taken 1. Instead of pointing to stacked data, we now used heap-allocated data for passing to the end_bridge_callback in app_queue 2. Since bridges can end multiple times on a single logical call, we wait until the final bridge is broken to actually set any queue variables. This is accomplished through reference-counting and the use of an end_bridge_callback_data_fixup function in app_queue.c (closes issue #14260) Reported by: ccesario Patches: 14260.patch uploaded by putnopvut (license 60) Tested by: ccesario ........ 2009-01-27 15:19 +0000 [r171540] Olle Johansson * /, channels/chan_sip.c: Merged revisions 171528 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r171528 | oej | 2009-01-27 16:00:19 +0100 (Tis, 27 Jan 2009) | 23 lines Solving the same issue, but a bit different in trunk... Merged revisions 171527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171527 | oej | 2009-01-27 15:33:20 +0100 (Tis, 27 Jan 2009) | 13 lines Use the same branch tag in CANCEL as in INVITE Originally putnopvut implemented some changes in revision 142079 that according to the bug report seemed to have worked then, but somehow fails now. I guess code, as humans, get old and forget stuff. Anyway, this bug caused CANCEL not to work with picky systems. Thanks Fredrik for pointing out where the bug in the SIP messaging was. (closes issue #14346) Reported by: oej Patches: bug14346.diff uploaded by oej (license 306) Tested by: oej ........ ................ 2009-01-26 14:58 +0000 [r171361] Olle Johansson * /, channels/chan_sip.c: Merged revisions 171326 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r171326 | oej | 2009-01-26 14:44:40 +0100 (MÃ¥n, 26 Jan 2009) | 17 lines Merged revisions 171264 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171264 | oej | 2009-01-26 13:51:53 +0100 (MÃ¥n, 26 Jan 2009) | 9 lines Don't retransmit 401 on REGISTER requests when alwaysauthreject=yes (closes issue #14284) Reported by: klaus3000 Patches: patch_chan_sip_unreliable_1.4.23_14284.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ ................ 2009-01-26 00:04 +0000 [r171190] Tilghman Lesher * channels/chan_oss.c, /: Merged revisions 171188 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r171188 | tilghman | 2009-01-25 17:58:00 -0600 (Sun, 25 Jan 2009) | 13 lines Merged revisions 171187 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171187 | tilghman | 2009-01-25 17:44:01 -0600 (Sun, 25 Jan 2009) | 6 lines Correctly track the hookstate (closes issue #13686) Reported by: itiliti Patches: 20081013__bug13686.diff.txt uploaded by Corydon76 (license 14) ........ ................ 2009-01-25 13:40 +0000 [r170982] Sean Bright * /, apps/app_page.c: Merged revisions 170980 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170980 | seanbright | 2009-01-25 08:35:48 -0500 (Sun, 25 Jan 2009) | 16 lines Merged revisions 170979 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170979 | seanbright | 2009-01-25 08:33:20 -0500 (Sun, 25 Jan 2009) | 9 lines Resolve a logic error that was causing Page() to crash when more than one channel was specified. (closes issue #14308) Reported by: bluefox Patches: 20090124__bug14308.diff.txt uploaded by seanbright (license 71) Tested by: kc0bvu ........ ................ 2009-01-25 02:52 +0000 [r170945] Russell Bryant * include/asterisk/utils.h, /: Merged revisions 170943 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r170943 | russell | 2009-01-24 20:49:30 -0600 (Sat, 24 Jan 2009) | 6 lines Change ARRAY_LEN() to be more C++ safe. When the second part of this macro is written as 0[a] instead of a[0], it will force a failure if the macro is used on a C++ object that overloads the [] operator. ........ 2009-01-24 13:57 +0000 [r170839] Tilghman Lesher * configs/res_odbc.conf.sample, /: Merged revisions 170837 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170837 | tilghman | 2009-01-24 07:55:53 -0600 (Sat, 24 Jan 2009) | 9 lines Merged revisions 170836 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170836 | tilghman | 2009-01-24 07:55:02 -0600 (Sat, 24 Jan 2009) | 2 lines Remove superfluous implementation note (closes issue #14319) ........ ................ 2009-01-23 23:53 +0000 [r170831] Richard Mudgett * /, doc/tex/Makefile: Merged revisions 170794 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r170794 | rmudgett | 2009-01-23 17:10:34 -0600 (Fri, 23 Jan 2009) | 1 line Fix asterisk.pdf generation if branch name has an underscore in it. ........ 2009-01-23 22:59 +0000 [r170792] Russell Bryant * /, doc/tex/Makefile: Merged revisions 170790 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r170790 | russell | 2009-01-23 16:58:37 -0600 (Fri, 23 Jan 2009) | 2 lines Don't blow up if a branch name has an underscore in it ........ 2009-01-23 20:57 +0000 [r170693-170722] Mark Michelson * configs/res_odbc.conf.sample, /: Merged revisions 170720 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170720 | mmichelson | 2009-01-23 14:56:07 -0600 (Fri, 23 Jan 2009) | 16 lines Merged revisions 170719 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170719 | mmichelson | 2009-01-23 14:55:26 -0600 (Fri, 23 Jan 2009) | 8 lines Add notes to the idlecheck explanation in res_odbc.conf.sample (closes issue #14319) Reported by: klaus3000 Patches: patch_idlecheck_res_odbc.conf.sample.txt uploaded by klaus3000 (license 65) ........ ................ * contrib/i18n.testsuite.conf, /: Merged revisions 170677 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170677 | mmichelson | 2009-01-23 14:23:00 -0600 (Fri, 23 Jan 2009) | 22 lines Merged revisions 170671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170671 | mmichelson | 2009-01-23 14:21:51 -0600 (Fri, 23 Jan 2009) | 14 lines Update contrib/i18n.testsuite.conf to not use deprecated syntax * Convert Wait,1 to Wait(1) * Convert SetLanguage to Set(CHANNEL(language)) * Use 'n' for all priorities beyond the first Also added test for Chinese numbers, too. (closes issue #14320) Reported by: dant Patches: i18n.testsuite.conf.issue14320.v2.diff uploaded by dant (license 670) ........ ................ 2009-01-23 20:20 +0000 [r170664] Joshua Colp * main/channel.c, /: Merged revisions 170652 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170652 | file | 2009-01-23 16:18:05 -0400 (Fri, 23 Jan 2009) | 11 lines Merged revisions 170648 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170648 | file | 2009-01-23 16:16:39 -0400 (Fri, 23 Jan 2009) | 4 lines When a channel is answered make sure any indications currently playing stop. Usually the phone would do this but if the channel was already answered then they are being generated by Asterisk and we darn well need to stop them. (closes issue #14249) Reported by: RadicAlish ........ ................ 2009-01-23 19:37 +0000 [r170637] Tilghman Lesher * channels/chan_iax2.c, /: Merged revisions 170608 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170608 | tilghman | 2009-01-23 13:25:10 -0600 (Fri, 23 Jan 2009) | 9 lines Merged revisions 170588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170588 | tilghman | 2009-01-23 13:20:44 -0600 (Fri, 23 Jan 2009) | 2 lines Additions to AST-2009-001 ........ ................ 2009-01-23 19:10 +0000 [r170507-170571] Joshua Colp * apps/app_dial.c, /: Merged revisions 170569 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170569 | file | 2009-01-23 15:09:18 -0400 (Fri, 23 Jan 2009) | 11 lines Merged revisions 170568 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170568 | file | 2009-01-23 15:06:54 -0400 (Fri, 23 Jan 2009) | 4 lines When a call is forwarded stop any active indications. The new channel will provide an indication, if need be, itself. (closes issue #14310) Reported by: RadicAlish ........ ................ * /, channels/chan_sip.c: Merged revisions 170505 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170505 | file | 2009-01-23 14:09:45 -0400 (Fri, 23 Jan 2009) | 11 lines Merged revisions 170504 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170504 | file | 2009-01-23 14:04:08 -0400 (Fri, 23 Jan 2009) | 4 lines Use the on hold flag to see if the call is on hold or not. It is possible that our address for them will still be valid even though they are on hold. (closes issue #14295) Reported by: klaus3000 ........ ................ 2009-01-23 17:49 +0000 [r170502] Michiel van Baak * /, channels/chan_h323.c: Merged revisions 170501 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r170501 | mvanbaak | 2009-01-23 18:46:02 +0100 (Fri, 23 Jan 2009) | 1 line let's use SENTINEL where needed ........ 2009-01-23 16:35 +0000 [r170458] Doug Bailey * channels/chan_dahdi.c: MWI messages included in CID spill was not being properly handled and prevented the call from being processed (issue #14313) Reported by: seandarcy Tested by: dbailey 2009-01-23 15:51 +0000 [r170395] Mark Michelson * main/channel.c, /: Merged revisions 170393 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170393 | mmichelson | 2009-01-23 09:44:27 -0600 (Fri, 23 Jan 2009) | 36 lines Merged revisions 170392 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170392 | mmichelson | 2009-01-23 09:40:39 -0600 (Fri, 23 Jan 2009) | 28 lines Fix broken call pickup There was a subtle change in ast_do_masquerade which resulted in failed attempts to pickup calls. The problem was that the value of the AST_FLAG_OUTGOING flag was copied from the clone to the original channel. In the case of call pickup, this meant that the AST_FLAG_OUTGOING flag ended up being cleared on the channel that was attempting to execute the pickup. Because this flag was not set, when ast_read came across an answer frame, it ignored it. The result of this was that the calling channel was never properly answered. This fix changes the behavior in ast_do_masquerade to set the flags on the original channel to the union of the flags on the clone channel. This way, if the AST_FLAG_OUTGOING flag is set on either of the two channels involved in the masquerade, the resulting channel will have the flag set as well. (closes issue #14206) Reported by: francesco_r Patches: 14206.patch uploaded by putnopvut (license 60) Tested by: francesco_r, aragon, putnopvut ........ ................ 2009-01-22 20:06 +0000 [r170242] Joshua Colp * main/rtp.c, /: Merged revisions 170240 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170240 | file | 2009-01-22 16:04:39 -0400 (Thu, 22 Jan 2009) | 14 lines Merged revisions 170239 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170239 | file | 2009-01-22 16:02:35 -0400 (Thu, 22 Jan 2009) | 7 lines Don't crash if RTCP is not enabled on an RTP structure but statistics are output. (closes issue #14234) Reported by: jcovert Patches: rtp.c.patch-1.6.0.3 uploaded by jcovert (license 551) rtp.c.patch-svn-165599 uploaded by jcovert (license 551) ........ ................ 2009-01-22 17:21 +0000 [r170178] Tilghman Lesher * pbx/pbx_config.c, /: Merged revisions 170165 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170165 | tilghman | 2009-01-22 11:19:28 -0600 (Thu, 22 Jan 2009) | 13 lines Merged revisions 170158 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170158 | tilghman | 2009-01-22 11:18:07 -0600 (Thu, 22 Jan 2009) | 6 lines Allow global variables after substitution to be as long as other variables. (closes issue #14263) Reported by: markd Patches: 20090120__bug14263.diff.txt uploaded by Corydon76 (license 14) ........ ................ 2009-01-22 16:54 +0000 [r170049-170150] Joshua Colp * /, apps/app_meetme.c: Merged revisions 170148 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170148 | file | 2009-01-22 12:52:21 -0400 (Thu, 22 Jan 2009) | 11 lines Merged revisions 170147 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170147 | file | 2009-01-22 12:50:54 -0400 (Thu, 22 Jan 2009) | 4 lines If we are unable to request a DAHDI pseudo channel and we are using the user introduction without review option make sure it gets unset so other code does not blindly assume a DAHDI pseudo channel exists. (closes issue #14282) Reported by: cheesegrits ........ ................ * main/pbx.c, /: Merged revisions 170051 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r170051 | file | 2009-01-22 11:14:50 -0400 (Thu, 22 Jan 2009) | 13 lines Merged revisions 170050 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170050 | file | 2009-01-22 11:13:56 -0400 (Thu, 22 Jan 2009) | 6 lines Do a string comparison instead of pointer comparison since some people specify the context they are actually in as an argument to get around some funkiness. (closes issue #14011) Reported by: dveiga Patches: pbx.c.patch uploaded by dveiga (license 665) ........ ................ * apps/app_parkandannounce.c, /: Merged revisions 170047 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r170047 | file | 2009-01-22 11:01:54 -0400 (Thu, 22 Jan 2009) | 4 lines Clear the autoloop flag when parsing and setting the context/extension/priority to go back to. When the channel executes a PBX again we want it to start out at the point we explicitly say and at that point it will not yet be doing autoloop. (closes issue #14304) Reported by: jcovert ........ 2009-01-22 00:46 +0000 [r169946] Tilghman Lesher * /, include/asterisk/linkedlists.h: Merged revisions 169944 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r169944 | tilghman | 2009-01-21 18:44:52 -0600 (Wed, 21 Jan 2009) | 16 lines Merged revisions 169943 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r169943 | tilghman | 2009-01-21 18:43:31 -0600 (Wed, 21 Jan 2009) | 9 lines AST_RWLOCK_INIT_VALUE is always defined. What we really wanted to ask is whether autoconf detected a static initializer value. This fixes rwlocks on all such platforms (mainly, Mac OS X). (closes issue #13767) Reported by: jcovert Patches: 20090121__bug13767.diff.txt uploaded by Corydon76 (license 14) Tested by: jcovert, Corydon76 ........ ................ 2009-01-21 23:28 +0000 [r169871] Joshua Colp * main/pbx.c, /: Merged revisions 169869 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r169869 | file | 2009-01-21 19:25:27 -0400 (Wed, 21 Jan 2009) | 11 lines Merged revisions 169867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r169867 | file | 2009-01-21 19:20:47 -0400 (Wed, 21 Jan 2009) | 4 lines Read lock the contexts to maintain the locking order when we are notified that the state of a device has changed. (closes issue #13839) Reported by: mcallist ........ ................ 2009-01-21 22:23 +0000 [r169830] Michiel van Baak * /, doc/tex/extensions.tex: Merged revisions 169793 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r169793 | mvanbaak | 2009-01-21 23:04:16 +0100 (Wed, 21 Jan 2009) | 2 lines remove duplicated sentence. ........ 2009-01-21 22:11 +0000 [r169792-169796] Mark Michelson * /, main/say.c: Merged revisions 169794 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r169794 | mmichelson | 2009-01-21 16:10:02 -0600 (Wed, 21 Jan 2009) | 17 lines Fix a crash when saying certain numbers in Chinese This commit fixes a crash that was occurring when attempting to say a number between 10000 and 100000 due to dividing by 0. This also removes some places where a "zero" is spoken when it should not be. (closes issue #14291) Reported by: dant Patches: say.c-14291.diff uploaded by dant (license 670) Tested by: dant ........ * /, channels/chan_sip.c: Merged revisions 169791 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r169791 | mmichelson | 2009-01-21 15:53:55 -0600 (Wed, 21 Jan 2009) | 18 lines Further fix some oddities in sip show users and sip show peers logic ccesario on IRC pointed out that his sip peers were not displayed properly when he would issue the command "sip show peers." The problem was that the onlymatchonip field was used to determine if the endpoint was a "peer" or "user." The tricky part is that a "friend" is supposed to be treated as both a "user" and a "peer" but the logic would not allow "friends" to show up as "peers" since onlymatchonip was set to FALSE for friends. I have modified the sip_peer structure to more explicitly keep track of what type endpoint it is so that the various manager and CLI commands will display the expected information Reported by ccesario via IRC Tested by ccesario ........ 2009-01-21 21:05 +0000 [r169725] Tilghman Lesher * main/asterisk.c, /: Merged revisions 169723 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r169723 | tilghman | 2009-01-21 15:03:40 -0600 (Wed, 21 Jan 2009) | 15 lines Merged revisions 169722 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r169722 | tilghman | 2009-01-21 15:02:32 -0600 (Wed, 21 Jan 2009) | 8 lines Extra NULLs in the output cause some terminal types to abort in the middle of a color code, causing terminal weirdness. (closes issue #14130) Reported by: coolmig Patches: 20090121__bug14130.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76, coolmig ........ ................ 2009-01-21 17:40 +0000 [r169674] Steve Murphy * utils/refcounter.c, /: Merged revisions 169673 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r169673 | murf | 2009-01-21 10:21:40 -0700 (Wed, 21 Jan 2009) | 14 lines This patch corrects a segfault reported in 14289, due to a null ptr being refd. Yes, seanbright is right in the bug comments, that is the fix. Sorry for this oversight; I guess my personal usage didn't have this happen! murf (closes issue #14289) Reported by: jamesgolovich ........ 2009-01-21 10:49 +0000 [r169622-169626] Russell Bryant * /: Merged revisions 169625 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r169625 | russell | 2009-01-21 04:49:00 -0600 (Wed, 21 Jan 2009) | 2 lines Remove properties that erroneously got merged into trunk ........ * main/tcptls.c, /: Merged revisions 169620 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r169620 | russell | 2009-01-21 04:26:07 -0600 (Wed, 21 Jan 2009) | 10 lines Fix a regression in TCP support. This patch fixes a problem that caused chan_sip to think that every open TCP session was to a remote address of 0.0.0.0:0. (closes issue #14287) Reported by: jamesgolovich Patches: bug-14287.diff.txt uploaded by jamesgolovich (license 176) ........ 2009-01-21 00:35 +0000 [r169559-169613] Mark Michelson * apps/app_queue.c, /: Merged revisions 169611 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r169611 | mmichelson | 2009-01-20 18:33:32 -0600 (Tue, 20 Jan 2009) | 22 lines Fix device state parsing issues for channel names with multiple slashes The fix being applied is a bit different for trunk and the 1.6.X branches. For trunk, we only wish to strip off the characters beyond the second slash if the channel is a Local channel (i.e. we are removing the /n from the device name). Other channel technologies with multiple slashes (e.g. DAHDI) need the information after the second slash in order to get the proper device state information. In addition to this fix, the 1.6.X branches are receiving a much more important fix as well. The problem in 1.6.X is that the member's device name was being directly changed instead of having a copy changed. This meant that we would strip off the second slash and trailing characters and then leave the member's device name like that permanently thereafter. (closes issue #14014) Reported by: kebl0155 Patches: 14014_number2.patch uploaded by putnopvut (license 60) Tested by: kebl0155 ........ * apps/app_queue.c, /: Merged revisions 169574 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r169574 | mmichelson | 2009-01-20 15:57:24 -0600 (Tue, 20 Jan 2009) | 6 lines Use the default timeout for a queue instead of -1 (closes issue #14272) Reported by: timking ........ * /, channels/chan_sip.c: Merged revisions 169557 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r169557 | mmichelson | 2009-01-20 14:10:31 -0600 (Tue, 20 Jan 2009) | 19 lines Convert the character pointers in a sip_request to be pointer offsets When an ast_str expands to hold more data, any pointers that were pointing to the data prior to the expansion will be pointing at invalid memory. This change makes such pointers used in chan_sip.c instead be offsets from the beginning of the string so that the same math may be applied no matter where in memory the string resides. To help ease this transition, a macro called REQ_OFFSET_TO_STR has been added to chan_sip.c so that given a sip_request and an offset, the string at that offset is returned. (closes issue #14220) Reported by: riksta Tested by: putnopvut Review http://reviewboard.digium.com/r/126/ ........ 2009-01-20 19:31 +0000 [r169488-169554] Terry Wilson * /, main/features.c: Merged revisions 169510 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r169510 | twilson | 2009-01-20 13:22:24 -0600 (Tue, 20 Jan 2009) | 7 lines Make a proper builtin attended transfer to parking work This is an ugly hack from 1.4 that allows the timeout callback from a parked call to use the right channel name for the callback when the park is done with a builtin attended transfer (that isn't completed early). This hasn't ever worked in trunk and no one has complained yet, so eh. ........ * /, main/features.c: Merged revisions 169486 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r169486 | twilson | 2009-01-20 12:48:14 -0600 (Tue, 20 Jan 2009) | 13 lines Merged revisions 169485 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r169485 | twilson | 2009-01-20 12:40:56 -0600 (Tue, 20 Jan 2009) | 6 lines Don't play audio to the channel if we've masqueraded (closes issue #14066) Reported by: bluefox Tested by: otherwiseguy, bluefox ........ ................ 2009-01-19 20:10 +0000 [r169368] Tilghman Lesher * main/manager.c, /, apps/app_userevent.c: Merged revisions 169365 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r169365 | tilghman | 2009-01-19 14:05:52 -0600 (Mon, 19 Jan 2009) | 11 lines Merged revisions 169364 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r169364 | tilghman | 2009-01-19 13:49:25 -0600 (Mon, 19 Jan 2009) | 4 lines Truncate userevents at the end of a line, when the command exceeds the buffer. (closes issue #14278) Reported by: fnordian ........ ................ 2009-01-19 15:55 +0000 [r169213] Mark Michelson * channels/chan_local.c, /: Merged revisions 169211 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r169211 | mmichelson | 2009-01-19 09:54:06 -0600 (Mon, 19 Jan 2009) | 21 lines Merged revisions 169210 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r169210 | mmichelson | 2009-01-19 09:52:15 -0600 (Mon, 19 Jan 2009) | 13 lines Prevent a crash in chan_local due to a potential NULL pointer dereference Move the check for if both channels on a local_pvt have generators to below where p->chan is checked for NULLity (NULLness?). This prevents a crash from occurring if p->chan is NULL. (closes issue #14189) Reported by: sascha Patches: 14189.patch uploaded by putnopvut (license 60) Tested by: sascha ........ ................ 2009-01-17 18:46 +0000 [r169154] Doug Bailey * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Add discriminator for when ring pulse alert signal is used to preface MWI spills This prevents the situation when MWI messages are added to caller ID spills causing the channel to be hung up 2009-01-17 01:59 +0000 [r168981-169082] Terry Wilson * main/tcptls.c, /, main/http.c, include/asterisk/tcptls.h: Merged revisions 169080 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r169080 | twilson | 2009-01-16 19:56:36 -0600 (Fri, 16 Jan 2009) | 8 lines Fix qualify for TCP peer (closes issue #14192) Reported by: pabelanger Patches: asterisk-bug14192.diff.txt uploaded by jamesgolovich (license 176) Tested by: jamesgolovich ........ * /, channels/chan_sip.c: Merged revisions 169044 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r169044 | twilson | 2009-01-16 18:03:39 -0600 (Fri, 16 Jan 2009) | 8 lines Fix port :0 added to SIP INVITE URI when outboundproxy used (closes issue #14233) Reported by: chris-mac Patches: asterisk-bug14233.diff.txt uploaded by jamesgolovich (license 176) Tested by: jamesgolovich, chris-mac, otherwiseguy ........ * /, main/features.c: Merged revisions 168941 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168941 | twilson | 2009-01-16 16:16:23 -0600 (Fri, 16 Jan 2009) | 19 lines Merged revisions 168716 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168716 | twilson | 2009-01-15 12:22:49 -0600 (Thu, 15 Jan 2009) | 12 lines Convert call to park_call_full to masq_park_call_announce Since we removed the AST_PBX_KEEPALIVE return value, we need to use masqueraded parking, otherwise we will try to call ast_hangup() in __pbx_run() and in do_parking_thread() and then promptly crash. (closes issue #14215) Reported by: waverly360 Tested by: otherwiseguy (closes issue #14228) Reported by: kobaz Tested by: otherwiseguy ........ ................ 2009-01-16 22:46 +0000 [r168979] Mark Michelson * /, channels/chan_sip.c: Merged revisions 168976 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168976 | mmichelson | 2009-01-16 16:43:09 -0600 (Fri, 16 Jan 2009) | 26 lines Merged revisions 168975 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168975 | mmichelson | 2009-01-16 16:42:13 -0600 (Fri, 16 Jan 2009) | 18 lines Account for possible NULL pointer when we receive a 408 in response to a REGISTER It may be that by the time we receive a reply to a REGISTER request, the attempt has timed out and thus the registry structure pointed to by the corresponding sip_pvt has gone away. This situation was handled properly for a 200 OK response, but the 408 case assumed that the sip_registry struct was non-NULL, thus potentially causing a crash This commit fixes this assumption and prints out a message to the console if we should receive a late 408 response to a REGISTER (closes issue #14211) Reported by: aborghi Patches: 14211.diff uploaded by putnopvut (license 60) Tested by: aborghi ........ ................ 2009-01-16 18:55 +0000 [r168836] Tilghman Lesher * include/asterisk/say.h, apps/app_voicemail.c, /, main/say.c: Merged revisions 168832 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168832 | tilghman | 2009-01-16 12:49:09 -0600 (Fri, 16 Jan 2009) | 13 lines Merged revisions 168828 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168828 | tilghman | 2009-01-16 12:41:35 -0600 (Fri, 16 Jan 2009) | 6 lines Fix the conjugation of Russian and Ukrainian languages. (related to issue #12475) Reported by: chappell Patches: vm_multilang.patch uploaded by chappell (license 8) ........ ................ 2009-01-16 00:47 +0000 [r168739-168748] Steve Murphy * res/ael/pval.c, /: Merged revisions 168746 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168746 | murf | 2009-01-15 17:34:31 -0700 (Thu, 15 Jan 2009) | 20 lines Merged revisions 168745 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168745 | murf | 2009-01-15 17:19:12 -0700 (Thu, 15 Jan 2009) | 14 lines This patch fixes a problem where a goto (or jump, in this case) fails a consistency check because it can't find a matching extension. The problem was a missing instruction to end the range notation in the code where it converts the pattern into a regex and uses the regex code to determine the match. I tested using the AEL code the user supplied, and now, the consistency check passes. (closes issue #14141) Reported by: dimas ........ ................ * main/ast_expr2.c, /, main/ast_expr2.h, main/ast_expr2.y: Merged revisions 168737 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r168737 | murf | 2009-01-15 13:54:59 -0700 (Thu, 15 Jan 2009) | 16 lines This patch allows null args in ast_expr2 func calls, and fixes commas being converted to pipes, which was 1.4 type stuff. If the user says count=ENUMLOOKUP(${EXTEN},ALL,c,,enum.mydomain.tld); then it won't complain about the empty arg (c,,...) and fabled's patch won't let it swap the commas for pipes. Ran it thru my dialplan and no complaints. (closes issue #14169) Reported by: fabled Patches: function-argument-separator-fix.diff uploaded by fabled (license 448) ........ 2009-01-15 19:17 +0000 [r168729] Mark Michelson * channels/chan_sip.c: Merged revisions 168728 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r168728 | mmichelson | 2009-01-15 13:16:29 -0600 (Thu, 15 Jan 2009) | 3 lines Fix the compactheaders option in sip.conf ........ 2009-01-15 19:05 +0000 [r168727] Olle Johansson * /, configs/extconfig.conf.sample: Merged revisions 168722 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168722 | oej | 2009-01-15 19:47:14 +0100 (Tor, 15 Jan 2009) | 10 lines Merged revisions 168721 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168721 | oej | 2009-01-15 19:43:43 +0100 (Tor, 15 Jan 2009) | 2 lines Meetme actually has realtime but wasn't documented ........ ................ 2009-01-15 19:00 +0000 [r168726] Mark Michelson * /, channels/chan_sip.c: Merged revisions 168725 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r168725 | mmichelson | 2009-01-15 13:00:06 -0600 (Thu, 15 Jan 2009) | 17 lines Remove an unneeded condition for line addition to a SIP request/response In Asterisk 1.4 and 1.6.0, the sip_request structure had a statically allocated buffer to hold the text of the request. There was a check in the add_line function to not attempt to write the line into the buffer if we did not have room for it. In trunk and Asterisk versions starting with 1.6.1, an expandable ast_str structure is used to hold the text. Since it may grow to fit an arbitrarily sized string, this check in add_line is no longer valid. I found this oddity while attempting to fix issue #14220; however, I do not believe that this is the fix for that issue since the output supplied by the reporter did not contain the warning message that would be printed had this condition been satisfied. ........ 2009-01-15 18:20 +0000 [r168714-168715] Olle Johansson * /, configs/sip.conf.sample: Merged revisions 168711 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r168711 | oej | 2009-01-15 18:55:53 +0100 (Tor, 15 Jan 2009) | 4 lines Clarify some misunderstandings and make it even more clear that you can refer to a peer in the register= line. ........ * /, channels/chan_sip.c: Merged revisions 168712 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r168712 | oej | 2009-01-15 19:08:59 +0100 (Tor, 15 Jan 2009) | 3 lines Make sure that we have the same terminology in sip.conf.sample and the source code warning. Thanks Nick Lewis for pointing this out in the bug tracker. ........ 2009-01-15 15:37 +0000 [r168707] Sean Bright * /, apps/app_meetme.c: Merged revisions 168705 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r168705 | seanbright | 2009-01-15 10:33:18 -0500 (Thu, 15 Jan 2009) | 11 lines Add a missing unlock and properly handle the 'maxusers' setting on MeetMe conferences. We were using the 'user number' field to compare against the maximum allowed users, which works assuming users with lower user numbers didn't leave the conference. (closes issue #14117) Reported by: sergedevorop Patches: 20090114__bug14117-2.diff.txt uploaded by seanbright (license 71) Tested by: sergedevorop ........ 2009-01-15 00:15 +0000 [r168631] Mark Michelson * apps/app_queue.c, /: Merged revisions 168629 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168629 | mmichelson | 2009-01-14 18:14:17 -0600 (Wed, 14 Jan 2009) | 24 lines Merged revisions 168628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168628 | mmichelson | 2009-01-14 18:11:01 -0600 (Wed, 14 Jan 2009) | 16 lines Fix some crashes from bad datastore handling in app_queue.c * The queue_transfer_fixup function was searching for and removing the datastore from the incorrect channel, so this was fixed. * Most datastore operations regarding the queue_transfer datastore were being done without the channel locked, so proper channel locking was added, too. (closes issue #14086) Reported by: ZX81 Patches: 14086v2.patch uploaded by putnopvut (license 60) Tested by: ZX81, festr ........ ................ 2009-01-14 21:55 +0000 [r168625] Richard Mudgett * channels/misdn/isdn_lib.c, /: Merged revisions 168623 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168623 | rmudgett | 2009-01-14 15:51:06 -0600 (Wed, 14 Jan 2009) | 11 lines Merged revisions 168622 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168622 | rmudgett | 2009-01-14 15:48:22 -0600 (Wed, 14 Jan 2009) | 4 lines * Fixed create_process() allocation of process ID values. The allocated process IDs could overflow their respective NT and TE fields. Affects outgoing calls. ........ ................ 2009-01-14 21:30 +0000 [r168621] Steve Murphy * /, apps/app_page.c: Merged revisions 168613 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168613 | murf | 2009-01-14 13:51:26 -0700 (Wed, 14 Jan 2009) | 9 lines Merged revisions 168608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168608 | murf | 2009-01-14 12:34:35 -0700 (Wed, 14 Jan 2009) | 1 line app_page was failing to compile in dev-mode on my gcc-4.2.4 system. This change gets rid of the warning. ........ ................ 2009-01-14 21:00 +0000 [r168618] Sean Bright * contrib/scripts/autosupport, /: Merged revisions 168615 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168615 | seanbright | 2009-01-14 15:58:26 -0500 (Wed, 14 Jan 2009) | 16 lines Merged revisions 168614 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168614 | seanbright | 2009-01-14 15:52:00 -0500 (Wed, 14 Jan 2009) | 9 lines Update autosupport script to supply info for both Zaptel and DAHDI in 1.4 and be sure to run dahdi_test in 1.6.x and trunk instead of zttest. (closes issue #14132) Reported by: dsedivec Patches: asterisk-1.4-autosupport.patch uploaded by dsedivec (license 638) asterisk-trunk-autosupport.patch uploaded by dsedivec (license 638) ........ ................ 2009-01-14 20:18 +0000 [r168611] Mark Michelson * /, channels/chan_sip.c: Merged revisions 168610 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r168610 | mmichelson | 2009-01-14 14:13:48 -0600 (Wed, 14 Jan 2009) | 9 lines Restore the "sip show users" and "sip show user" CLI commands (closes issue #14180) Reported by: amorsen Patches: sip_show_users_161v3.diff uploaded by putnopvut (license 60) Tested by: blitzrage, amorsen ........ 2009-01-14 19:12 +0000 [r168606] Tilghman Lesher * main/udptl.c, /: Merged revisions 168604 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168604 | tilghman | 2009-01-14 13:11:14 -0600 (Wed, 14 Jan 2009) | 14 lines Merged revisions 168603 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168603 | tilghman | 2009-01-14 13:02:55 -0600 (Wed, 14 Jan 2009) | 7 lines Don't read into a buffer without first checking if a value is beyond the end. (closes issue #13600) Reported by: atis Patches: 20090106__bug13600.diff.txt uploaded by Corydon76 (license 14) Tested by: atis ........ ................ 2009-01-14 02:11 +0000 [r168582-168596] Terry Wilson * /, apps/app_page.c: Merged revisions 168594 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168594 | twilson | 2009-01-13 20:00:40 -0600 (Tue, 13 Jan 2009) | 27 lines Merged revisions 168593 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168593 | twilson | 2009-01-13 19:27:18 -0600 (Tue, 13 Jan 2009) | 20 lines Don't overflow when paging more than 128 extensions The number of available slots for calls in app_page was hardcoded to 128. Proper bounds checking was not in place to enforce this limit, so if more than 128 extensions were passed to the Page() app, Asterisk would crash. This patch instead dynamically allocates memory for the ast_dial structures and removes the (non-functional) arbitrary limit. This issue would have special importance to anyone who is dynamically creating the argument passed to the Page application and allowing more than 128 extensions to be added by an outside user via some external interface. The patch posted by a_villacis was slightly modified for some coding guidelines and other cleanups. Thanks, a_villacis! (closes issue #14217) Reported by: a_villacis Patches: 20080912-asterisk-app_page-fix-buffer-overflow.patch uploaded by a (license 660) Tested by: otherwiseguy ........ ................ * /, res/res_http_post.c: Merged revisions 168588 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r168588 | twilson | 2009-01-13 17:05:43 -0600 (Tue, 13 Jan 2009) | 5 lines Fully overwrite a same-named file when uploading (closes issue #14190) Reported by: timking ........ * /, channels/chan_sip.c: Merged revisions 168578 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168578 | twilson | 2009-01-13 16:22:34 -0600 (Tue, 13 Jan 2009) | 14 lines Merged revisions 168551 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168551 | twilson | 2009-01-13 12:34:14 -0600 (Tue, 13 Jan 2009) | 7 lines Don't pass a value with a side effect to a macro (closes issue #14176) Reported by: paraeco Patches: chan_sip.c.diff uploaded by paraeco (license 658) ........ ................ 2009-01-13 19:35 +0000 [r168565] Russell Bryant * main/indications.c, main/channel.c, channels/chan_misdn.c, channels/chan_skinny.c, funcs/func_channel.c, main/app.c, res/snmp/agent.c, res/res_indications.c, channels/chan_unistim.c, main/pbx.c, apps/app_read.c, /, include/asterisk/indications.h, apps/app_readexten.c, apps/app_disa.c, include/asterisk/channel.h: Merged revisions 168562 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168562 | russell | 2009-01-13 13:22:13 -0600 (Tue, 13 Jan 2009) | 10 lines Merged revisions 168561 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168561 | russell | 2009-01-13 13:13:05 -0600 (Tue, 13 Jan 2009) | 2 lines Revert unnecessary indications API change from rev 122314 ........ ................ 2009-01-13 17:52 +0000 [r168528-168549] Tilghman Lesher * /, funcs/func_logic.c: Merged revisions 168547 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168547 | tilghman | 2009-01-13 11:51:12 -0600 (Tue, 13 Jan 2009) | 13 lines Merged revisions 168546 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168546 | tilghman | 2009-01-13 11:48:00 -0600 (Tue, 13 Jan 2009) | 6 lines If either conditional is NULL, don't try copying it. (closes issue #14226) Reported by: caspy Patches: 20090113__bug14226.diff.txt uploaded by Corydon76 (license 14) ........ ................ * /, channels/chan_alsa.c: Merged revisions 168526 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168526 | tilghman | 2009-01-12 17:45:51 -0600 (Mon, 12 Jan 2009) | 12 lines Merged revisions 167095 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167095 | tilghman | 2008-12-31 18:01:22 -0600 (Wed, 31 Dec 2008) | 5 lines Repeat attempts to write when we receive -EAGAIN from the driver, as detailed in the ALSA sample code (see http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html#a32) Reported by: Jerry Geis (via the -users list) Fixed by: me (license 14) ........ ................ 2009-01-12 23:13 +0000 [r168524] Mark Michelson * main/srv.c, /: Merged revisions 168523 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r168523 | mmichelson | 2009-01-12 17:12:30 -0600 (Mon, 12 Jan 2009) | 11 lines bump the verbosity of a message in srv.c up by one. It used to be at this level prior to a large patch merge which converted ast_verbose calls to ast_verb (closes issue #14221) Reported by: jcovert Patches: srv.c.patch uploaded by jcovert (license 551) ........ 2009-01-12 22:00 +0000 [r168510-168519] Jeff Peeler * /, res/res_agi.c: Merged revisions 168517 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168517 | jpeeler | 2009-01-12 15:51:46 -0600 (Mon, 12 Jan 2009) | 12 lines Merged revisions 168516 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168516 | jpeeler | 2009-01-12 15:42:34 -0600 (Mon, 12 Jan 2009) | 5 lines (closes issue #13881) Reported by: hoowa Update the app CDR field for AGI commands that are not executing an application via "exec". ........ ................ * /, channels/chan_agent.c: Merged revisions 168508 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168508 | jpeeler | 2009-01-12 14:53:04 -0600 (Mon, 12 Jan 2009) | 15 lines Merged revisions 168507 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168507 | jpeeler | 2009-01-12 14:26:22 -0600 (Mon, 12 Jan 2009) | 9 lines (closes issue #12269) Reported by: IgorG Tested by: denisgalvao This gits rid of the notion of an owning_app allowing the request and hangup to be initiated by different threads. Originating from an active agent channel requires this. The implementation primarily changes __login_exec to wait on a condition variable rather than a lock. Review: http://reviewboard.digium.com/r/35/ ........ ................ 2009-01-12 17:26 +0000 [r168500] Olle Johansson * /, apps/app_minivm.c: Merged revisions 168497 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r168497 | oej | 2009-01-12 17:31:27 +0100 (MÃ¥n, 12 Jan 2009) | 2 lines Better to use the proper app name ........ 2009-01-12 15:05 +0000 [r168488] Mark Michelson * /, channels/chan_sip.c: Merged revisions 168486 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ ........ 2009-01-12 14:58 +0000 [r168484] Russell Bryant * /, configs/indications.conf.sample: Merged revisions 168481 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168481 | russell | 2009-01-12 08:57:49 -0600 (Mon, 12 Jan 2009) | 10 lines Merged revisions 168480 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168480 | russell | 2009-01-12 08:57:27 -0600 (Mon, 12 Jan 2009) | 2 lines s/ringdance/ringcadence/ for Bulgaria ........ ................ 2009-01-10 01:44 +0000 [r168336] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 168334 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r168334 | tilghman | 2009-01-09 19:42:45 -0600 (Fri, 09 Jan 2009) | 2 lines sizeof for a stringfield is 4. Kinda low for reconstructing a field value. ........ 2009-01-09 23:18 +0000 [r168272] Kevin P. Fleming * sounds/Makefile, /: Merged revisions 168270 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168270 | kpfleming | 2009-01-09 17:16:08 -0600 (Fri, 09 Jan 2009) | 9 lines Merged revisions 168267 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168267 | kpfleming | 2009-01-09 17:12:29 -0600 (Fri, 09 Jan 2009) | 1 line update to use new sound file packages that include license files ........ ................ 2009-01-09 23:12 +0000 [r168266] Richard Mudgett * channels/chan_misdn.c, /: Merged revisions 168192 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168192 | rmudgett | 2009-01-09 15:43:30 -0600 (Fri, 09 Jan 2009) | 10 lines Merged revisions 168191 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168191 | rmudgett | 2009-01-09 15:28:42 -0600 (Fri, 09 Jan 2009) | 3 lines * Fix for JIRA AST-175/ABE-1757 * Miscellaneous doxygen comments added. ........ ................ 2009-01-09 22:23 +0000 [r168209] Russell Bryant * /, res/res_musiconhold.c: Merged revisions 168200 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168200 | russell | 2009-01-09 16:21:05 -0600 (Fri, 09 Jan 2009) | 10 lines Merged revisions 168198 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168198 | russell | 2009-01-09 16:14:38 -0600 (Fri, 09 Jan 2009) | 2 lines Make this compile for mvanbaak ........ ................ 2009-01-09 21:57 +0000 [r168196] Mark Michelson * /, channels/chan_sip.c: Merged revisions 168193 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168193 | mmichelson | 2009-01-09 15:53:26 -0600 (Fri, 09 Jan 2009) | 21 lines Merged revisions 168128 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168128 | mmichelson | 2009-01-09 14:08:04 -0600 (Fri, 09 Jan 2009) | 13 lines Add check_via calls to more request handlers INFO, NOTIFY, OPTIONS, REFER, and MESSAGE requests were not checking the topmost Via to determine where to send the response. Adding check_via calls to those request handlers solves this. (closes issue #13071) Reported by: baron Patches: check_via.patch uploaded by baron (license 531) Tested by: baron ........ ................ 2009-01-09 20:30 +0000 [r168157] Terry Wilson * /, res/res_phoneprov.c: Merged revisions 168142 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r168142 | twilson | 2009-01-09 14:25:25 -0600 (Fri, 09 Jan 2009) | 7 lines Don't leak memory if phoneprov.conf does not exist (closes issue #14203) Reported by: jamesgolovich Patches: asterisk-phoneprovleak.diff.txt uploaded by jamesgolovich (license 176) ........ 2009-01-09 18:42 +0000 [r168092] Tilghman Lesher * /, res/res_agi.c: Merged revisions 168090 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r168090 | tilghman | 2009-01-09 12:30:55 -0600 (Fri, 09 Jan 2009) | 3 lines When using ast_str with a non-ast_str-enabled API, we need to update the buffer or otherwise, we cannot use ast_str_strlen(). ........ 2009-01-09 16:41 +0000 [r168015] Matthew Nicholson * /, main/logger.c: Merged revisions 168014 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r168014 | mnicholson | 2009-01-09 10:32:34 -0600 (Fri, 09 Jan 2009) | 5 lines Use ast_safe_system() in logger.c instead of system() (closes issue #14194) Reported by: pabelanger ........ 2009-01-09 00:45 +0000 [r167972] Terry Wilson * apps/app_dial.c, /: Merged revisions 167935 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r167935 | twilson | 2009-01-08 18:13:12 -0600 (Thu, 08 Jan 2009) | 2 lines Set peer context and exten values so MACRO_EXTEN and MACRO_CONTEXT will be set ........ 2009-01-08 22:45 +0000 [r167836-167905] Tilghman Lesher * /, res/res_agi.c: Merged revisions 167894 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r167894 | tilghman | 2009-01-08 16:37:20 -0600 (Thu, 08 Jan 2009) | 13 lines Merged revisions 167840 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167840 | tilghman | 2009-01-08 16:08:56 -0600 (Thu, 08 Jan 2009) | 6 lines Don't truncate database results at 255 chars. (closes issue #14069) Reported by: evandro Patches: 20081214__bug14069.diff.txt uploaded by Corydon76 (license 14) ........ ................ * /, apps/app_minivm.c: Merged revisions 167835 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r167835 | tilghman | 2009-01-08 15:32:45 -0600 (Thu, 08 Jan 2009) | 6 lines Textual changes, consistency in status variable naming, and other minor bugs. (closes issue #13943) Reported by: Marquis Patches: minivm_trunk_fixes3.patch uploaded by Marquis (license 32) ........ 2009-01-08 17:28 +0000 [r167701-167727] Kevin P. Fleming * /, channels/chan_sip.c: Merged revisions 167720 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r167720 | kpfleming | 2009-01-08 11:26:03 -0600 (Thu, 08 Jan 2009) | 9 lines Merged revisions 167714 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167714 | kpfleming | 2009-01-08 11:24:21 -0600 (Thu, 08 Jan 2009) | 1 line remove an unnecessary argument to queue_request() ........ ................ * /, channels/chan_sip.c: Merged revisions 167700 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r167700 | kpfleming | 2009-01-08 10:43:26 -0600 (Thu, 08 Jan 2009) | 12 lines Merged revisions 167620 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167620 | kpfleming | 2009-01-07 17:32:21 -0600 (Wed, 07 Jan 2009) | 5 lines When a SIP request or response arrives for a dialog with an associated Asterisk channel, and the lock on that channel cannot be obtained because it is held by another thread, instead of dropping the request/response, queue it for later processing when the channel lock becomes available. http://reviewboard.digium.com/r/123/ ........ ................ 2009-01-08 14:30 +0000 [r167663] Leif Madsen * contrib/scripts/sip-friends.sql, /: Merged revisions 167662 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r167662 | lmadsen | 2009-01-08 09:27:53 -0500 (Thu, 08 Jan 2009) | 1 line Oops... fix the fieldname I changed yesterday to be right. ........ 2009-01-07 22:37 +0000 [r167544-167573] Russell Bryant * /, main/file.c: Merged revisions 167569 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r167569 | russell | 2009-01-07 16:36:34 -0600 (Wed, 07 Jan 2009) | 10 lines Merged revisions 167566 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167566 | russell | 2009-01-07 16:35:36 -0600 (Wed, 07 Jan 2009) | 2 lines Fix the last couple of places where free() was improperly used directly. ........ ................ * /, main/file.c: Merged revisions 167555 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r167555 | russell | 2009-01-07 16:27:23 -0600 (Wed, 07 Jan 2009) | 10 lines Merged revisions 167554 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167554 | russell | 2009-01-07 16:26:42 -0600 (Wed, 07 Jan 2009) | 2 lines Don't fclose() the file early, the filestream destructor will handle it. ........ ................ * /, main/file.c: Merged revisions 167546 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r167546 | russell | 2009-01-07 16:20:31 -0600 (Wed, 07 Jan 2009) | 10 lines Merged revisions 167545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167545 | russell | 2009-01-07 16:19:47 -0600 (Wed, 07 Jan 2009) | 2 lines Only try to close the file if one was actually opened ........ ................ * /, main/file.c: Merged revisions 167542 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r167542 | russell | 2009-01-07 16:05:29 -0600 (Wed, 07 Jan 2009) | 12 lines Merged revisions 167541 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167541 | russell | 2009-01-07 16:03:59 -0600 (Wed, 07 Jan 2009) | 4 lines Don't use free() directly. This caused a crash since ast_filestream is now an ao2 object. Reported by JunK-Y on IRC, #asterisk-dev ........ ................ 2009-01-07 18:32 +0000 [r167502] BJ Weschke * apps/app_followme.c, /: Merged revisions 167478 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r167478 | bweschke | 2009-01-07 13:20:31 -0500 (Wed, 07 Jan 2009) | 7 lines Answer the channel if it has not already been answered and we've already found a valid profile for followme. (closes issue #14140) Reported by: dimas Patches: 14140.patch uploaded by dimas ........ 2009-01-07 18:27 +0000 [r167491] Leif Madsen * /, configs/queues.conf.sample: Merged revisions 167477 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r167477 | lmadsen | 2009-01-07 13:18:45 -0500 (Wed, 07 Jan 2009) | 8 lines Update queues.conf.sample documentation. Update the queues.conf.sample documentation to mention that you need to preload chan_local.so as well if you plan on using Local channels for queue members, and you're preloading pbx_config.so. (closes issue #14179) Reported by: CrashHD Tested by: CrashHD ........ 2009-01-07 17:46 +0000 [r167456] Russell Bryant * main/indications.c, /: Merged revisions 167442 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r167442 | russell | 2009-01-07 11:35:39 -0600 (Wed, 07 Jan 2009) | 12 lines Merged revisions 167432 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167432 | russell | 2009-01-07 11:29:53 -0600 (Wed, 07 Jan 2009) | 4 lines Treat an empty string the same way as a NULL country argument. In passing, simplify the handling of returning a default tone zone. ........ ................ 2009-01-07 14:41 +0000 [r167376] Leif Madsen * contrib/scripts/sip-friends.sql, /: Merged revisions 167373 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r167373 | lmadsen | 2009-01-07 09:26:19 -0500 (Wed, 07 Jan 2009) | 1 line Update the sip-friends.sql file to use the non-deprecated 'defaultname' instead of 'username' and remove an extra comma that would cause the script to fail as-is ........ 2009-01-06 21:38 +0000 [r167306] Mark Michelson * main/db.c, /: Merged revisions 167301 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r167301 | mmichelson | 2009-01-06 15:36:44 -0600 (Tue, 06 Jan 2009) | 16 lines Merged revisions 167299 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167299 | mmichelson | 2009-01-06 15:35:57 -0600 (Tue, 06 Jan 2009) | 8 lines Use the correct variable when creating the format string (closes issue #14177) Reported by: nic_bellamy Patches: asterisk-trunk-svn-r167242-ast_db_gettree.patch uploaded by nic (license 299) ........ ................ 2009-01-06 21:10 +0000 [r167268] Tilghman Lesher * channels/chan_iax2.c, /: Merged revisions 167265 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r167265 | tilghman | 2009-01-06 15:02:33 -0600 (Tue, 06 Jan 2009) | 16 lines Merged revisions 167260 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r167260 | tilghman | 2009-01-06 14:48:05 -0600 (Tue, 06 Jan 2009) | 9 lines Merged revisions 167259 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r167259 | tilghman | 2009-01-06 14:44:03 -0600 (Tue, 06 Jan 2009) | 2 lines Security fix AST-2009-001. ........ ................ ................ 2009-01-05 17:10 +0000 [r167182] Mark Michelson * /, channels/chan_sip.c: Merged revisions 167180 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r167180 | mmichelson | 2009-01-05 10:59:36 -0600 (Mon, 05 Jan 2009) | 49 lines Merged revisions 167179 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167179 | mmichelson | 2009-01-05 10:51:59 -0600 (Mon, 05 Jan 2009) | 41 lines A couple of changes to T.38 SDP attribute handling There are some boolean attributes for T.38 such as T38FaxFillBitRemoval, T38FaxTranscodingMMR, and T38FaxTranscodingJBIG. By simply being present, we should treat these as a "true" value. The current code, however, was requiring a 1 or 0 as the value of the attribute in order to parse it. This is due to the fact that there are some T.38 endpoints and gateways that also transmit this information incorrectly. This patch follows the "be liberal in what you accept and strict in what you send" philosophy by accepting both the correctly- and incorrectly-formatted attributes, but only sending information as it is supposed to be sent. It was also discovered that a particular type of T.38 gateway sends some non-standard T.38 SDP attributes. Instead of using T38FaxMaxDatagram and T38MaxBitRate, it used T38MaxDatagram and T38FaxMaxRate respectively. We now will properly accept these attributes as well. Note that there are a lot of patches cited in the below commit message template. This is because the person who submitted these patches is an awesome person and wrote 1.4, 1.6.0, and 1.6.1 variants. (closes issue #13976) Reported by: linulin Patches: chan_sip.c.1.4-update1.diff uploaded by arcivanov (license 648) chan_sip.c.1.6.0-update1.diff uploaded by arcivanov (license 648) chan_sip.c.1.6.1-update1.diff uploaded by arcivanov (license 648) chan_sip.c.1.4-relaxedT38_update1.diff uploaded by arcivanov (license 648) chan_sip.c.1.6.0-relaxedT38_update1.diff uploaded by arcivanov (license 648) chan_sip.c.1.6.1-relaxedT38_update1.diff uploaded by arcivanov (license 648) Tested by: arcivanov ........ ................ 2009-01-05 16:46 +0000 [r167178] Tilghman Lesher * /, UPGRADE-1.6.txt: Merged revisions 167176 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r167176 | tilghman | 2009-01-05 10:44:47 -0600 (Mon, 05 Jan 2009) | 7 lines More clearly explain that quote marks are no longer necessary. (closes issue #13718) Reported by: davidw Patches: 20081020__bug13718.diff.txt uploaded by Corydon76 (license 14) Tested by: blitzrage ........ 2008-12-31 19:38 +0000 [r166957] Tilghman Lesher * channels/chan_local.c, /: Merged revisions 166954 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r166954 | tilghman | 2008-12-31 13:34:28 -0600 (Wed, 31 Dec 2008) | 12 lines Merged revisions 166953 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166953 | tilghman | 2008-12-31 13:20:35 -0600 (Wed, 31 Dec 2008) | 5 lines Also inherit the musiconhold class. (Closes #14153) Reported by: Jerry Geis, via the users list. Patch by: me (license 14) ........ ................ 2008-12-30 20:57 +0000 [r166910] Terry Wilson * phoneprov/polycom_line.xml, doc/realtimetext.txt, /, res/res_phoneprov.c, doc/sip-retransmit.txt, doc/tex/phoneprov.tex, res/res_http_post.c: Merged revisions 166908 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r166908 | twilson | 2008-12-30 14:50:05 -0600 (Tue, 30 Dec 2008) | 2 lines Fix some svn:keywords ........ 2008-12-29 18:16 +0000 [r166863] Mark Michelson * apps/app_queue.c, apps/app_dial.c, /: Merged revisions 166861 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r166861 | mmichelson | 2008-12-29 12:04:52 -0600 (Mon, 29 Dec 2008) | 14 lines Update app_queue to deal with the removal of AST_PBX_KEEPALIVE When placing a call to a queue which ran a gosub on the member's channel, Asterisk would crash every time, stemming from the fact that the member's channel was being hung up unexpectedly when the Gosub completed. The necessary change was pretty much copied and pasted from app_dial's similar changes made last week. I also took the opportunity to change a LOG_DEBUG message in app_dial to use ast_debug. I am guessing this was due to a direct merge from 1.4 that was not corrected to use trunk's preferred syntax. ........ 2008-12-29 14:52 +0000 [r166858] Joshua Colp * channels/chan_sip.c: Per kpfleming add a note describing why you must never change the first element of peer_finding_info. 2008-12-28 15:16 +0000 [r166775] Russell Bryant * channels/misdn_config.c, /: Merged revisions 166773 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r166773 | russell | 2008-12-28 09:15:14 -0600 (Sun, 28 Dec 2008) | 12 lines Merged revisions 166772 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166772 | russell | 2008-12-28 09:13:48 -0600 (Sun, 28 Dec 2008) | 4 lines Use strncat() instead of an sprintf() in which source and target buffers overlap http://lists.digium.com/pipermail/asterisk-dev/2008-December/035919.html ........ ................ 2008-12-24 01:15 +0000 [r166730] Steve Murphy * apps/app_queue.c, include/asterisk/features.h, apps/app_dial.c, main/pbx.c, /, main/features.c, apps/app_macro.c, include/asterisk/pbx.h: Merged revisions 166665 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk This merged from trunk with no conflicts. I tested mostly the 'tired' cases, and for the most part ignored the tests for reconnecting and dialing in to fetch a parked call, after the first case. ................ r166665 | murf | 2008-12-23 11:13:49 -0700 (Tue, 23 Dec 2008) | 153 lines Merged revisions 166093 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 In order to merge this 1.4 patch into trunk, I had to resolve some conflicts and wait for Russell to make some changes to res_agi. I re-ran all the tests; 39 calls in all, and made fairly careful notes and comparisons: I don't want this to blow up some aspect of asterisk; I completely removed the KEEPALIVE from the pbx.h decls. The first 3 scenarios involving feature park; feature xfer to 700; hookflash park to Park() app call all behave the same, don't appear to leave hung channels, and no crashes. ........ r166093 | murf | 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) | 131 lines This merges the masqpark branch into 1.4 These changes eliminate the need for (and use of) the KEEPALIVE return code in res_features.c; There are other places that use this result code for similar purposes at a higher level, these appear to be left alone in 1.4, but attacked in trunk. The reason these changes are being made in 1.4, is that parking ends a channel's life, in some situations, and the code in the bridge (and some other places), was not checking the result code properly, and dereferencing the channel pointer, which could lead to memory corruption and crashes. Calling the masq_park function eliminates this danger in higher levels. A series of previous commits have replaced some parking calls with masq_park, but this patch puts them ALL to rest, (except one, purposely left alone because a masquerade is done anyway), and gets rid of the code that tests the KEEPALIVE result, and the NOHANGUP_PEER result codes. While bug 13820 inspired this work, this patch does not solve all the problems mentioned there. I have tested this patch (again) to make sure I have not introduced regressions. Crashes that occurred when a parked party hung up while the parking party was listening to the numbers of the parking stall being assigned, is eliminated. These are the cases where parking code may be activated: 1. Feature one touch (eg. *3) 2. Feature blind xfer to parking lot (eg ##700) 3. Run Park() app from dialplan (eg sip xfer to 700) (eg. dahdi hookflash xfer to 700) 4. Run Park via manager. The interesting testing cases for parking are: I. A calls B, A parks B a. B hangs up while A is getting the numbers announced. b. B hangs up after A gets the announcement, but before the parking time expires c. B waits, time expires, A is redialed, A answers, B and A are connected, after which, B hangs up. d. C picks up B while still in parking lot. II. A calls B, B parks A a. A hangs up while B is getting the numbers announced. b. A hangs up after B gets the announcement, but before the parking time expires c. A waits, time expires, B is redialed, B answers, A and B are connected, after which, A hangs up. d. C picks up A while still in parking lot. Testing this throroughly involves acting all the permutations of I and II, in situations 1,2,3, and 4. Since I added a few more changes (ALL references to KEEPALIVE in the bridge code eliimated (I missed one earlier), I retested most of the above cases, and no crashes. H-extension weirdness. Current h-extension execution is not completely correct for several of the cases. For the case where A calls B, and A parks B, the 'h' exten is run on A's channel as soon as the park is accomplished. This is expected behavior. But when A calls B, and B parks A, this will be current behavior: After B parks A, B is hung up by the system, and the 'h' (hangup) exten gets run, but the channel mentioned will be a derivative of A's... Thus, if A is DAHDI/1, and B is DAHDI/2, the h-extension will be run on channel Parked/DAHDI/1-1, and the start/answer/end info will be those relating to Channel A. And, in the case where A is reconnected to B after the park time expires, when both parties hang up after the joyful reunion, no h-exten will be run at all. In the case where C picks up A from the parking lot, when either A or C hang up, the h-exten will be run for the C channel. CDR's are a separate issue, and not addressed here. As to WHY this strange behavior occurs, the answer lies in the procedure followed to accomplish handing over the channel to the parking manager thread. This procedure is called masquerading. In the process, a duplicate copy of the channel is created, and most of the active data is given to the new copy. The original channel gets its name changed to XXX and keeps the PBX information for the sake of the original thread (preserving its role as a call originator, if it had this role to begin with), while the new channel is without this info and becomes a call target (a "peer"). In this case, the parking lot manager thread is handed the new (masqueraded) channel. It will not run an h-exten on the channel if it hangs up while in the parking lot. The h exten will be run on the original channel instead, in the original thread, after the bridge completes. See bug 13820 for our intentions as to how to clean up the h exten behavior. Review: http://reviewboard.digium.com/r/29/ ........ ................ 2008-12-23 20:56 +0000 [r166698] Tilghman Lesher * include/asterisk/app.h, /, channels/chan_sip.c, main/app.c: Merged revisions 166696 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r166696 | tilghman | 2008-12-23 14:47:08 -0600 (Tue, 23 Dec 2008) | 7 lines Allow semicolons and extended characters in user-specified SIP headers. (closes issue #14110) Reported by: gork Patches: 20081222__bug14110__2.diff.txt uploaded by Corydon76 (license 14) Tested by: gork, putnopvut ........ 2008-12-23 15:20 +0000 [r166571] Mark Michelson * main/channel.c, /: Merged revisions 166569 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r166569 | mmichelson | 2008-12-23 09:17:54 -0600 (Tue, 23 Dec 2008) | 20 lines Merged revisions 166568 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166568 | mmichelson | 2008-12-23 09:16:26 -0600 (Tue, 23 Dec 2008) | 12 lines Fix a crash resulting from a datastore with inheritance but no duplicate callback The fix for this is to simply set the newly created datastore's data pointer to NULL if it is inherited but has no duplicate callback. (closes issue #14113) Reported by: francesco_r Patches: 14113.patch uploaded by putnopvut (license 60) Tested by: francesco_r ........ ................ 2008-12-23 04:34 +0000 [r166535] Tilghman Lesher * main/channel.c, /: Merged revisions 166533 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r166533 | tilghman | 2008-12-22 22:32:15 -0600 (Mon, 22 Dec 2008) | 11 lines Merged revisions 166509 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166509 | tilghman | 2008-12-22 22:05:25 -0600 (Mon, 22 Dec 2008) | 4 lines Use the integer form of condition for integer comparisons. (closes issue #14127) Reported by: andrew ........ ................ 2008-12-22 23:27 +0000 [r166440-166472] Mark Michelson * /, res/res_agi.c: Merged revisions 166470 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r166470 | mmichelson | 2008-12-22 17:25:34 -0600 (Mon, 22 Dec 2008) | 11 lines Always use the value of the AGISIGHUP when running an AGI. Prior to this patch, the value of AGISIGUP was not always honored when set on a channel. (closes issue #13711) Reported by: fmueller Patches: 13711.patch uploaded by putnopvut (license 60) ........ * channels/chan_dahdi.c, /: Merged revisions 166382 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r166382 | mmichelson | 2008-12-22 15:08:03 -0600 (Mon, 22 Dec 2008) | 44 lines Merged revisions 166380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166380 | mmichelson | 2008-12-22 14:56:29 -0600 (Mon, 22 Dec 2008) | 36 lines Fix a deadlock relating to channel locks and autoservice It has been discovered that if a channel is locked prior to a call to ast_autoservice_stop, then it is likely that a deadlock will occur. The reason is that the call to ast_autoservice_stop has a check built into it to be sure that the thread running autoservice is not currently trying to manipulate the channel we are about to pull out of autoservice. The autoservice thread, however, cannot advance beyond where it currently is, though, because it is trying to acquire the lock of the channel for which autoservice is attempting to be stopped. The gist of all this is that a channel MUST NOT be locked when attempting to stop autoservice on the channel. In this particular case, the channel was locked by a call to ast_read. A call to ast_exists_extension led to autoservice being started and stopped due to the existence of dialplan switches. It may be that there are future commits which handle the same symptoms but in a different location, but based on my looks through the code, it is very rare to see a construct such as this one. (closes issue #14057) Reported by: rtrauntvein Patches: 14057v3.patch uploaded by putnopvut (license 60) Tested by: rtrauntvein Review: http://reviewboard.digium.com/r/107/ ........ ................ 2008-12-22 21:46 +0000 [r166277-166438] Russell Bryant * /, res/res_musiconhold.c: Merged revisions 166436 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r166436 | russell | 2008-12-22 15:45:28 -0600 (Mon, 22 Dec 2008) | 2 lines Cosmetic change - don't mix struct initializer styles. ........ * /, res/res_musiconhold.c: Merged revisions 166377 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r166377 | russell | 2008-12-22 14:26:48 -0600 (Mon, 22 Dec 2008) | 2 lines Fix a bad typo. ........ * main/astobj2.c, /: Merged revisions 166342 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r166342 | russell | 2008-12-22 11:44:23 -0600 (Mon, 22 Dec 2008) | 2 lines Remove some error messages. This is the default handler that is valid to use. ........ * /, main/utils.c: Merged revisions 166317 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r166317 | russell | 2008-12-22 11:29:10 -0600 (Mon, 22 Dec 2008) | 10 lines Merged revisions 166297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166297 | russell | 2008-12-22 11:22:56 -0600 (Mon, 22 Dec 2008) | 2 lines Fix up timeout handling in ast_carefulwrite(). ........ ................ * include/asterisk/utils.h, main/manager.c, /, main/utils.c: Merged revisions 166282 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r166282 | russell | 2008-12-22 11:09:36 -0600 (Mon, 22 Dec 2008) | 12 lines Introduce ast_careful_fwrite() and use in AMI to prevent partial writes. This patch introduces a function to do careful writes on a file stream which will handle timeouts and partial writes. It is currently used in AMI to address the issue that has been reported. However, there are probably a few other places where this could be used. (closes issue #13546) Reported by: srt Tested by: russell http://reviewboard.digium.com/r/104/ ........ * /, res/res_musiconhold.c: Merged revisions 166273 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r166273 | russell | 2008-12-22 10:10:40 -0600 (Mon, 22 Dec 2008) | 7 lines Re-work ref count handling of MoH classes using astobj2 to resolve crashes. (closes issue #13566) Reported by: igorcarneiro Tested by: russell Review: http://reviewboard.digium.com/r/106/ ........ 2008-12-22 16:17 +0000 [r166275] Mark Michelson * /, funcs/func_timeout.c, main/file.c: Merged revisions 166267 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r166267 | mmichelson | 2008-12-22 10:07:59 -0600 (Mon, 22 Dec 2008) | 17 lines Fix a file playback crash and explicitly initialize values in func_timeout.c A crash was brought up on the bugtracker. The first run through valgrind was full of legitimate complaints of uninitialized values in func_timeout when setting a response timeout. These were fixed but the crash persisted. A second run through showed the real problem. The reference counting used for filestreams was incorrect because there were some missing increments when a frame was read from a format module. (closes issue #14118) Reported by: blitzrage Patches: 14118v2.patch uploaded by putnopvut (license 60) Tested by: blitzrage ........ 2008-12-22 16:10 +0000 [r166272] Joshua Colp * main/dnsmgr.c, /: Merged revisions 166268 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r166268 | file | 2008-12-22 12:08:13 -0400 (Mon, 22 Dec 2008) | 7 lines Record the previous port in the temporary address structure so that the comparison does not treat the host as having changed even if it did not. This would have been uninitialized before and would have led to a baddddd port. (closes issue #13628) Reported by: pananix Patches: bug13628.patch uploaded by jpeeler (license 325) Tested by: file, blitzrage ........ 2008-12-22 14:19 +0000 [r166260] Russell Bryant * /, res/res_agi.c: Merged revisions 166258 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r166258 | russell | 2008-12-22 08:16:54 -0600 (Mon, 22 Dec 2008) | 26 lines Remove AST_PBX_KEEPALIVE usage from res_agi. This patch removes the usage of AST_PBX_KEEPALIVE from res_agi. The only usage was for the AGI command, "asyncagi break". This patch removes this feature. Normally, a feature would not be removed like this. However, this code is broken and usage of it will result in a memory leak. Usage of this feature will make the AGI code return a result of AST_PBX_KEEPALIVE. The PBX handler assumes that another thread has assumed ownership of the channel. The channel thread will exit without destroying the channel. Unfortunately, _no_ thread has ownership of the channel at this point. There are a couple of serious problems here: 1) The only way to recover the caller is to issue a channel redirect. This will work, but this will be done with a masquerade, and the old ast_channel structure will be lost. 2) Until the channel redirect happens, there is no code servicing the channel. That means nothing is reading audio or handling events coming from the channel. This is very bad. The recommended way to get this same "break" functionality is to issue the redirect while the channel is still being handled by the AGI code. That way, there will be no memory leak, and there will be no period of time that the channel is not being serviced. ........ 2008-12-19 23:45 +0000 [r166098-166164] Mark Michelson * /, main/audiohook.c: Merged revisions 166162 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r166162 | mmichelson | 2008-12-19 17:45:00 -0600 (Fri, 19 Dec 2008) | 6 lines Get rid of an extra space. I don't know how this crept back in when I had already fixed it earlier ........ * funcs/func_audiohookinherit.c: Switch documentation formats for func_audiohookinherit.c 1.6.1 does not have xml documentation, so I reverted to the old way here. * main/channel.c, funcs/func_audiohookinherit.c (added), /, include/asterisk/audiohook.h, main/audiohook.c: Merged revisions 166092,166095 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r166092 | mmichelson | 2008-12-19 16:26:16 -0600 (Fri, 19 Dec 2008) | 28 lines Adding a new dialplan function AUDIOHOOK_INHERIT This function is being added as a method to allow for an audiohook to move to a new channel during a channel masquerade. The most obvious use for such a facility is for MixMonitor when a transfer is performed. Prior to the addition of this functionality, if a channel running MixMonitor was transferred by another party, then the recording would stop once the transfer had completed. By using AUDIOHOOK_INHERIT, you can make MixMonitor continue recording the call even after the transfer has completed. It has also been determined that since this is seen by most as a bug fix and is not an invasive change, this functionality will also be backported to 1.4 and merged into the 1.6.0 branches, even though they are feature-frozen. (closes issue #13538) Reported by: mbit Patches: 13538.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/102/ ........ r166095 | mmichelson | 2008-12-19 16:40:57 -0600 (Fri, 19 Dec 2008) | 5 lines Remove the verbatim tag from the author line I could have sworn I already did that before, though... ........ 2008-12-19 15:08 +0000 [r165892] Russell Bryant * apps/app_chanspy.c, /: Merged revisions 165890 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r165890 | russell | 2008-12-19 09:05:09 -0600 (Fri, 19 Dec 2008) | 17 lines Merged revisions 165889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165889 | russell | 2008-12-19 09:03:02 -0600 (Fri, 19 Dec 2008) | 9 lines Ensure that the chanspy datastore is fully initialized. This patch resolved some random crash issues observed by a user on a BSD system (closes issue #14111) Reported by: ys Patches: app_chanspy.c.diff uploaded by ys (license 281) ........ ................ 2008-12-18 Leif Madsen * Asterisk 1.6.1-beta4 released. 2008-12-18 21:57 +0000 [r165808] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 165797 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r165797 | tilghman | 2008-12-18 15:41:02 -0600 (Thu, 18 Dec 2008) | 15 lines Merged revisions 165767 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165767 | tilghman | 2008-12-18 15:14:47 -0600 (Thu, 18 Dec 2008) | 8 lines Add mutexes around accesses to the IMAP library interface. This prevents certain crashes, especially when shared mailboxes are used. (closes issue #13653) Reported by: howardwilkinson Patches: asterisk-1.4.21.2-appvoicemail-sharedimap-lock.patch uploaded by howardwilkinson (license 590) Tested by: jpeeler ........ ................ 2008-12-18 21:47 +0000 [r165804] Russell Bryant * /, main/utils.c: Merged revisions 165801 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r165801 | russell | 2008-12-18 15:44:47 -0600 (Thu, 18 Dec 2008) | 19 lines Merged revisions 165796 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165796 | russell | 2008-12-18 15:39:25 -0600 (Thu, 18 Dec 2008) | 11 lines Make ast_carefulwrite() be more careful. This patch handles some additional cases that could result in partial writes to the file description. This was done to address complaints about partial writes on AMI. (issue #13546) (more changes needed to address potential problems in 1.6) Reported by: srt Tested by: russell Review: http://reviewboard.digium.com/r/99/ ........ ................ 2008-12-18 21:24 +0000 [r165794] Joshua Colp * apps/app_queue.c, channels/chan_oss.c, channels/chan_dahdi.c, channels/chan_misdn.c, /, channels/chan_sip.c, pbx/pbx_ael.c: Merged revisions 165792 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r165792 | file | 2008-12-18 17:21:44 -0400 (Thu, 18 Dec 2008) | 6 lines Numerous documentation updates. (closes issue #13970) Reported by: pkempgen Patches: __20081217_cli_usage_fixes.patch.txt uploaded by blitzrage (license 10) ........ 2008-12-18 19:45 +0000 [r165728] Russell Bryant * apps/app_dial.c, main/pbx.c, /, include/asterisk/pbx.h: Merged revisions 165723 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r165723 | russell | 2008-12-18 13:33:42 -0600 (Thu, 18 Dec 2008) | 14 lines Remove the need for AST_PBX_KEEPALIVE with the GoSub option from Dial. This is part of an effort to completely remove AST_PBX_KEEPALIVE and other similar return codes from the source. While this usage was perfectly safe, there are others that are problematic. Since we know ahead of time that we do not want to PBX to destroy the channel, the PBX API has been changed so that information can be provided as an argument, instead, thus removing the need for the KEEPALIVE return value. Further changes to get rid of KEEPALIVE and related code is being done by murf. There is a patch up for that on review 29. Review: http://reviewboard.digium.com/r/98/ ........ 2008-12-18 19:36 +0000 [r165725] Mark Michelson * res/res_odbc.c, /: Merged revisions 165724 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r165724 | mmichelson | 2008-12-18 13:34:33 -0600 (Thu, 18 Dec 2008) | 8 lines Fix crashes in res_odbc. The variable "class" was being set NULL just prior to being dereferenced in an ao2_link call. I have moved the setting of the variable to NULL until after the ao2_link call. ........ 2008-12-18 18:58 +0000 [r165664] Russell Bryant * /, res/res_musiconhold.c: Merged revisions 165662 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r165662 | russell | 2008-12-18 12:54:47 -0600 (Thu, 18 Dec 2008) | 15 lines Merged revisions 165661 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165661 | russell | 2008-12-18 12:52:18 -0600 (Thu, 18 Dec 2008) | 7 lines Set the process group ID on the MOH process so that all children will get killed (closes issue #14099) Reported by: caspy Patches: res_musiconhold.c.patch.killpg.try2 uploaded by caspy (license 645) ........ ................ 2008-12-18 18:47 +0000 [r165660] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 165658 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r165658 | tilghman | 2008-12-18 12:36:48 -0600 (Thu, 18 Dec 2008) | 2 lines Fix 2 resource leaks and fix another pipe-to-comma conversion ........ 2008-12-18 17:59 +0000 [r165605-165606] Joshua Colp * /, channels/chan_sip.c: Merge in changes to return chan_sip to matching based on how it was previously done and how it is done in trunk. It will do name based for users and friends and IP based for peers. (closes issue #14107) Reported by: jsmith * main/rtp.c, /: Merged revisions 165599 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r165599 | file | 2008-12-18 13:13:32 -0400 (Thu, 18 Dec 2008) | 11 lines Merged revisions 165591 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165591 | file | 2008-12-18 13:11:42 -0400 (Thu, 18 Dec 2008) | 4 lines Only care about a compatible codec for early bridging if we are actually bridging to another channel. If we are not we actually want to bring the audio back to us. (closes issue #13545) Reported by: davidw ........ ................ 2008-12-18 16:48 +0000 [r165543] Tilghman Lesher * res/res_odbc.c, /: Merged revisions 165541 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r165541 | tilghman | 2008-12-18 10:36:48 -0600 (Thu, 18 Dec 2008) | 2 lines Fix reference counts of the class and add an assertion to the end. ........ 2008-12-17 21:48 +0000 [r165332] Mark Michelson * res/res_odbc.c, /: Merged revisions 165330 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r165330 | mmichelson | 2008-12-17 15:46:19 -0600 (Wed, 17 Dec 2008) | 3 lines Fix a refcount leak in res_odbc ........ 2008-12-17 21:31 +0000 [r165329] Tilghman Lesher * /, apps/app_macro.c: Merged revisions 165325 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r165325 | tilghman | 2008-12-17 15:28:51 -0600 (Wed, 17 Dec 2008) | 2 lines Oops, broke trunk ........ 2008-12-17 21:25 +0000 [r165324] Mark Michelson * apps/app_directory.c, apps/app_queue.c, apps/app_voicemail.c, /, res/res_realtime.c: Merged revisions 165318 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r165318 | mmichelson | 2008-12-17 15:17:20 -0600 (Wed, 17 Dec 2008) | 15 lines Merged revisions 165255 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165255 | mmichelson | 2008-12-17 14:51:38 -0600 (Wed, 17 Dec 2008) | 7 lines Fix some memory leaks found while looking at how realtime configs are handled. Also cleaned up some coding guidelines violations in app_realtime.c, mostly related to spacing ........ ................ 2008-12-17 21:22 +0000 [r165323] Tilghman Lesher * /, apps/app_macro.c: Merged revisions 165319 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r165319 | tilghman | 2008-12-17 15:18:57 -0600 (Wed, 17 Dec 2008) | 11 lines Merged revisions 165317 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165317 | tilghman | 2008-12-17 15:14:37 -0600 (Wed, 17 Dec 2008) | 4 lines Reverse the fix from issue #6176 and add proper handling for that issue. (Closes issue #13962, closes issue #13363) Fixed by myself (license 14) ........ ................ 2008-12-17 21:02 +0000 [r165279] Steve Murphy * /, utils/extconf.c: Merged revisions 165254 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r165254 | murf | 2008-12-17 13:50:19 -0700 (Wed, 17 Dec 2008) | 5 lines This patch is here committed to satisfy the buildbot, who has a problem with the const. ........ 2008-12-17 20:02 +0000 [r165242] Terry Wilson * /, res/res_phoneprov.c: Merged revisions 165219 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r165219 | twilson | 2008-12-17 13:55:10 -0600 (Wed, 17 Dec 2008) | 2 lines Polycom phones close the connection after reading a little bit of the firmware files, we should stop sending in that case. Also, make that case print out a debug statement instead of a scary WARNING. ........ 2008-12-17 19:54 +0000 [r165218] Joshua Colp * /, channels/chan_sip.c: Merged revisions 165216 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r165216 | file | 2008-12-17 15:52:40 -0400 (Wed, 17 Dec 2008) | 4 lines Call proxy_update so that the IP address gets populated. Sending stuff to 0.0.0.0 is silly! (closes issue #14055) Reported by: chris-mac ........ 2008-12-17 17:56 +0000 [r165146] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 165142-165143 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r165142 | mmichelson | 2008-12-17 11:52:50 -0600 (Wed, 17 Dec 2008) | 10 lines Use the create_vm_state_from_user function in a place where it was not being used before. Also, I've moved the urgent folder check in messagecount() up a bit so that the flow is a bit better. This was something I noticed while taking a look at issue #13973, although I don't think this is the underlying cause of the issue. ........ r165143 | mmichelson | 2008-12-17 11:53:37 -0600 (Wed, 17 Dec 2008) | 3 lines And actually assign the function to a pointer... ........ 2008-12-17 05:53 +0000 [r165093] Steve Murphy * utils/conf2ael.c, pbx/ael/ael-test/ref.ael-vtest13, utils/check_expr.c, utils/Makefile, pbx/ael/ael-test/ref.ael-vtest17, /, pbx/pbx_ael.c, utils/ael_main.c, utils/extconf.c: Merged revisions 165071 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk I might add here that in I tested the merged fixes from trunk in both 1.6.0 and 1.6.1 via both 'make' and ./runtests in the ael regression tests for all but DEBUG_CHANNEL_LOCKS, DEBUG_SCHEDULER, and CHANNEL_TRACE options. ........ r165071 | murf | 2008-12-16 22:04:56 -0700 (Tue, 16 Dec 2008) | 31 lines A possibly "horrible fix" for a "horribly broken" situation. As stuff shifts around in the asterisk code, the miscellaneous inclusions from the standalone stuff gets broken. There's no easy fix for this situation. I made sure that everything in utils builds without problem ***AND*** that aelparse runs the regressions correctly with the following make menuselect options both on and off: DONT_OPTIMIZE DEBUG_THREADS DEBUG_CHANNEL_LOCKS MALLOC_DEBUG MTX_PROFILE DEBUG_SCHEDULER DEBUG_THREADLOCALS DETECT_DEADLOCKS CHANNEL_TRACE I think from now on, I'm going to #undef all these features in the various utils native files; I guess I could do the same for the copied-in files, surrounded by STANDALONE ifdef. A standalone isn't going to care about threads, mutexes, etc. ........ 2008-12-16 23:07 +0000 [r164980] Mark Michelson * /, channels/chan_sip.c: Merged revisions 164978 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r164978 | mmichelson | 2008-12-16 17:06:04 -0600 (Tue, 16 Dec 2008) | 15 lines Merged revisions 164977 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164977 | mmichelson | 2008-12-16 17:04:27 -0600 (Tue, 16 Dec 2008) | 7 lines After looking through SIP registration code most of the day, this is one of the few things I could find that was just plain wrong. Even though it probably isn't possible for it to happen, it seems weird to have code that checks if a pointer is NULL and then immediately dereferences that pointer if it was NULL. ........ ................ 2008-12-16 22:52 +0000 [r164960] Jeff Peeler * /, apps/app_record.c: Merged revisions 164942 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r164942 | jpeeler | 2008-12-16 16:45:39 -0600 (Tue, 16 Dec 2008) | 6 lines (closes issue #13669) Reported by: pj Delete file recording if recording terminated from a hangup. ........ 2008-12-16 21:40 +0000 [r164813-164884] Russell Bryant * /, main/utils.c: Merged revisions 164882 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r164882 | russell | 2008-12-16 15:39:15 -0600 (Tue, 16 Dec 2008) | 17 lines Merged revisions 164881 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164881 | russell | 2008-12-16 15:38:29 -0600 (Tue, 16 Dec 2008) | 9 lines Fix an issue where DEBUG_THREADS may erroneously report that a thread is exiting while holding a lock. If the last lock attempt was a trylock, and it failed, it will still be in the list of locks so that it can be reported. (closes issue #13219) Reported by: pj ........ ................ * /, apps/app_macro.c: Merged revisions 164877 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r164877 | russell | 2008-12-16 15:12:49 -0600 (Tue, 16 Dec 2008) | 14 lines Merged revisions 164876 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164876 | russell | 2008-12-16 15:10:44 -0600 (Tue, 16 Dec 2008) | 6 lines Do not dereference the channel if AST_PBX_KEEPALIVE has been returned. This is a bug I noticed while looking at the code for app_macro. This return code means that another thread has assumed ownership of the channel and it can no longer be touched. (I hate this return code with a passion, by the way.) ........ ................ * main/manager.c, /: Merged revisions 164807 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r164807 | russell | 2008-12-16 14:41:51 -0600 (Tue, 16 Dec 2008) | 17 lines Merged revisions 164806 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164806 | russell | 2008-12-16 14:35:25 -0600 (Tue, 16 Dec 2008) | 9 lines Add "restart gracefully" to the AMI blacklist of CLI commands. "module unload" was already identified as a command that can not be used from the AMI. "restart gracefully" effectively unloads all modules, and will run in to the same problems. (closes issue #13894) Reported by: kernelsensei ........ ................ 2008-12-16 20:18 +0000 [r164805] Steve Murphy * main/pbx.c, /: Merged revisions 164801 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r164801 | murf | 2008-12-16 13:04:46 -0700 (Tue, 16 Dec 2008) | 36 lines (closes issue #14076) Reported by: toc Tested by: murf OK, Well this issue has had its share of flip-flopping. I found the following: 1. the code in question, in ext_cmp1 in pbx.c, would not allow two extensions that vary only by any dashes contained within them, to be defined in the same context. 2. for input dialstrings, dashes are NOT ignored. So, skipping them when sorting patterns seemed a bit silly. Thus, you might declare ext 891 in a context, but if you try dialing 8-9-1, it will NOT match 891. So, I proposed to remove the code from ext_cmp1 to skip the spaces and dashes. Just kept us from declaring 891 and 8-9-1 in the same context, forcing users to generate otherwise uselessly obfuscated dialplan code to get the same effect. Then, I tried out 1.4, and found that: 1. you can declare 891 and 8-9-1 in the same context! 2. You can't define 891, and have 8-9-1 match it! Nor can you define 8-9-1, and have 891 match it! So, it appears that my proposal simply restores the pbx to behaving as it did in 1.4. ........ 2008-12-16 19:54 +0000 [r164799] Tilghman Lesher * contrib/scripts/safe_asterisk, /: Merged revisions 164798 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r164798 | tilghman | 2008-12-16 13:54:11 -0600 (Tue, 16 Dec 2008) | 4 lines Set up umask as a possible configuration option. (closes issue #13753) Reported by: irroot ........ 2008-12-16 17:18 +0000 [r164677-164739] Russell Bryant * include/asterisk/threadstorage.h, /, main/threadstorage.c: Merged revisions 164737 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r164737 | russell | 2008-12-16 11:14:01 -0600 (Tue, 16 Dec 2008) | 22 lines Merged revisions 164736 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164736 | russell | 2008-12-16 11:06:29 -0600 (Tue, 16 Dec 2008) | 14 lines Fix memory leak and invalid reporting issues with DEBUG_THREADLOCALS. One issue was that the ast_mutex_* API was being used within the context of the thread local data destructors. We would go off and allocate more thread local data while the pthread lib was in the middle of destroying it all. This led to a memory leak. Another issue was an invalid argument being provided to the the object_add API call. (closes issue #13678) Reported by: ys Tested by: Russell ........ ................ * /, channels/chan_sip.c: Merged revisions 164675 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r164675 | russell | 2008-12-16 10:00:29 -0600 (Tue, 16 Dec 2008) | 19 lines Merged revisions 164672 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164672 | russell | 2008-12-16 09:56:37 -0600 (Tue, 16 Dec 2008) | 11 lines Fix a memory leak related to the use of the "setvar" configuration option. The problem was that these variables were being appended to the list of vars on the sip_pvt every time a re-registration or re-subscription came in. Since it's just a waste of memory to put them there unless the request was an INVITE, then the fix is to check the request type before copying the vars. (closes issue #14037) Reported by: marvinek Tested by: russell ........ ................ 2008-12-16 15:47 +0000 [r164662] Joshua Colp * /, channels/chan_sip.c: Merged revisions 164659 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r164659 | file | 2008-12-16 11:44:28 -0400 (Tue, 16 Dec 2008) | 4 lines When using externhost make sure the port gets set to the bindaddr port if one was not specified in the externhost value itself. (closes issue #13634) Reported by: performer ........ 2008-12-16 15:42 +0000 [r164658] Steve Murphy * main/pbx.c, /: Merged revisions 164648 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r164648 | murf | 2008-12-16 08:31:54 -0700 (Tue, 16 Dec 2008) | 13 lines Merged revisions 164634 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164634 | murf | 2008-12-16 08:15:58 -0700 (Tue, 16 Dec 2008) | 5 lines I added a sentence to clarify why - and ' ' are ignored in patterns as per bug 14076. Leif says he'll put some stuff about it in the extensions.conf sample, etc. ........ ................ 2008-12-16 15:02 +0000 [r164521-164625] Russell Bryant * /, apps/app_minivm.c: Merged revisions 164623 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r164623 | russell | 2008-12-16 09:00:27 -0600 (Tue, 16 Dec 2008) | 5 lines Set MINIVM_ACCMESS_STATUS in all cases. Also, remove a variable that was not needed. (closes issue #14081) Reported by: pkempgen ........ * /, res/res_musiconhold.c: Merged revisions 164606 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r164606 | russell | 2008-12-16 08:31:02 -0600 (Tue, 16 Dec 2008) | 13 lines Merged revisions 164605 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164605 | russell | 2008-12-16 08:28:10 -0600 (Tue, 16 Dec 2008) | 5 lines Don't try to change working directory if a directory was not configured. (closes issue #14089) Reported by: caspy ........ ................ * channels/chan_dahdi.c, /: Merged revisions 164602 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r164602 | russell | 2008-12-16 08:17:45 -0600 (Tue, 16 Dec 2008) | 7 lines Fix usage of the DAHDI_VMWI ioctl. (closes issue #14090) Reported by: alecdavis Patches: chan_dahdi.VMWI_ioctl.diff.txt uploaded by alecdavis (license 585) ........ * channels/chan_iax2.c, /: Merged revisions 164525 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r164525 | russell | 2008-12-15 16:25:46 -0600 (Mon, 15 Dec 2008) | 6 lines Open a timer before loading configuration so that the trunking configuration option will take effect. (closes issue #14082) Reported by: seandarcy ........ * channels/chan_iax2.c, /: Merged revisions 164522 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r164522 | russell | 2008-12-15 16:22:43 -0600 (Mon, 15 Dec 2008) | 4 lines Fix log message to refer to the generic timing interface, not DAHDI specifically (inspired by issue #14082) ........ * main/frame.c, /: Merged revisions 164519 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r164519 | russell | 2008-12-15 15:53:30 -0600 (Mon, 15 Dec 2008) | 7 lines Make sure we handle a uint32_t payload in ast_frdup() (closes issue #14080) Reported by: fnordian Patches: frame.patch uploaded by fnordian (license 110) ........ 2008-12-15 19:54 +0000 [r164421-164425] Mark Michelson * /, include/asterisk/pbx.h: Merged revisions 164423 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r164423 | mmichelson | 2008-12-15 13:53:29 -0600 (Mon, 15 Dec 2008) | 11 lines Merged revisions 164422 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164422 | mmichelson | 2008-12-15 13:53:08 -0600 (Mon, 15 Dec 2008) | 3 lines Add the deadlock note to ast_spawn_extension as well ........ ................ * /, include/asterisk/channel.h, include/asterisk/pbx.h: Merged revisions 164419 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r164419 | mmichelson | 2008-12-15 13:51:24 -0600 (Mon, 15 Dec 2008) | 12 lines Merged revisions 164416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164416 | mmichelson | 2008-12-15 13:45:07 -0600 (Mon, 15 Dec 2008) | 4 lines Add notes to autoservice and pbx doxygen regarding a potential deadlock scenario so that it is avoided in the future ........ ................ 2008-12-15 18:27 +0000 [r164355] Tilghman Lesher * /, cdr/cdr_pgsql.c: Merged revisions 164349 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r164349 | tilghman | 2008-12-15 12:09:58 -0600 (Mon, 15 Dec 2008) | 4 lines When querying for the structure of the CDR table, remove the schema, if it exists. (Closes issue #14058) ........ 2008-12-15 18:14 +0000 [r164314-164353] Joshua Colp * /, channels/chan_sip.c: Merged revisions 164351 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r164351 | file | 2008-12-15 14:12:24 -0400 (Mon, 15 Dec 2008) | 13 lines Merged revisions 164350 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164350 | file | 2008-12-15 14:11:21 -0400 (Mon, 15 Dec 2008) | 6 lines Do not try to unlock a non-existant channel if the transfer fails. (closes issue #13800) Reported by: dwagner Patches: asterisk-1.4.22-chan-sip-nullp.patch uploaded by tweety (license 608) ........ ................ * /, main/file.c: Merged revisions 164312 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r164312 | file | 2008-12-15 13:24:28 -0400 (Mon, 15 Dec 2008) | 4 lines Use ast_seekstream to return the file stream back to the beginning instead of directly seeking to zero. This is because some audio formats have headers at the front that need to be skipped, which will be done by the format module. (closes issue #14079) Reported by: elguero ........ 2008-12-15 16:32 +0000 [r164276-164300] Russell Bryant * main/channel.c, /, main/features.c: Merged revisions 164203 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r164203 | russell | 2008-12-15 08:40:24 -0600 (Mon, 15 Dec 2008) | 39 lines Merged revisions 164201 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164201 | russell | 2008-12-15 08:31:37 -0600 (Mon, 15 Dec 2008) | 31 lines Handle a case where a call can be bridged to a channel that is still ringing. The issue that was reported was about a case where a RINGING channel got redirected to an extension to pick up a call from parking. Once the parked call got taken out of parking, it heard silence until the other side answered. Ideally, the caller that was parked would get a ringing indication. This patch fixes this case so that the caller receives ringback once it comes out of parking until the other side answers. The fixes are: - Make sure we remember that a channel was an outgoing channel when doing a masquerade. This prevents an erroneous ast_answer() call on the channel, which causes a bogus 200 OK to be sent in the case of SIP. - Add some additional comments to explain related parts of code. - Update the handling of the ast_channel visible_indication field. Storing values that are not stateful is pointless. Control frames that are events or commands should be ignored. - When a bridge first starts, check to see if the peer channel needs to be given ringing indication because the calling side is still ringing. - Rework ast_indicate_data() a bit for the sake of readability. (closes issue #13747) Reported by: davidw Tested by: russell Review: http://reviewboard.digium.com/r/90/ ........ ................ * /, pbx/pbx_dundi.c: Merged revisions 164272 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r164272 | russell | 2008-12-15 10:17:55 -0600 (Mon, 15 Dec 2008) | 8 lines When a reload is issued, always process the configuration for dundi.conf. The reason is that a reload can be used to refresh DNS lookups for defined peers. Even if the config file hasn't changed, we want to process it for that purpose. (closes issue #13776) Reported by: kombjuder ........ 2008-12-15 16:18 +0000 [r164273-164274] Mark Michelson * apps/app_queue.c, /: Merged revisions 164270 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r164270 | mmichelson | 2008-12-15 10:16:47 -0600 (Mon, 15 Dec 2008) | 4 lines Fix a compile warning and a logic error that could have been bad for non-realtime queues ........ * apps/app_queue.c, /: Merged revisions 164268 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r164268 | mmichelson | 2008-12-15 10:10:43 -0600 (Mon, 15 Dec 2008) | 17 lines Fix up a few issues with regards to queues * Fix reference counting used in the __queues_show function * Add code to be sure that the "queue show" command does not print information for a realtime queue which has been deleted from the backend * Add a missing unref to the realtime queue loading function for the case where a queue is in the module's container but has been deleted from the realtime backend (closes issue #14033) Reported by: cristiandimache Patches: 14033.patch uploaded by putnopvut (license 60) Tested by: cristiandimache ........ 2008-12-15 15:50 +0000 [r164266] Joshua Colp * /, configure, include/asterisk/autoconfig.h.in, apps/app_fax.c, configure.ac: Merged revisions 164257 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r164257 | file | 2008-12-15 11:41:22 -0400 (Mon, 15 Dec 2008) | 4 lines Make app_fax compatible with newer versions of spandsp. This remains backwards compatible with earlier versions though so do not fret. (closes issue #14073) Reported by: seandarcy ........ 2008-12-13 01:01 +0000 [r163914] Joshua Colp * apps/app_chanspy.c, /: Merged revisions 163912 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r163912 | file | 2008-12-12 20:59:24 -0400 (Fri, 12 Dec 2008) | 2 lines Only detach and destroy the whisper audiohooks if they are actually in use. ........ 2008-12-13 00:08 +0000 [r163875] Terry Wilson * apps/app_queue.c, /: Merged revisions 163873 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r163873 | twilson | 2008-12-12 17:48:26 -0600 (Fri, 12 Dec 2008) | 6 lines When using realtime queues, app_queue wasn't updating the strategy if it was changed in the realtime backend. This patch resolves the issue for almost all situations. It is currently not supported to switch to the linear strategy via realtime since the ao2_container for members will have been set to have multiple buckets and therefore the members would be unordered. (closes issue #14034) Reported by: cristiandimache Tested by: otherwiseguy, cristiandimache ........ 2008-12-12 23:08 +0000 [r163830] Russell Bryant * /: Merged revisions 163829 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ ........ 2008-12-12 22:05 +0000 [r163764] Tilghman Lesher * main/asterisk.c, main/editline/read.c, /: Merged revisions 163762 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r163762 | tilghman | 2008-12-12 16:04:26 -0600 (Fri, 12 Dec 2008) | 14 lines Merged revisions 163761 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163761 | tilghman | 2008-12-12 16:03:10 -0600 (Fri, 12 Dec 2008) | 7 lines Simple fix for Ctrl-C not immediately exiting Asterisk, but also add a pointer inside editline to look back to asterisk.c, so others don't spend as much time as I did looking (in the wrong place) for the appropriate function. Reported by: ZX81, via the #asterisk-users channel Fixed by: me (license 14) ........ ................ 2008-12-12 19:58 +0000 [r163715] Steve Murphy * channels/chan_dahdi.c, /: Merged revisions 163675 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r163675 | murf | 2008-12-12 12:16:32 -0700 (Fri, 12 Dec 2008) | 1 line demote always-appearing debug message (for certain boards) to ast_debug lev 3 msg instead ........ 2008-12-12 18:53 +0000 [r163656-163672] Russell Bryant * main/tcptls.c, /, channels/chan_sip.c: Merged revisions 163670 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r163670 | russell | 2008-12-12 12:45:03 -0600 (Fri, 12 Dec 2008) | 6 lines Rename a number of tcptls_session variables. There are no functional changes here. The name "ser" was used in a lot of places. However, it is a relic from when the struct was a server_instance, not a session_instance. It was renamed since it represents both a server or client connection. ........ * /, channels/chan_sip.c: Merged revisions 163667 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r163667 | russell | 2008-12-12 12:33:27 -0600 (Fri, 12 Dec 2008) | 5 lines Fix a small race condition in sip_tcp_locate(). We must increase the reference count on the tcptls_session _before_ unlocking the thread list. ........ * /, channels/chan_sip.c: Merged revisions 163642 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r163642 | russell | 2008-12-12 12:19:47 -0600 (Fri, 12 Dec 2008) | 7 lines Resolve crashes when using SIP TCP/TLS with qualify. The problem was a reference count error on the tcptls_session structure. (closes issue #13989) Reported by: Nugget ........ 2008-12-12 18:19 +0000 [r163640] Joshua Colp * /, channels/chan_sip.c: Merged revisions 163629 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r163629 | file | 2008-12-12 14:17:12 -0400 (Fri, 12 Dec 2008) | 4 lines When a device registers we need to unlink them (if linked) from the peers_by_ip container and link them back in since their IP address has changed. This would have manifested itself if you configured a new device (as type=peer), registered, and then tried to place a call from the device. Since the peer was not linked into the peers_by_ip container it would have never been found. (closes issue #13811) Reported by: pj ........ 2008-12-12 17:26 +0000 [r163624] Michiel van Baak * res/res_monitor.c, /: Merged revisions 163612 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r163612 | mvanbaak | 2008-12-12 18:22:47 +0100 (Fri, 12 Dec 2008) | 7 lines Document default Monitor file location. (closes issue #14065) Reported by: kshumard Patches: res_monitor.documentation.patch.txt uploaded by kshumard (license 92) ........ 2008-12-12 16:57 +0000 [r163581] Joshua Colp * main/channel.c, /, channels/chan_sip.c: Merged revisions 163579 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r163579 | file | 2008-12-12 12:55:15 -0400 (Fri, 12 Dec 2008) | 4 lines Since chan_sip is callback devicestate driven do not pass in actual states, pass in unknown so we get asked. Additionally do not pass in an actual device state value in ast_setstate since the channel may be callback driven. (closes issue #13525) Reported by: pj ........ 2008-12-12 14:48 +0000 [r163514-163515] Russell Bryant * main/channel.c, main/autoservice.c, /, include/asterisk/channel.h: Merged revisions 163449 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r163449 | russell | 2008-12-12 07:55:30 -0600 (Fri, 12 Dec 2008) | 34 lines Merged revisions 163448 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163448 | russell | 2008-12-12 07:44:08 -0600 (Fri, 12 Dec 2008) | 26 lines Resolve issues that could cause DTMF to be processed out of order. These changes come from team/russell/issue_12658 1) Change autoservice to put digits on the head of the channel's frame readq instead of the tail. If there were frames on the readq that autoservice had not yet read, the previous code would have resulted in out of order processing. This required a new API call to queue a frame to the head of the queue instead of the tail. 2) Change up the processing of DTMF in ast_read(). Some of the problems were the result of having two sources of pending DTMF frames. There was the dtmfq and the more generic readq. Both were used for pending DTMF in various scenarios. Simplifying things to only use the frame readq avoids some of the problems. 3) Fix a bug where a DTMF END frame could get passed through when it shouldn't have. If code set END_DTMF_ONLY in the middle of digit emulation, and a digit arrived before emulation was complete, digits would get processed out of order. (closes issue #12658) Reported by: dimas Tested by: russell, file Review: http://reviewboard.digium.com/r/85/ ........ ................ * /, pbx/pbx_dundi.c: Merged revisions 163512 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r163512 | russell | 2008-12-12 08:44:06 -0600 (Fri, 12 Dec 2008) | 13 lines Merged revisions 163511 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163511 | russell | 2008-12-12 08:40:31 -0600 (Fri, 12 Dec 2008) | 5 lines Specify uint32_t for variables storing a CRC32 so that it is actually 32 bits on 64-bit machines, as well. (inspired by issue #13879) ........ ................ 2008-12-11 23:48 +0000 [r163386] Tilghman Lesher * main/asterisk.c, /: Merged revisions 163384 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r163384 | tilghman | 2008-12-11 17:38:56 -0600 (Thu, 11 Dec 2008) | 16 lines Merged revisions 163383 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163383 | tilghman | 2008-12-11 17:35:55 -0600 (Thu, 11 Dec 2008) | 9 lines When a Ctrl-C or Ctrl-D ends a remote console, on certain shells, the terminal is messed up. By intercepting those events with a signal handler in the remote console, we can avoid those issues. (closes issue #13464) Reported by: tzafrir Patches: 20081110__bug13464.diff.txt uploaded by Corydon76 (license 14) Tested by: blitzrage ........ ................ 2008-12-11 22:52 +0000 [r163319] Matt Nicholson * /, pbx/pbx_dundi.c: Merged revisions 163317 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r163317 | mnicholson | 2008-12-11 16:49:59 -0600 (Thu, 11 Dec 2008) | 16 lines Merged revisions 163316 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163316 | mnicholson | 2008-12-11 16:44:31 -0600 (Thu, 11 Dec 2008) | 9 lines Clean up the dundi cache every 5 minutes. (closes issue #13819) Reported by: adomjan Patches: pbx_dundi.c-clearcache.patch uploaded by adomjan (license 487) dundi_clearecache3.diff uploaded by mnicholson (license 96) Tested by: adomjan ........ ................ 2008-12-11 21:50 +0000 [r163252-163256] Russell Bryant * /, funcs/func_strings.c, funcs/func_cut.c: Merged revisions 163254 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r163254 | russell | 2008-12-11 15:48:08 -0600 (Thu, 11 Dec 2008) | 16 lines Merged revisions 163253 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163253 | russell | 2008-12-11 15:46:29 -0600 (Thu, 11 Dec 2008) | 8 lines Fix some observed slowdowns in dialplan processing. The change is to remove autoservice usage from dialplan functions that do not need it because they do not perform operations that potentially block. (closes issue #13940) Reported by: tbelder ........ ................ * /, res/res_timing_pthread.c: Merged revisions 163241 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r163241 | russell | 2008-12-11 15:21:31 -0600 (Thu, 11 Dec 2008) | 8 lines Fix a problem where continuous mode will get inadvertently get turned off if set_rate() is used while continuous mode was already turned on. (closes issue #13738) Reported by: smurfix Patches: res.patch.fixed uploaded by smurfix (license 547) ........ 2008-12-11 21:00 +0000 [r163214] Mark Michelson * apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged revisions 163213 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r163213 | mmichelson | 2008-12-11 14:57:44 -0600 (Thu, 11 Dec 2008) | 9 lines Add an option to voicemail.conf to allow urgent messages to be forwarded as not urgent. (closes issue #14063) Reported by: jaroth Patches: urgfwd_v2.patch uploaded by jaroth (license 50) ........ 2008-12-11 20:10 +0000 [r163173] Russell Bryant * main/channel.c, /: Merged revisions 163171 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r163171 | russell | 2008-12-11 14:07:47 -0600 (Thu, 11 Dec 2008) | 16 lines Fix the "failed" extension for outgoing calls. The conversion to use ast_check_hangup() everywhere instead of checking the softhangup flag directly introduced this problem. The issue is that ast_check_hangup() checked for tech_pvt to be NULL. Unfortunately, this will be NULL is some valid circumstances, such as with a dummy channel. The fix is simple. Don't check tech_pvt. It's pointless, because the code path that sets this to NULL is when the channel hangup callback gets called. This happens inside of ast_hangup(), which is the same function responsible for freeing the channel. Any code calling ast_check_hangup() better not be calling it after that point, and if so, we have a bigger problem at hand. (closes issue #14035) Reported by: erogoza ........ 2008-12-11 20:05 +0000 [r163170] Tilghman Lesher * /, configure, configure.ac: Merged revisions 163168 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r163168 | tilghman | 2008-12-11 14:02:35 -0600 (Thu, 11 Dec 2008) | 5 lines Sometimes even Linux needs -lm to link libtonezone, such as when libtonezone is compiled statically. (closes issue #13887) Reported by: tzafrir ........ 2008-12-11 17:16 +0000 [r163100] Russell Bryant * /, main/features.c: Merged revisions 163094 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r163094 | russell | 2008-12-11 11:06:16 -0600 (Thu, 11 Dec 2008) | 19 lines Merged revisions 163092 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163092 | russell | 2008-12-11 10:54:51 -0600 (Thu, 11 Dec 2008) | 11 lines Fix an issue that made it so you could only have a single caller executing a custom feature at a time. This was especially problematic when custom features ran for any appreciable amount of time. The fix turned out to be quite simple. The dynamic features are now stored in a read/write list instead of a list using a mutex. (closes issue #13478) Reported by: neutrino88 Fix suggested by file ........ ................ 2008-12-11 16:54 +0000 [r163091] Tilghman Lesher * /, res/res_agi.c: Merged revisions 163089 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r163089 | tilghman | 2008-12-11 10:52:24 -0600 (Thu, 11 Dec 2008) | 13 lines Merged revisions 163088 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163088 | tilghman | 2008-12-11 10:51:27 -0600 (Thu, 11 Dec 2008) | 6 lines Don't wait forever, if there's a specified recording timeout. (closes issue #13885) Reported by: bamby Patches: res_agi.c.patch uploaded by bamby (license 430) ........ ................ 2008-12-11 16:49 +0000 [r163083-163087] Mark Michelson * apps/app_queue.c, /: Merged revisions 163085 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r163085 | mmichelson | 2008-12-11 10:47:34 -0600 (Thu, 11 Dec 2008) | 12 lines Merged revisions 163084 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163084 | mmichelson | 2008-12-11 10:46:22 -0600 (Thu, 11 Dec 2008) | 4 lines Revert this cast to long. Using time_t here causes build failures on a FreeBSD 32-bit build. ........ ................ * apps/app_queue.c, /: Merged revisions 163081 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r163081 | mmichelson | 2008-12-11 10:33:16 -0600 (Thu, 11 Dec 2008) | 22 lines Merged revisions 163080 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163080 | mmichelson | 2008-12-11 10:24:43 -0600 (Thu, 11 Dec 2008) | 14 lines Fix a potential crash due to unsafe datastore handling. This patch also contains a conversion from using long to time_t for representing times for a queue, as well as some whitespace fixes. (closes issue #14060) Reported by: nivek Patches: datastore_fixup.patch.corrected uploaded by nivek (license 636) with slight modification from me Tested by: nivek ........ ................ 2008-12-11 15:07 +0000 [r163006] Joshua Colp * /, channels/chan_sip.c: Merged revisions 162997 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r162997 | file | 2008-12-11 11:05:49 -0400 (Thu, 11 Dec 2008) | 4 lines When a device registers to use it is entirely possible that they may be in use, so tell the core that we don't know the devstate and have it ask us for it. (closes issue #13525) Reported by: pj ........ 2008-12-10 23:13 +0000 [r162949] Tilghman Lesher * main/pbx.c, /: Merged revisions 162922,162930 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r162922 | tilghman | 2008-12-10 16:48:09 -0600 (Wed, 10 Dec 2008) | 7 lines Checking global variables here actually overwrote the previous substitution by channel variables, and in any case, was redundant; pbx_substitute_variables_helper ALREADY does substitution for global variables. (closes issue #13327) Reported by: pj ........ r162930 | tilghman | 2008-12-10 17:01:14 -0600 (Wed, 10 Dec 2008) | 2 lines Previously missing line, now the substitution works correctly ........ 2008-12-10 22:54 +0000 [r162896-162929] Jeff Peeler * /, res/res_musiconhold.c: Merged revisions 162927 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162927 | jpeeler | 2008-12-10 16:53:34 -0600 (Wed, 10 Dec 2008) | 11 lines Merged revisions 162926 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162926 | jpeeler | 2008-12-10 16:52:51 -0600 (Wed, 10 Dec 2008) | 3 lines Oops, inverted logic for a strcasecmp check. Pointed out by mmichelson, thanks! ........ ................ * /, res/res_musiconhold.c: Merged revisions 162891 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162891 | jpeeler | 2008-12-10 16:11:46 -0600 (Wed, 10 Dec 2008) | 13 lines Merged revisions 162874 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162874 | jpeeler | 2008-12-10 16:04:18 -0600 (Wed, 10 Dec 2008) | 5 lines (closes issue #13229) Reported by: clegall_proformatique Ensure that moh_generate does not return prematurely before local_ast_moh_stop is called. Also, the sleep in mp3_spawn now only occurs for http locations since it seems to have been added originally only for failing media streams. ........ ................ 2008-12-10 19:05 +0000 [r162741-162807] Joshua Colp * /, channels/chan_sip.c: Merged revisions 162805 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162805 | file | 2008-12-10 15:02:57 -0400 (Wed, 10 Dec 2008) | 13 lines Merged revisions 162804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162804 | file | 2008-12-10 15:01:17 -0400 (Wed, 10 Dec 2008) | 6 lines Fix subscription based MWI up a bit. We only want to put sip: at the beginning of the URI if it is not already there and revert code to ignore destination check if subscribing for MWI. (closes issue #12560) Reported by: vsauer Patches: patch001.diff uploaded by ramonpeek (license 266) ........ ................ * /, channels/chan_sip.c: Merged revisions 162739 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162739 | file | 2008-12-10 13:53:09 -0400 (Wed, 10 Dec 2008) | 13 lines Merged revisions 162738 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162738 | file | 2008-12-10 13:50:43 -0400 (Wed, 10 Dec 2008) | 6 lines When a SIP peer unregisters set the expiry time back to 0 so that the 200 OK contains an expires of 0. (closes issue #13599) Reported by: hjourdain Patches: chan_sip.c.diff uploaded by hjourdain (license 583) ........ ................ 2008-12-10 16:39 +0000 [r162666-162669] Mark Michelson * doc/tex/misdn.tex, /: Merged revisions 162667 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162667 | mmichelson | 2008-12-10 10:39:10 -0600 (Wed, 10 Dec 2008) | 16 lines Merged revisions 162659 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162659 | mmichelson | 2008-12-10 10:10:25 -0600 (Wed, 10 Dec 2008) | 8 lines Add missing documentation to misdn.txt (closes issue #14052) Reported by: festr Patches: misdn.txt.patch uploaded by festr (license 443) ........ ................ * /, channels/chan_sip.c: Merged revisions 162664 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162664 | mmichelson | 2008-12-10 10:34:35 -0600 (Wed, 10 Dec 2008) | 19 lines Merged revisions 162663 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162663 | mmichelson | 2008-12-10 10:24:56 -0600 (Wed, 10 Dec 2008) | 11 lines Revert fix for issue 13570. It has caused more problems than it helped to fix. (closes issue #13783) Reported by: navkumar (closes issue #14025) Reported by: ffs ........ ................ 2008-12-10 16:08 +0000 [r162622-162658] Joshua Colp * main/rtp.c, /: Merged revisions 162656 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162656 | file | 2008-12-10 12:06:59 -0400 (Wed, 10 Dec 2008) | 13 lines Merged revisions 162653 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162653 | file | 2008-12-10 12:05:29 -0400 (Wed, 10 Dec 2008) | 6 lines Increment the sequence number on the end packets for RFC2833. After reading the RFC some more and doing some testing I agree with this change. (closes issue #12983) Reported by: vt Patches: dtmf_inc_seqnum_on_end_pkts.diff uploaded by vt (license 520) ........ ................ * /, channels/chan_sip.c: Merged revisions 162619 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r162619 | file | 2008-12-10 11:22:26 -0400 (Wed, 10 Dec 2008) | 4 lines When transmitting a register set the socket port to the local one for the transport being used, not the port for the remote server. (closes issue #13633) Reported by: performer ........ 2008-12-10 11:37 +0000 [r162585] Michiel van Baak * /, res/snmp/agent.c: Merged revisions 162583 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r162583 | mvanbaak | 2008-12-10 12:34:09 +0100 (Wed, 10 Dec 2008) | 5 lines Make res_snmp.so compile on OpenBSD. OpenBSD uses an old version of gcc which throws an error if you use a macro that's not #defined ........ 2008-12-09 23:45 +0000 [r162490] Mark Michelson * include/asterisk/stringfields.h, /: Merged revisions 162488 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r162488 | kpfleming | 2008-12-09 17:41:02 -0600 (Tue, 09 Dec 2008) | 1 line it does help if the compiler attribute syntax is correct ........ 2008-12-09 23:12 +0000 [r162472] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 162466 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162466 | tilghman | 2008-12-09 17:10:34 -0600 (Tue, 09 Dec 2008) | 9 lines Merged revisions 162463 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162463 | tilghman | 2008-12-09 17:08:53 -0600 (Tue, 09 Dec 2008) | 2 lines Oops, should be "tz", not "zonetag". ........ ................ 2008-12-09 22:34 +0000 [r162416] Russell Bryant * main/asterisk.c, include/asterisk/utils.h, /, main/utils.c: Merged revisions 162414 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162414 | russell | 2008-12-09 16:25:06 -0600 (Tue, 09 Dec 2008) | 16 lines Merged revisions 162413 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162413 | russell | 2008-12-09 16:17:39 -0600 (Tue, 09 Dec 2008) | 8 lines Remove the test_for_thread_safety() function completely. The test is not valid. Besides, if we actually suspected that recursive mutexes were not working, we would get a ton of LOG_ERROR messages when DEBUG_THREADS is turned on. (inspired by a discussion on the asterisk-dev list) ........ ................ 2008-12-09 22:02 +0000 [r162372] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 162355 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162355 | tilghman | 2008-12-09 15:57:09 -0600 (Tue, 09 Dec 2008) | 11 lines Merged revisions 162348 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162348 | tilghman | 2008-12-09 15:53:25 -0600 (Tue, 09 Dec 2008) | 4 lines We appear to have documented tz= in the [general] section of voicemail.conf, without actually having implemented it. Oops. (Reported by Olivier on the -users list) ........ ................ 2008-12-09 21:18 +0000 [r162344] Joshua Colp * /, apps/app_directed_pickup.c: Merged revisions 162342 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162342 | file | 2008-12-09 17:16:37 -0400 (Tue, 09 Dec 2008) | 11 lines Merged revisions 162341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162341 | file | 2008-12-09 17:14:29 -0400 (Tue, 09 Dec 2008) | 4 lines Add 'down' as a valid state for directed call pickup. This creeps up when we receive session progress when dialing a device and not ringing. (closes issue #14005) Reported by: ddl ........ ................ 2008-12-09 21:03 +0000 [r162302] Russell Bryant * /, apps/app_meetme.c: Merged revisions 162291 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162291 | russell | 2008-12-09 14:59:54 -0600 (Tue, 09 Dec 2008) | 17 lines Merged revisions 162286 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162286 | russell | 2008-12-09 14:57:35 -0600 (Tue, 09 Dec 2008) | 9 lines Fix an issue where callers on an incoming call on an SLA trunk would not hear ringback. We need to make sure that we don't start writing audio to the trunk channel until we're actually ready to answer it. Otherwise, the channel driver will treat it as inband progress, even though all they are getting is silence. (closes issue #12471) Reported by: mthomasslo ........ ................ 2008-12-09 20:48 +0000 [r162278] Joshua Colp * /, apps/app_festival.c: Merged revisions 162275 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162275 | file | 2008-12-09 16:46:11 -0400 (Tue, 09 Dec 2008) | 11 lines Merged revisions 162273 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162273 | file | 2008-12-09 16:44:32 -0400 (Tue, 09 Dec 2008) | 4 lines Fix double declaration of 'x' on the PPC platform. (closes issue #14038) Reported by: ffloimair ........ ................ 2008-12-09 20:47 +0000 [r162277] Steve Murphy * res/ael/ael.flex, /, res/ael/ael_lex.c: Merged revisions 162271 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162271 | murf | 2008-12-09 13:40:31 -0700 (Tue, 09 Dec 2008) | 9 lines Merged revisions 162264 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162264 | murf | 2008-12-09 13:20:54 -0700 (Tue, 09 Dec 2008) | 1 line In discussion with seanbright on #asterisk-dev, I have added a default rule, and an option to suppress the default rule from being generated in the flex output, for the sake of those OS's where they didn't tweak flex's ECHO macro, and the compiler doesn't like it. The regressions are OK with this. ........ ................ 2008-12-09 20:31 +0000 [r162269] Mark Michelson * main/pbx.c, /: Merged revisions 162266 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162266 | mmichelson | 2008-12-09 14:30:07 -0600 (Tue, 09 Dec 2008) | 14 lines Merged revisions 162265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162265 | mmichelson | 2008-12-09 14:28:44 -0600 (Tue, 09 Dec 2008) | 6 lines If we fail to start a thread for the pbx to run in, we need to be sure to decrease the number of active calls on the system. This fix may relate to ABE-1713, but it is not certain yet. ........ ................ 2008-12-09 19:52 +0000 [r162202-162207] Joshua Colp * main/rtp.c, /: Merged revisions 162205 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162205 | file | 2008-12-09 15:48:35 -0400 (Tue, 09 Dec 2008) | 14 lines Merged revisions 162204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162204 | file | 2008-12-09 15:47:07 -0400 (Tue, 09 Dec 2008) | 7 lines Make sure that the timestamp for DTMF is not the same as the previous voice frame and do not send audio when transmitting DTMF as this confuses some equipment. (closes issue #13209) Reported by: ip-rob Patches: 13209.diff uploaded by file (license 11) Tested by: ip-rob, bujones ........ ................ * main/rtp.c, /: Merged revisions 162197 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162197 | file | 2008-12-09 15:08:39 -0400 (Tue, 09 Dec 2008) | 11 lines Merged revisions 162188 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162188 | file | 2008-12-09 15:06:14 -0400 (Tue, 09 Dec 2008) | 4 lines Take video into account when early bridging RTP. (closes issue #13535) Reported by: davidw ........ ................ 2008-12-09 18:49 +0000 [r162082-162142] Steve Murphy * res/ael/ael.flex, /, res/ael/ael_lex.c: Merged revisions 162140 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162140 | murf | 2008-12-09 11:35:35 -0700 (Tue, 09 Dec 2008) | 9 lines Merged revisions 162136 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162136 | murf | 2008-12-09 11:13:39 -0700 (Tue, 09 Dec 2008) | 1 line Previous fix used ast_malloc and ast_copy_string and messed up the standalone stuff. Fixed. ........ ................ * res/ael/ael.flex, res/ael/pval.c, /, include/asterisk/pval.h, res/ael/ael_lex.c: Merged revisions 162079 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162079 | murf | 2008-12-09 10:18:03 -0700 (Tue, 09 Dec 2008) | 53 lines Merged revisions 162013 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162013 | murf | 2008-12-09 09:31:55 -0700 (Tue, 09 Dec 2008) | 45 lines (closes issue #14019) Reported by: ckjohnsonme Patches: 14019.diff uploaded by murf (license 17) Tested by: ckjohnsonme, murf This crash was the result of a few small errors that would combine in 64-bit land to result in a crash. 32-bit land might have seen these combine to mysteriously drop the args to an application call, in certain circumstances. Also, in trying to find this bug, I spotted a situation in the flex input, where, in passing back a 'word' to the parser, it would allocate a buffer larger than necessary. I changed the usage in such situations, so that strdup was not used, but rather, an ast_malloc, followed by ast_copy_string. I removed a field from the pval struct, in u2, that was never getting used, and set in one spot in the code. I believe it was an artifact of a previous fix to make switch cases work invisibly with extens. And, for goto's I removed a '!' from before a strcmp, that has been there since the initial merging of AEL2, that might prevent the proper target of a goto from being found. This was pretty harmless on its own, as it would just louse up a consistency check for users. Many thanks to ckjohnsonme for providing a simplified and complete set of information about the bug, that helped considerably in finding and fixing the problem. Now, to get aelparse up and running again in trunk, and out of its "horribly broken" state, so I can run the regression suite! ........ ................ 2008-12-09 16:50 +0000 [r161963-162018] Russell Bryant * /, apps/app_disa.c: Merged revisions 162016 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162016 | russell | 2008-12-09 10:47:39 -0600 (Tue, 09 Dec 2008) | 13 lines Merged revisions 162014 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162014 | russell | 2008-12-09 10:46:53 -0600 (Tue, 09 Dec 2008) | 5 lines Allow DISA to handle extensions that start with #. (closes issue #13330) Reported by: jcovert ........ ................ * /, main/app.c: Merged revisions 161951 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r161951 | russell | 2008-12-09 08:57:39 -0600 (Tue, 09 Dec 2008) | 23 lines Merged revisions 161948 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r161948 | russell | 2008-12-09 08:52:25 -0600 (Tue, 09 Dec 2008) | 15 lines Fix a problem with GROUP() settings on a masquerade. The previous code carried over group settings from the old channel to the new one. However, it did nothing with the group settings that were already on the new channel. This patch removes all group settings that already existed on the new channel. I have a more complicated version of this patch which addresses only the most blatant problem with this, which is that a channel can end up with multiple group settings in the same category. However, I could not think of a use case for keeping any of the group settings from the old channel, so I went this route for now. (closes AST-152) ........ ................ 2008-12-08 20:55 +0000 [r161835] Joshua Colp * contrib/scripts/autosupport, /, contrib/scripts/autosupport.8: Merged revisions 161830 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r161830 | file | 2008-12-08 16:53:50 -0400 (Mon, 08 Dec 2008) | 2 lines Update autosupport script with a few changes. ........ 2008-12-08 18:52 +0000 [r161792] Tilghman Lesher * main/manager.c, /: Merged revisions 161790 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r161790 | tilghman | 2008-12-08 12:49:50 -0600 (Mon, 08 Dec 2008) | 6 lines Allocate enough space initially for the message. (closes issue #14027) Reported by: junky Patches: M14027.diff uploaded by junky (license 177) ........ 2008-12-08 18:49 +0000 [r161729-161789] Joshua Colp * main/pbx.c, /: Merged revisions 161787 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r161787 | file | 2008-12-08 14:47:32 -0400 (Mon, 08 Dec 2008) | 6 lines Fix a regression introduced when the PBX timeouts were converted to milliseconds. collect_digits now gets milliseconds fed to it, not seconds. (closes issue #14012) Reported by: dveiga Patches: 14012.patch uploaded by bkruse (license 132) ........ * /, channels/chan_sip.c: Merged revisions 161726 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r161726 | file | 2008-12-08 13:53:32 -0400 (Mon, 08 Dec 2008) | 13 lines Merged revisions 161725 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r161725 | file | 2008-12-08 13:52:10 -0400 (Mon, 08 Dec 2008) | 6 lines Make the usereqphone option work again. (closes issue #13474) Reported by: mmaguire Patches: 20080912_bug13474.diff uploaded by mmaguire (license 571) ........ ................ 2008-12-08 17:24 +0000 [r161722] Matt Nicholson * /, channels/chan_sip.c: Merged revisions 161721 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r161721 | mnicholson | 2008-12-08 11:23:41 -0600 (Mon, 08 Dec 2008) | 7 lines Fix a crash that can occur on a transfer in chan_sip when attempting to collect rtp stats. (closes issue #13956) Reported by: chris-mac Tested by: chris-mac ........ 2008-12-05 23:29 +0000 [r161496] Mark Michelson * apps/app_stack.c, /: Merged revisions 161493 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r161493 | mmichelson | 2008-12-05 17:24:38 -0600 (Fri, 05 Dec 2008) | 8 lines If the autoloop flag is set on a channel, then we need to add 1 to the priority when checking if the extension exists. Otherwise, gosubs will fail. This was discovered when investigating an asterisk-users mailing list post made by Gary Hawkins. ........ 2008-12-05 21:16 +0000 [r161352-161429] Sean Bright * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions 161427 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r161427 | seanbright | 2008-12-05 16:08:43 -0500 (Fri, 05 Dec 2008) | 22 lines Merged revisions 161426 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r161426 | seanbright | 2008-12-05 16:02:20 -0500 (Fri, 05 Dec 2008) | 15 lines Merged revisions 161421 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r161421 | seanbright | 2008-12-05 15:50:23 -0500 (Fri, 05 Dec 2008) | 8 lines Fix build errors on FreeBSD (uint -> unsigned int). (closes issue #14006) Reported by: alphaque Patches: astobj2.h-patch uploaded by alphaque (license 259) (Slightly modified by seanbright) ........ ................ ................ * apps/app_voicemail.c, /: Merged revisions 161349-161350 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r161349 | seanbright | 2008-12-05 10:56:15 -0500 (Fri, 05 Dec 2008) | 5 lines When using IMAP_STORAGE, it's important to convert bare newlines (\n) in emailbody and pagerbody to CR-LF so that the IMAP server doesn't spit out an error. This was informally reported on #asterisk-dev a few weeks ago. Reviewed by Mark M. on IRC. ........ r161350 | seanbright | 2008-12-05 11:04:36 -0500 (Fri, 05 Dec 2008) | 2 lines Use ast_free() instead of free(), pointed out by eliel on IRC. ........ 2008-12-05 14:18 +0000 [r161285-161290] Russell Bryant * main/pbx.c, /: Merged revisions 161288 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r161288 | russell | 2008-12-05 08:16:24 -0600 (Fri, 05 Dec 2008) | 10 lines Merged revisions 161287 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r161287 | russell | 2008-12-05 08:12:14 -0600 (Fri, 05 Dec 2008) | 2 lines Fix a NULL format string warning found by buildbot. ........ ................ * /, apps/app_minivm.c: Merged revisions 161252 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r161252 | russell | 2008-12-05 07:46:01 -0600 (Fri, 05 Dec 2008) | 2 lines Resolve a compiler warning from buildbot about a NULL format string. ........ 2008-12-05 05:42 +0000 [r161182] Tilghman Lesher * main/config.c, /: Merged revisions 161181 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r161181 | tilghman | 2008-12-04 23:41:41 -0600 (Thu, 04 Dec 2008) | 11 lines The first file should have a blank config filename in the structure, so that when a save occurs to a different filename, everything goes to the alternate filename, instead of appending to the original. This is important for the AMI command UpdateConfig. (closes issue #13301) Reported by: trevo Patches: 20081113__bug13301.diff.txt uploaded by Corydon76 (license 14) 20081113__bug13301__1.6.0.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76, blitzrage ........ 2008-12-05 02:52 +0000 [r161149] Sean Bright * apps/app_voicemail.c, /: Merged revisions 161147 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r161147 | seanbright | 2008-12-04 21:47:54 -0500 (Thu, 04 Dec 2008) | 3 lines Check the return value of fread/fwrite so the compiler doesn't complain. Only a problem when IMAP_STORAGE is enabled. ........ 2008-12-04 18:37 +0000 [r161016] Jeff Peeler * main/rtp.c, /: Merged revisions 161014 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r161014 | jpeeler | 2008-12-04 12:32:20 -0600 (Thu, 04 Dec 2008) | 17 lines Merged revisions 161013 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r161013 | jpeeler | 2008-12-04 12:30:41 -0600 (Thu, 04 Dec 2008) | 9 lines (closes issue #13835) Reported by: matt_b Tested by: jpeeler This mirrors a check that was present in ast_rtp_read to also be in ast_rtp_raw_write to not schedule sending the receiver report if the remote RTCP endpoint address isn't present in the RTCP structure. Closes AST-142. ........ ................ 2008-12-04 16:48 +0000 [r160947] Mark Michelson * /, main/callerid.c: Merged revisions 160945 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r160945 | mmichelson | 2008-12-04 10:45:06 -0600 (Thu, 04 Dec 2008) | 23 lines Merged revisions 160943 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160943 | mmichelson | 2008-12-04 10:44:18 -0600 (Thu, 04 Dec 2008) | 15 lines Fix a callerid parsing issue. If someone formatted callerid like the following: "name " (including the quotation marks), then the parts would be parsed as name: "name number: number This is because the closing quotation mark was not discovered since the number and everything after was parsed out of the string earlier. Now, there is a check to see if the closing quote occurs after the number, so that we can know if we should strip off the opening quote on the name. Closes AST-158 ........ ................ 2008-12-04 01:41 +0000 [r160858-160859] Richard Mudgett * funcs/func_callerid.c, /: Merged revisions 160856 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r160856 | rmudgett | 2008-12-03 19:36:39 -0600 (Wed, 03 Dec 2008) | 1 line Jcolp pointed out that num will also match number ........ * funcs/func_callerid.c, /: Merged revisions 160854 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r160854 | rmudgett | 2008-12-03 19:14:22 -0600 (Wed, 03 Dec 2008) | 4 lines * Found a couple more places where num/number needed to be done so 1.4 upgraders will not have problems. * Added curly braces and minor tweaks. ........ 2008-12-03 22:02 +0000 [r160811] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 160791 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r160791 | tilghman | 2008-12-03 15:58:21 -0600 (Wed, 03 Dec 2008) | 9 lines Merged revisions 160770 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160770 | tilghman | 2008-12-03 15:54:07 -0600 (Wed, 03 Dec 2008) | 2 lines Some compilers warn on null format strings; some don't (caught by buildbot) ........ ................ 2008-12-03 21:40 +0000 [r160766] Steve Murphy * funcs/func_callerid.c, /: Merged revisions 160760 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r160760 | murf | 2008-12-03 14:09:15 -0700 (Wed, 03 Dec 2008) | 23 lines Merged revisions 160703 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160703 | murf | 2008-12-03 13:41:42 -0700 (Wed, 03 Dec 2008) | 11 lines (closes issue #13597) Reported by: john8675309 Patches: patch.13597 uploaded by murf (license 17) Tested by: murf, john8675309 This patch causes the setcid func to update the CDR clid after setting the channel field. I also notice that in trunk, the num/number of 1.4 is left out; I decided to include the option to use either in trunk, so as not to have 1.4 upgraders not to have problems. ........ ................ 2008-12-03 20:36 +0000 [r160702] Jason Parker * main/manager.c, /: Merged revisions 160699-160700 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r160699 | qwell | 2008-12-03 14:32:20 -0600 (Wed, 03 Dec 2008) | 7 lines Fix typo when ListCategories returns none. (closes issue #13994) Reported by: mika Patches: ListCategoriesActionPatch.diff uploaded by mika (license 624) ........ r160700 | qwell | 2008-12-03 14:35:36 -0600 (Wed, 03 Dec 2008) | 1 line Another place this is missing ........ 2008-12-03 19:49 +0000 [r160665] Eliel C. Sardanons * /, channels/iax2-provision.c: Merged revisions 160663 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r160663 | eliel | 2008-12-03 17:25:30 -0200 (Wed, 03 Dec 2008) | 13 lines - iax2-provision was not freeing iax_templates structure when unloading the chan_iax2.so module. - Move the code to start using the LIST macros. Review: http://reviewboard.digium.com/r/72 (closes issue #13232) Reported by: eliel Patches: iax2-provision.patch.txt uploaded by eliel (license 64) (with minor changes pointed by Mark Michelson on review board) Tested by: eliel ........ 2008-12-03 18:42 +0000 [r160628] Mark Michelson * apps/app_queue.c, apps/app_stack.c, apps/app_dial.c, /: Merged revisions 160626 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r160626 | mmichelson | 2008-12-03 12:37:46 -0600 (Wed, 03 Dec 2008) | 16 lines Add some safety measures when using gosub, especially when using the options for app_dial and app_queue to run a gosub when the call is answered. * Check for the existence of the gosub target in gosub_exec. If it is nonexistent, then this will cause errors when we attempt to actually run the gosub, including a definite memory leak and potential crashes. Return an error in this situation * Check the return value of pbx_exec in app_dial and app_queue before attempting to actually run the gosub routine. If there was an error, we should not attempt to run the gosub. * Change a '|' to a ',' in app_queue. * Add some extra curly braces where they had been missing previously. (closes issue #13548) Reported by: fiddur ........ 2008-12-03 17:41 +0000 [r160561] Tilghman Lesher * pbx/pbx_spool.c, /: Merged revisions 160559 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r160559 | tilghman | 2008-12-03 11:38:59 -0600 (Wed, 03 Dec 2008) | 14 lines Merged revisions 160558 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160558 | tilghman | 2008-12-03 11:34:34 -0600 (Wed, 03 Dec 2008) | 7 lines If an entry is added to the directory during a scan when another entry expires, then that new entry will not be processed promptly, but must wait for either a future entry to start or a current entry's retry to occur. If no other entries exist in the directory (other than the new entries) when a bunch expire, then the new entries must wait until another new entry is added to be processed. This was a rather weird race condition, really. Fixes AST-147. ........ ................ 2008-12-03 17:10 +0000 [r160557] Mark Michelson * apps/app_queue.c, /: Merged revisions 160555 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r160555 | mmichelson | 2008-12-03 11:07:09 -0600 (Wed, 03 Dec 2008) | 11 lines When investigating issue #13548, I found that gosub handling in app_queue was just completely wrong, mostly because the channel operations being performed were being done on the incorrect channel. With this set of changes, a gosub will correctly run on the answering queue member's channel. There are still crash issues which occur if there are dialplan syntax errors, so I cannot yet close the referenced issue. ........ 2008-12-03 17:02 +0000 [r160483-160554] Tilghman Lesher * pbx/pbx_spool.c, /: Merged revisions 160552 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r160552 | tilghman | 2008-12-03 11:01:03 -0600 (Wed, 03 Dec 2008) | 11 lines Merged revisions 160551 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160551 | tilghman | 2008-12-03 10:58:34 -0600 (Wed, 03 Dec 2008) | 4 lines Don't start scanning the directory until all modules are loaded, because some required modules (channels, apps, functions) may not yet be in memory yet. Fixes AST-149. ........ ................ * /, channels/chan_sip.c: Merged revisions 160481 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r160481 | tilghman | 2008-12-03 08:11:53 -0600 (Wed, 03 Dec 2008) | 14 lines Merged revisions 160480 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160480 | tilghman | 2008-12-03 08:09:35 -0600 (Wed, 03 Dec 2008) | 7 lines Jon Bonilla (Manwe) pointed out on the -dev list: "I guess that having only ip-phones in mind is not a good approach. Since it is possible to have a sip proxy connected to asterisk we could receive a 407 (unauthorized) or 483 (too many hops) as response and dialog ending would not be a good behavior." So modified. ........ ................ 2008-12-02 18:05 +0000 [r160329-160339] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 160333 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r160333 | jpeeler | 2008-12-02 12:04:51 -0600 (Tue, 02 Dec 2008) | 1 line remove duplicate comment that I accidentally merged ........ * channels/chan_dahdi.c, /: Merged revisions 160319 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r160319 | jpeeler | 2008-12-02 12:00:24 -0600 (Tue, 02 Dec 2008) | 7 lines (closes issue #13786) Reported by: tzafrir Readding DAHDI_CHECK_HOOKSTATE define that was removed in r134260 which fixes not being able to make outgoing calls on some FXO adapters: http://lists.digium.com/pipermail/asterisk-users/2008-November/thread.html#221553 ........ 2008-12-02 18:03 +0000 [r160234-160325] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 160308 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r160308 | tilghman | 2008-12-02 11:56:24 -0600 (Tue, 02 Dec 2008) | 17 lines Merged revisions 160297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160297 | tilghman | 2008-12-02 11:42:09 -0600 (Tue, 02 Dec 2008) | 10 lines When the text does not match exactly (e.g. RTP/SAVP), then the %n conversion fails, and the resulting integer is garbage. Thus, we must initialize the integer and check it afterwards for success. (closes issue #14000) Reported by: folke Patches: asterisk-sipbg-sscanf-1.4.22.diff uploaded by folke (license 626) asterisk-sipbg-sscanf-1.6.0.1.diff uploaded by folke (license 626) asterisk-sipbg-sscanf-trunk-r159896.diff uploaded by folke (license 626) ........ ................ * include/asterisk/stringfields.h, apps/app_voicemail.c, main/cli.c, main/pbx.c, main/frame.c, /, channels/chan_features.c: Merged revisions 160208 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r160208 | tilghman | 2008-12-01 18:37:21 -0600 (Mon, 01 Dec 2008) | 10 lines Merged revisions 160207 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160207 | tilghman | 2008-12-01 18:25:16 -0600 (Mon, 01 Dec 2008) | 3 lines Ensure that Asterisk builds with --enable-dev-mode, even on the latest gcc and glibc. ........ ................ 2008-12-01 23:53 +0000 [r160175] Sean Bright * channels/chan_phone.c, main/manager.c, /, utils/smsq.c: Merged revisions 160170-160172 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r160170 | seanbright | 2008-12-01 18:08:24 -0500 (Mon, 01 Dec 2008) | 1 line Pay attention to the return value of system(), even if we basically ignore it. ................ r160171 | seanbright | 2008-12-01 18:18:48 -0500 (Mon, 01 Dec 2008) | 1 line Silence a build warning. (chan_phone.c:810: warning: value computed is not used) ................ r160172 | seanbright | 2008-12-01 18:37:49 -0500 (Mon, 01 Dec 2008) | 10 lines Merged revisions 159976 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159976 | mvanbaak | 2008-12-01 11:08:36 -0500 (Mon, 01 Dec 2008) | 3 lines Get rid of the useless format string and argument in the Bogus/ manager channelname. Noted by kpfleming and name Bogus/manager suggested by eliel ........ ................ 2008-12-01 Tilghman Lesher * Released 1.6.1-beta3 2008-12-01 21:46 +0000 [r160101] Tilghman Lesher * /, configure, configure.ac: Merged revisions 160097 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r160097 | tilghman | 2008-12-01 15:23:37 -0600 (Mon, 01 Dec 2008) | 2 lines Use AST_EXT_LIB_SETUP before using AST_EXT_LIB_CHECK or bad things happen. ........ 2008-12-01 17:45 +0000 [r160006] Russell Bryant * channels/chan_iax2.c, /: Merged revisions 160004 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r160004 | russell | 2008-12-01 11:34:31 -0600 (Mon, 01 Dec 2008) | 14 lines Merged revisions 160003 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160003 | russell | 2008-12-01 11:27:30 -0600 (Mon, 01 Dec 2008) | 6 lines Apply some logic used in iax2_indicate() to iax2_setoption(), as well, since they both have the potential to send control frames in the middle of call setup. We have to wait until we have received a message back from the remote end before we try to send any more frames. Otherwise, the remote end will consider it invalid, and we'll get stuck in an INVAL/VNAK storm. ........ ................ 2008-12-01 16:06 +0000 [r159975] Michiel van Baak * main/manager.c, /: Merged revisions 159898 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r159898 | mvanbaak | 2008-12-01 15:09:59 +0100 (Mon, 01 Dec 2008) | 11 lines Merged revisions 159897 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159897 | mvanbaak | 2008-12-01 15:05:41 +0100 (Mon, 01 Dec 2008) | 4 lines make manager compile on OpenBSD. The last (10th) argument to ast_channel_alloc here should be a pointer and NULL is not really a pointer. ........ ................ 2008-12-01 14:57 +0000 [r159920] Russell Bryant * .cleancount, /: Merged revisions 159911 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r159911 | russell | 2008-12-01 08:56:10 -0600 (Mon, 01 Dec 2008) | 10 lines Merged revisions 159900 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159900 | russell | 2008-12-01 08:52:56 -0600 (Mon, 01 Dec 2008) | 2 lines Force a "make clean" to avoid a bizarre build issue ... ........ ................ 2008-11-29 18:34 +0000 [r159854] Tilghman Lesher * /, apps/app_readexten.c: Merged revisions 159853 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r159853 | tilghman | 2008-11-29 12:33:18 -0600 (Sat, 29 Nov 2008) | 2 lines Allow the '#' sign to exist within an extension (inspired by issue #13330) ........ 2008-11-29 18:16 +0000 [r159851] Kevin P. Fleming * channels/chan_iax2.c, cdr/cdr_tds.c, include/asterisk/logger.h, include/asterisk/res_odbc.h, channels/chan_misdn.c, include/asterisk/astmm.h, include/asterisk/lock.h, utils/extconf.c, makeopts.in, main/dns.c, funcs/Makefile, include/asterisk/stringfields.h, include/asterisk/utils.h, include/asterisk/devicestate.h, /, include/asterisk/dundi.h, configure.ac, utils/astman.c, include/asterisk/cli.h, include/asterisk/channel.h, include/asterisk/manager.h, res/res_config_sqlite.c, utils/conf2ael.c, utils/frame.c, channels/misdn_config.c, main/ast_expr2.c, Makefile, main/srv.c, include/asterisk/compat.h, configure, channels/misdn/ie.c, include/asterisk/module.h, main/features.c, include/asterisk/linkedlists.h, main/logger.c, main/event.c, include/asterisk/dlinkedlists.h, include/asterisk/strings.h, utils/check_expr.c, channels/chan_vpb.cc, channels/chan_sip.c, main/Makefile, include/asterisk/enum.h, channels/chan_agent.c, main/utils.c, include/jitterbuf.h: Merged revisions 159818 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r159818 | kpfleming | 2008-11-29 11:57:39 -0600 (Sat, 29 Nov 2008) | 18 lines incorporates r159808 from branches/1.4: ------------------------------------------------------------------------ r159808 | kpfleming | 2008-11-29 10:58:29 -0600 (Sat, 29 Nov 2008) | 7 lines update dev-mode compiler flags to match the ones used by default on Ubuntu Intrepid, so all developers will see the same warnings and errors since this branch already had some printf format attributes, enable checking for them and tag functions that didn't have them format attributes in a consistent way ------------------------------------------------------------------------ in addition: move some format attributes from main/utils.c to the header files they belong in, and fix up references to the relevant functions based on new compiler warnings ........ 2008-11-26 19:58 +0000 [r159561] Mark Michelson * apps/app_dial.c, /: Merged revisions 159554 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r159554 | mmichelson | 2008-11-26 13:57:11 -0600 (Wed, 26 Nov 2008) | 19 lines Add some necessary hangup commands in the case that forwarding a call fails 1) Hang up the original destination if the local channel cannot be requested. 2) Hang up the local channel (in addition to the original destination) if ast_call fails when calling the newly created local channel. This prevents channels from sticking around forever in the case of a botched call forward (e.g. to an extension which does not exist). (closes issue #13764) Reported by: davidw Patches: 13764_v2.patch uploaded by putnopvut (license 60) Tested by: putnopvut, davidw ........ 2008-11-26 19:17 +0000 [r159535] Kevin P. Fleming * agi/Makefile, utils/Makefile, /, Makefile.moddir_rules, Makefile.rules: Merged revisions 159534 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r159534 | kpfleming | 2008-11-26 13:08:56 -0600 (Wed, 26 Nov 2008) | 11 lines Merged revisions 159476 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159476 | kpfleming | 2008-11-26 12:36:24 -0600 (Wed, 26 Nov 2008) | 7 lines simplify (and slightly bug-fix) the recent developer-oriented COMPILE_DOUBLE mode ensure that 'make clean' removes dependency files for .i files that are created in COMPILE_DOUBLE mode ........ ................ 2008-11-26 18:38 +0000 [r159477] Tilghman Lesher * main/udptl.c, /: Merged revisions 159475 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r159475 | tilghman | 2008-11-26 12:33:04 -0600 (Wed, 26 Nov 2008) | 7 lines If the config file does not exist, then the first use crashes Asterisk. (closes issue #13848) Reported by: klaus3000 Patches: udptl.c.patch uploaded by eliel (license 64) Tested by: blitzrage ........ 2008-11-26 14:59 +0000 [r159438] Mark Michelson * /, channels/chan_agent.c: Merged revisions 159437 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r159437 | mmichelson | 2008-11-26 08:58:17 -0600 (Wed, 26 Nov 2008) | 10 lines Don't allow for configuration options to overwrite options set via channel variables on a reload. (closes issue #13921) Reported by: davidw Patches: 13921.patch uploaded by putnopvut (license 60) Tested by: davidw ........ 2008-11-26 03:19 +0000 [r159403] Jeff Peeler * /, main/features.c: Merged revisions 159402 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r159402 | jpeeler | 2008-11-25 21:18:01 -0600 (Tue, 25 Nov 2008) | 3 lines Always parse arguments in park_call_exec so that app_args is valid. This prevents a crash when executing Park from the dialplan with no arguments. ........ 2008-11-25 23:27 +0000 [r159375] Steve Murphy * channels/chan_iax2.c, main/cdr.c, /: Merged revisions 159360 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r159360 | murf | 2008-11-25 16:03:01 -0700 (Tue, 25 Nov 2008) | 23 lines Merged revisions 159316 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159316 | murf | 2008-11-25 15:41:10 -0700 (Tue, 25 Nov 2008) | 15 lines (closes issue #12694) Reported by: yraber Patches: 12694.2nd.diff uploaded by murf (license 17) Tested by: murf, laurav Thanks to file (Joshua Colp) for his IAX fix. the change to cdr.c allows no-answer to percolate up into CDR's, and feels like the right place to locate this fix; if BUSY is done here, no-answer should be, too. ........ ................ 2008-11-25 21:58 +0000 [r159249-159280] Tilghman Lesher * channels/chan_iax2.c, /: Merged revisions 159276 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r159276 | tilghman | 2008-11-25 15:57:59 -0600 (Tue, 25 Nov 2008) | 14 lines Merged revisions 159269 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159269 | tilghman | 2008-11-25 15:56:48 -0600 (Tue, 25 Nov 2008) | 7 lines Don't try to send a response on a NULL pvt. (closes issue #13919) Reported by: barthpbx Patches: chan_iax2.c.patch uploaded by eliel (license 64) Tested by: barthpbx ........ ................ * channels/chan_iax2.c, /: Merged revisions 159247 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r159247 | tilghman | 2008-11-25 15:42:42 -0600 (Tue, 25 Nov 2008) | 21 lines Merged revisions 159246 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r159246 | tilghman | 2008-11-25 15:40:28 -0600 (Tue, 25 Nov 2008) | 14 lines Merged revisions 159245 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r159245 | tilghman | 2008-11-25 15:37:06 -0600 (Tue, 25 Nov 2008) | 7 lines Regression fix for last security fix. Set the iseqno correctly. (closes issue #13918) Reported by: ffloimair Patches: 20081119__bug13918.diff.txt uploaded by Corydon76 (license 14) Tested by: ffloimair ........ ................ ................ 2008-11-25 16:21 +0000 [r159095] Terry Wilson * /, apps/app_festival.c: Merged revisions 159093 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r159093 | twilson | 2008-11-25 10:18:53 -0600 (Tue, 25 Nov 2008) | 2 lines Add missing variable declaration for PPC code ........ 2008-11-25 05:05 +0000 [r159053] Tilghman Lesher * channels/xpmr/xpmr.c, apps/app_rpt.c, channels/chan_usbradio.c, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 159050 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r159050 | tilghman | 2008-11-24 23:02:11 -0600 (Mon, 24 Nov 2008) | 10 lines Merged revisions 159025 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159025 | tilghman | 2008-11-24 22:50:00 -0600 (Mon, 24 Nov 2008) | 3 lines System call ioperm is non-portable, so check for its existence in autoconf. (Closes issue #13863) ........ ................ 2008-11-25 03:51 +0000 [r158993] Terry Wilson * channels/chan_usbradio.c, /: Merged revisions 158992 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158992 | twilson | 2008-11-24 21:49:30 -0600 (Mon, 24 Nov 2008) | 2 lines Make chan_usbradio compile under dev mode ........ 2008-11-25 00:41 +0000 [r158894-158927] Matt Nicholson * apps/app_queue.c, /, UPGRADE.txt: Merged revisions 158924 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158924 | mnicholson | 2008-11-24 18:05:41 -0600 (Mon, 24 Nov 2008) | 6 lines Make the Join event from app_queue use CallerIDNum insead of CallerID for indicating the callerid number just like the rest of asterisk. (closes issue #13883) Reported by: davidw ........ * /, main/file.c: Merged revisions 158925 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158925 | mnicholson | 2008-11-24 18:19:55 -0600 (Mon, 24 Nov 2008) | 2 lines Fix compiling in dev mode. ........ * include/asterisk/manager.h, main/manager.c, /, res/res_agi.c: Merged revisions 158876 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158876 | mnicholson | 2008-11-24 15:56:22 -0600 (Mon, 24 Nov 2008) | 7 lines Added EVENT_FLAG_AGI and used it for manager calls in res_agi.c (closes issue #13873) Reported by: fnordian Patches: ami_agievent.patch uploaded by fnordian (license 110) ........ 2008-11-24 21:53 +0000 [r158861] Tilghman Lesher * main/dsp.c, /: Merged revisions 158857 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158857 | tilghman | 2008-11-24 15:52:34 -0600 (Mon, 24 Nov 2008) | 3 lines Add a bit of documentation (thanks, I-MOD) on what the silence threshold constant actually does and what values are valid for it. ........ 2008-11-24 21:44 +0000 [r158855] Matt Nicholson * /, main/file.c: Merged revisions 158851 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158851 | mnicholson | 2008-11-24 15:27:26 -0600 (Mon, 24 Nov 2008) | 6 lines Make ast_streamfile() check the result of ast_openstream() before doing anything with it. (closes issue #13955) Reported by: chris-mac ........ 2008-11-22 17:00 +0000 [r158689-158701] Michiel van Baak * /, channels/chan_skinny.c: Merged revisions 158694 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158694 | mvanbaak | 2008-11-22 17:57:11 +0100 (Sat, 22 Nov 2008) | 8 lines dont send reorder tone after a device is hungup if a dialout is abandoned or failed. Without this reorder tone will play after hangup and both wedhorn's and my wife have threatened to use an axe on our asterisk box (closes issue #13948) Reported by: wedhorn Patches: switch.diff uploaded by wedhorn (license 30) ........ * /, channels/chan_skinny.c: Merged revisions 158688 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158688 | mvanbaak | 2008-11-22 17:06:38 +0100 (Sat, 22 Nov 2008) | 4 lines fix a very occasional core dump in chan_skinny found by wedhorn. (issue #13948) ........ 2008-11-21 23:45 +0000 [r158607] Steve Murphy * /, main/features.c: Merged revisions 158606 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r158606 | murf | 2008-11-21 16:40:46 -0700 (Fri, 21 Nov 2008) | 19 lines Merged revisions 158603 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158603 | murf | 2008-11-21 16:14:50 -0700 (Fri, 21 Nov 2008) | 11 lines In reference to the fix made for 13871, I was merging the fix into 1.6.0 and realized I missed the code in the h-exten block, and didn't catch it because my test case had the h-exten commented out. So, this corrects the code I missed, as a preventative against another crash report. Tested with the h-exten defined, all is well. ........ ................ 2008-11-21 23:15 +0000 [r158604] Tilghman Lesher * main/pbx.c, /: Merged revisions 158602 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r158602 | tilghman | 2008-11-21 17:14:11 -0600 (Fri, 21 Nov 2008) | 12 lines Merged revisions 158600 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158600 | tilghman | 2008-11-21 17:07:46 -0600 (Fri, 21 Nov 2008) | 5 lines The passed extension may not be the same in the list as the current entry, because we strip spaces when copying the extension into the structure. Therefore, use the copied item to place the item into the list. (found by lmadsen on -dev, fixed by me) ........ ................ 2008-11-21 22:57 +0000 [r158572] Steve Murphy * /, main/features.c: Merged revisions 158484 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r158484 | murf | 2008-11-21 14:47:16 -0700 (Fri, 21 Nov 2008) | 19 lines Merged revisions 158483 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158483 | murf | 2008-11-21 14:19:47 -0700 (Fri, 21 Nov 2008) | 11 lines (closes issue #13871) Reported by: mdu113 This one is totally my fault. The code doesn't even create a bridge CDR if the channel CDR has POST_DISABLED. I didn't check for that at the end of the bridge. Fixed with a few small insertions. Tested. Looks good. No cdr generated, no crash, no unnecc. data objects created either. ........ ................ 2008-11-21 22:13 +0000 [r158541] Russell Bryant * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions 158540 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r158540 | russell | 2008-11-21 16:12:37 -0600 (Fri, 21 Nov 2008) | 10 lines Merged revisions 158539 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158539 | russell | 2008-11-21 16:05:55 -0600 (Fri, 21 Nov 2008) | 2 lines When compiling with DEBUG_THREADS, report the real file/func/line for ao2_lock/ao2_unlock ........ ................ 2008-11-21 20:43 +0000 [r158450] Kevin P. Fleming * CHANGES, /, UPGRADE-1.2.txt, UPGRADE-1.4.txt, UPGRADE.txt, UPGRADE-1.6.txt: Merged revisions 158449 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158449 | kpfleming | 2008-11-21 14:42:37 -0600 (Fri, 21 Nov 2008) | 3 lines as suggested by jtodd, document the purposes of the CHANGES and UPGRADE files ........ 2008-11-21 19:42 +0000 [r158415] Jason Parker * main/manager.c, /: Merged revisions 158414 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158414 | qwell | 2008-11-21 13:40:57 -0600 (Fri, 21 Nov 2008) | 7 lines Make sure we add the Event header for CoreShowChannels. (closes issue #13334) Reported by: srt Patches: 13334_missing_event_header_in_core_show_channel.diff uploaded by srt (license 378) ........ 2008-11-21 17:17 +0000 [r158377] Terry Wilson * cdr/cdr_csv.c, /: Merged revisions 158374 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158374 | twilson | 2008-11-21 11:08:16 -0600 (Fri, 21 Nov 2008) | 8 lines Reloading the config and having no changes still initialized some settings to 0. Initialize settings after doing all of the cfg checks. (closes issue #13942) Reported by: davidw Patches: cdr_diff.txt uploaded by otherwiseguy (license 396) Tested by: davidw ........ 2008-11-21 01:23 +0000 [r158223-158268] Mark Michelson * /, channels/chan_sip.c: Merged revisions 158265-158266 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158265 | mmichelson | 2008-11-20 19:14:20 -0600 (Thu, 20 Nov 2008) | 4 lines Use some magic constants to get the right size for this sscanf statement. Thanks Richard! ........ r158266 | mmichelson | 2008-11-20 19:22:18 -0600 (Thu, 20 Nov 2008) | 3 lines Use a more expressive constant for a 64-bit scanned int ........ * /, channels/chan_sip.c: Merged revisions 158262 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158262 | mmichelson | 2008-11-20 18:59:23 -0600 (Thu, 20 Nov 2008) | 6 lines Fix the build for 32-bit systems. %lu is only 32-bits on 32-bit systems, so we need to use %llu instead. Of course %llu is 128-bits on 64-bit systems, so we have to cast to unsigned long long. No harm, but it's sure annoying. ........ * /, channels/chan_sip.c: Merged revisions 158230 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158230 | mmichelson | 2008-11-20 17:12:50 -0600 (Thu, 20 Nov 2008) | 20 lines Change the remote user agent session version variable from an int to a uint64_t. This prevents potential comparison problems from happening if the version string exceeds INT_MAX. This was an apparent problem for one user who could not properly place a call on hold since the version in the SDP of the re-INVITE to place the call on hold greatly exceeded INT_MAX. This also aligns with RFC 2327 better since it recommends using an NTP timestamp for the version (which is a 64-bit number). (closes issue #13531) Reported by: sgofferj Patches: 13531.patch uploaded by putnopvut (license 60) Tested by: sgofferj ........ * channels/chan_sip.c: Change this so it actually compiles. Thanks, Terry! 2008-11-20 19:43 +0000 [r158191] Sean Bright * res/ael/pval.c, /: Merged revisions 158188 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158188 | seanbright | 2008-11-20 14:41:23 -0500 (Thu, 20 Nov 2008) | 10 lines Fix one case where the application argument was not converted from a pipe to a comma. This was causing problems with switch statements with empty expressions. (closes issue #13901) Reported by: smurfix Patches: 20081118_bug13901.diff uploaded by seanbright (license 71) Tested by: seanbright Reviewed by: murf ........ 2008-11-20 18:23 +0000 [r158135] Terry Wilson * cdr/cdr_odbc.c, cdr/cdr_radius.c, cdr/cdr_custom.c, cdr/cdr_manager.c, cdr/cdr_csv.c, cdr/cdr_sqlite3_custom.c, /, cdr/cdr_sqlite.c, cdr/Makefile, cdr/cdr_adaptive_odbc.c, cdr/cdr_pgsql.c: Merged revisions 158072 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158072 | twilson | 2008-11-20 11:48:58 -0600 (Thu, 20 Nov 2008) | 2 lines Begin on a crusade to end trailing whitespace! ........ 2008-11-20 18:20 +0000 [r158084-158134] Mark Michelson * include/asterisk/frame.h, include/asterisk/file.h, main/frame.c, /, channels/chan_sip.c, main/file.c: Merged revisions 158133 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r158133 | mmichelson | 2008-11-20 12:20:00 -0600 (Thu, 20 Nov 2008) | 10 lines Merged revisions 158072 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158072 | twilson | 2008-11-20 11:48:58 -0600 (Thu, 20 Nov 2008) | 2 lines Begin on a crusade to end trailing whitespace! ........ ................ * /, channels/chan_sip.c: Merged revisions 158082 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r158082 | mmichelson | 2008-11-20 11:54:31 -0600 (Thu, 20 Nov 2008) | 24 lines Merged revisions 158071 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158071 | mmichelson | 2008-11-20 11:48:42 -0600 (Thu, 20 Nov 2008) | 16 lines We don't handle 4XX responses to BYE well. According to section 15 of RFC 3261, we should terminate a dialog if we receive a 481 or 408 in response to our BYE. Since I am aware of at least one phone manufacturer who may sometimes send a 404 as well, I am being liberal and saying that any 4XX response to a BYE should result in a terminated dialog. (closes issue #12994) Reported by: pabelanger Patches: 12994.patch uploaded by putnopvut (license 60) Closes AST-129 ........ ................ 2008-11-20 17:42 +0000 [r158069] Jeff Peeler * /, main/file.c: Merged revisions 158062 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158062 | jpeeler | 2008-11-20 11:37:31 -0600 (Thu, 20 Nov 2008) | 6 lines (closes issue #12929) Reported by: snyfer This handles the case for a zero length file to attempt to be streamed. Instead of failing from not playing any data, go ahead and return success as ast_streamfile should consider playing nothing a success when there is nothing to play. ........ 2008-11-20 17:40 +0000 [r158067] Mark Michelson * apps/app_dial.c, /, channels/chan_sip.c: Merged revisions 158066 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r158066 | mmichelson | 2008-11-20 11:39:06 -0600 (Thu, 20 Nov 2008) | 20 lines Merged revisions 158053 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov 2008) | 12 lines Make sure to set the hangup cause on the calling channel in the case that ast_call() fails. For incoming SIP channels, this was causing us to send a 603 instead of a 486 when the call-limit was reached on the destination channel. (closes issue #13867) Reported by: still_nsk Patches: 13867.diff uploaded by putnopvut (license 60) Tested by: blitzrage ........ ................ 2008-11-20 00:10 +0000 [r157975] Kevin P. Fleming * main/stdtime/Makefile, codecs/gsm/src, main/db1-ast/btree, channels/misdn/Makefile, main/db1-ast/recno, pbx/ael, res/ael, channels, main/db1-ast/Makefile, main/stdtime, main/db1-ast/hash, codecs/gsm/Makefile, main/db1-ast/db, Makefile.moddir_rules, main/db1-ast/mpool, res/ais, channels/misdn, res/snmp, Makefile.rules, pbx/Makefile, res/Makefile: Merged revisions 157974 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r157974 | kpfleming | 2008-11-19 18:08:12 -0600 (Wed, 19 Nov 2008) | 13 lines Merged revisions 157859 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157859 | kpfleming | 2008-11-19 15:34:47 -0600 (Wed, 19 Nov 2008) | 7 lines the gcc optimizer frequently finds broken code (use of uninitalized variables, unreachable code, etc.), which is good. however, developers usually compile with the optimizer turned off, because if they need to debug the resulting code, optimized code makes that process very difficult. this means that we get code changes committed that weren't adequately checked over for these sorts of problems. with this build system change, if (and only if) --enable-dev-mode was used and DONT_OPTIMIZE is turned on, when a source file is compiled it will actually be preprocessed (into a .i or .ii file), then compiled once with optimization (with the result sent to /dev/null) and again without optimization (but only if the first compile succeeded, of course). while making these changes, i did some cleanup work in Makefile.rules to move commonly-used combinations of flag variables into their own variables, to make the file easier to read and maintain ........ ................ 2008-11-19 18:29 +0000 [r157785] Tilghman Lesher * /, configure, configure.ac: Merged revisions 157784 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157784 | tilghman | 2008-11-19 12:28:14 -0600 (Wed, 19 Nov 2008) | 6 lines Add check for t38_terminal_init in spandsp (not found in 0.0.6, so it should fail reasonably) (closes issue #13473) Reported by: genie Patches: 20080916__bug13473.diff.txt uploaded by Corydon76 (license 14) ........ 2008-11-19 13:47 +0000 [r157719-157744] Kevin P. Fleming * /, res/res_agi.c: Merged revisions 157743 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157743 | kpfleming | 2008-11-19 07:45:48 -0600 (Wed, 19 Nov 2008) | 1 line correct small bug introduced during API conversion ........ * CHANGES, apps/app_stack.c, include/asterisk/agi.h, /, res/res_agi.c, UPGRADE.txt, UPGRADE-1.6.txt (added): Merged revisions 157706 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157706 | kpfleming | 2008-11-19 06:42:19 -0600 (Wed, 19 Nov 2008) | 5 lines make some corrections to the ast_agi_register_multiple(), ast_agi_unregister_multiple() and ast_agi_fdprintf() API calls to be consistent with API guidelines also, move UPGRADE.txt to UPGRADE-1.6.txt and make the new UPGRADE.txt contain information about upgrading between Asterisk 1.6 releases ........ 2008-11-19 01:08 +0000 [r157641] Tilghman Lesher * include/asterisk/logger.h, /, main/logger.c, main/utils.c, include/asterisk/strings.h: Merged revisions 157639 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157639 | tilghman | 2008-11-18 19:02:45 -0600 (Tue, 18 Nov 2008) | 7 lines Starting with a change to ensure that ast_verbose() preserves ABI compatibility in 1.6.1 (as compared to 1.6.0 and versions of 1.4), this change also deprecates the use of Asterisk with FreeBSD 4, given the central use of va_copy in core functions. va_copy() is C99, anyway, and we already require C99 for other purposes, so this isn't really a big change anyway. This change also simplifies some of the core ast_str_* functions. ........ 2008-11-19 01:00 +0000 [r157636] Mark Michelson * /, main/astmm.c: Merged revisions 157632 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157632 | mmichelson | 2008-11-18 18:59:48 -0600 (Tue, 18 Nov 2008) | 10 lines If malloc returns NULL, we need to return NULL immediately or else Asterisk will crash when attempting to dereference the NULL pointer (closes issue #13858) Reported by: eliel Patches: astmm.c.patch uploaded by eliel (license 64) ........ 2008-11-19 00:38 +0000 [r157602] Sean Bright * build_tools/make_buildopts_h, makeopts.in, Makefile, /, build_tools/make_version, configure, configure.ac: Merged revisions 157600 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157600 | seanbright | 2008-11-18 19:27:45 -0500 (Tue, 18 Nov 2008) | 10 lines Fix a few build problems on Solaris (and check for an md5 utility in configure instead of the icky loop I was doing before). (closes issue #13842) Reported by: snuffy Patches: bug13842_20081106.diff uploaded by snuffy (license 35) 13842.diff uploaded by seanbright (license 71) Tested by: snuffy ........ 2008-11-18 23:59 +0000 [r157429-157596] Mark Michelson * /, res/res_musiconhold.c: Merged revisions 157592 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157592 | mmichelson | 2008-11-18 17:59:02 -0600 (Tue, 18 Nov 2008) | 10 lines This change prevents a crash from occurring if res_musiconhold.so is unloaded and then Asterisk is stopped. The problem was that we are not unregistering the ast_moh_destroy function at exit. (closes issue #13761) Reported by: eliel Patches: res_musiconhold.c.patch uploaded by eliel (license 64) ........ * apps/app_voicemail.c, /: Merged revisions 157562 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157562 | mmichelson | 2008-11-18 17:28:23 -0600 (Tue, 18 Nov 2008) | 11 lines Fix the logic for when delete=yes when IMAP storage is in use so that the message is deleted from both local and IMAP storage. (closes issue #13642) Reported by: jaroth Patches: deleteyes.patch uploaded by jaroth (license 50) ........ * /, channels/chan_sip.c: Merged revisions 157512 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r157512 | mmichelson | 2008-11-18 16:54:08 -0600 (Tue, 18 Nov 2008) | 21 lines Merged revisions 157503 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157503 | mmichelson | 2008-11-18 16:47:57 -0600 (Tue, 18 Nov 2008) | 13 lines Add some missing invite state changes necessary in the sip_write function. Not setting the invite state correctly on the call was resulting in the Record application leaving empty files. I also have updated the doxygen comment next to the declaration of the INV_EARLY_MEDIA constant to reflect that we also use this state when we *send* a 18X response to an INVITE. (closes issue #13878) Reported by: nahuelgreco Patches: sip-early-media-recording-1.4.22.patch uploaded by nahuelgreco (license 162) Tested by: putnopvut ........ ................ * /, channels/chan_sip.c: Merged revisions 157496 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157496 | mmichelson | 2008-11-18 15:59:24 -0600 (Tue, 18 Nov 2008) | 6 lines Based on Russell's advice on the asterisk-dev list, I have changed from using a global lock in update_call_counter to using the locks within the sip_pvt and sip_peer structures instead. ........ * /, channels/chan_sip.c: Merged revisions 157427 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157427 | mmichelson | 2008-11-18 14:23:58 -0600 (Tue, 18 Nov 2008) | 13 lines * Add a lock to be used in the update_call_counter function. * Revert logic to mirror 1.4's in the sense that it will not allow the call counter to dip below 0. These two measures prevent potential races that could cause a SIP peer to appear to be busy forever. (closes issue #13668) Reported by: mjc Patches: hintfix_trunk_rev152649.patch uploaded by wolfelectronic (license 586) ........ 2008-11-18 19:18 +0000 [r157367] Jeff Peeler * /, apps/app_meetme.c: Merged revisions 157366 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r157366 | jpeeler | 2008-11-18 13:16:00 -0600 (Tue, 18 Nov 2008) | 14 lines Merged revisions 157365 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157365 | jpeeler | 2008-11-18 13:13:33 -0600 (Tue, 18 Nov 2008) | 6 lines (closes issue #13899) Reported by: akkornel This fix is the result of a bug fix in ast_app_separate_args r124395. If an argument does not exist it should always be set to a null string rather than a null pointer. ........ ................ 2008-11-18 18:32 +0000 [r157308] Mark Michelson * apps/app_followme.c, apps/app_dial.c, channels/chan_local.c, /, main/features.c, include/asterisk/channel.h: Merged revisions 157306 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r157306 | mmichelson | 2008-11-18 12:31:08 -0600 (Tue, 18 Nov 2008) | 20 lines Merged revisions 157305 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157305 | mmichelson | 2008-11-18 12:25:55 -0600 (Tue, 18 Nov 2008) | 12 lines Fix a crash in the end_bridge_callback of app_dial and app_followme which would occur at the end of an attended transfer. The error occurred because we initially stored a pointer to an ast_channel which then was hung up due to a masquerade. This commit adds a "fixup" callback to the bridge_config structure to allow for end_bridge_callback_data to be changed in the case that a new channel pointer is needed for the end_bridge_callback. ........ ................ 2008-11-18 18:20 +0000 [r157304] Steve Murphy * main/config.c, /: Merged revisions 157302 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157302 | murf | 2008-11-18 11:07:55 -0700 (Tue, 18 Nov 2008) | 18 lines (closes issue #13420) Reported by: alex70 Patches: 13420.13539.patch uploaded by murf (license 17) Tested by: murf, awk This fixes two problems: a spurious linefeed insertion probably left over from pre-precomment times. Only generated when category had no previous comments. The other problem: Insertions could get the line-numbering out of whack and generate negative line numbers, causing chunks of line numbers to be emitted, on the scale of the number of lines up to that point in the file. In such cases, abort the looping, and all is well. ........ 2008-11-17 22:39 +0000 [r157255] Tilghman Lesher * apps/app_dial.c, /: Merged revisions 157253 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157253 | tilghman | 2008-11-17 16:25:06 -0600 (Mon, 17 Nov 2008) | 8 lines Can't use items duplicated off the stack frame in an element returned from a function: in these cases, we have to use the heap, or garbage will result. (closes issue #13898) Reported by: alecdavis Patches: 20081114__bug13898__2.diff.txt uploaded by Corydon76 (license 14) Tested by: alecdavis ........ 2008-11-15 19:49 +0000 [r157108-157166] Kevin P. Fleming * Makefile, /, Makefile.moddir_rules, Makefile.rules: Merged revisions 157164 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r157164 | kpfleming | 2008-11-15 20:45:19 +0100 (Sat, 15 Nov 2008) | 13 lines Merged revisions 157162-157163 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157162 | kpfleming | 2008-11-15 20:24:24 +0100 (Sat, 15 Nov 2008) | 1 line dist-clean should remove dependency information files as well ........ r157163 | kpfleming | 2008-11-15 20:31:03 +0100 (Sat, 15 Nov 2008) | 1 line when an individual directory dist-clean is run, run clean in that directory first, and when running top-level dist-clean, do not run subdirectory clean operations twice ........ ................ * /, contrib/asterisk-ng-doxygen: Merged revisions 157105 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157105 | kpfleming | 2008-11-15 19:00:32 +0100 (Sat, 15 Nov 2008) | 13 lines major update to doxygen configuration file: 1) update to doxygen 1.5.x style file, as used in trunk 2) tell doxygen where are header files are, so include-file processing can be done 3) make all macros that are used to define variables/functions be expanded, so that doxygen will properly document the resulting variable/function 4) make all macros that are used to provide the contents of a variable (structure) be expanded, so that doxygen will be able to document the resulting fields 5) suppress compiler attributes (__attribute__(xxx)) from being seen by doxygen, so it will properly match up function definition and usage (for an example of th effect of this, look at the doxygen docs for ast_log() from before and afte this commit) ........ 2008-11-15 04:30 +0000 [r157040-157042] Russell Bryant * /, channels/chan_sip.c, main/features.c, main/taskprocessor.c: Merged revisions 157041 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157041 | russell | 2008-11-14 22:25:57 -0600 (Fri, 14 Nov 2008) | 3 lines Fix a few more places where the case insensitive hash should be used since the comparison is case insensitive. ........ * /, channels/chan_console.c: Merged revisions 157039 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157039 | russell | 2008-11-14 22:08:42 -0600 (Fri, 14 Nov 2008) | 3 lines Use the new case insensitive hash function for console interfaces. The comparison function is case insensitive. ........ 2008-11-14 21:21 +0000 [r156963] Mark Michelson * /, channels/chan_sip.c: Merged revisions 156962 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r156962 | mmichelson | 2008-11-14 15:19:58 -0600 (Fri, 14 Nov 2008) | 7 lines Revision 155513 of chan_sip.c in trunk inadvertently removed a very important line to set the "len" field for incoming SIP requests. The result was that all incoming SIP messages appeared to be 0-length, meaning Asterisk could do no meaningful processing of anything SIP-related ........ 2008-11-14 17:04 +0000 [r156913] Tilghman Lesher * main/manager.c, /: Merged revisions 156911 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r156911 | tilghman | 2008-11-14 11:02:00 -0600 (Fri, 14 Nov 2008) | 4 lines Ping is missing the standard double-newline after the event. (closes issue #13903) Reported by: kebl0155 ........ 2008-11-14 16:57 +0000 [r156819-156894] Mark Michelson * apps/app_queue.c, include/asterisk/strings.h: This is the 1.6.1 version of trunk commit 156883. It is functionally equivalent to the 1.6.0 commit * apps/app_voicemail.c, /: Merged revisions 156817 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156817 | mmichelson | 2008-11-14 09:20:03 -0600 (Fri, 14 Nov 2008) | 18 lines Merged revisions 156816 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156816 | mmichelson | 2008-11-14 09:18:59 -0600 (Fri, 14 Nov 2008) | 10 lines If the prompt to reenter a voicemail password timed out, it resulted in the password not being saved, even if the input matched what you gave when first prompted to enter a new password. This is because the return value of ast_readstring was checked, but not checked properly. This bug was discovered by Jared Smith during an Asterisk training course. Thanks for reporting it! ........ ................ 2008-11-14 00:44 +0000 [r156691-156757] Tilghman Lesher * apps/app_while.c, /: Merged revisions 156756 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156756 | tilghman | 2008-11-13 18:43:13 -0600 (Thu, 13 Nov 2008) | 13 lines Merged revisions 156755 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156755 | tilghman | 2008-11-13 18:41:37 -0600 (Thu, 13 Nov 2008) | 6 lines ast_waitfordigit() requires that the channel be up, for no good logical reason. This prevents While/EndWhile from working within the "h" extension. Reported by: jgalarneau (for ABE C.2) Fixed by: me ........ ................ * main/manager.c, /: Merged revisions 156690 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156690 | tilghman | 2008-11-13 15:30:41 -0600 (Thu, 13 Nov 2008) | 14 lines Merged revisions 156688 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156688 | tilghman | 2008-11-13 15:24:00 -0600 (Thu, 13 Nov 2008) | 7 lines Provide more space for all the data which can appear in an originating channel name. (closes issue #13398) Reported by: bamby Patches: manager.c.diff uploaded by bamby (license 430) ........ ................ 2008-11-13 19:29 +0000 [r156654] Brandon Kruse * main/manager.c: Merged revisions 156017 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r156017 | pari | 2008-11-11 17:02:43 -0600 (Tue, 11 Nov 2008) | 5 lines Patch by Ryan Brindley -- Make sure that manager refuses any duplicate 'new category' requests in updateconfig (closes issue #13539) ........ 2008-11-13 19:18 +0000 [r156650] Jeff Peeler * main/pbx.c, /: Merged revisions 156649 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r156649 | jpeeler | 2008-11-13 13:17:50 -0600 (Thu, 13 Nov 2008) | 6 lines (closes issue #13891) Reported by: smurfix This reverts a change I made in 116297. At the time it seemed the change was required to solve an issue with attempting a transfer but then letting it timeout without dialing any digits. However, I didn't realize that having an empty extension was possible. I'm removing the immediate return that was added in pbx_find_extension if the extension is null. ........ 2008-11-13 17:12 +0000 [r156614] Mark Michelson * autoconf/ast_c_compile_check.m4, /, configure: Merged revisions 156612 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r156612 | mmichelson | 2008-11-13 11:07:56 -0600 (Thu, 13 Nov 2008) | 4 lines Kevin sent a note indicating that this change is not necessary, so I am reverting it ........ 2008-11-12 21:36 +0000 [r156389] Tilghman Lesher * apps/app_dial.c, /: Merged revisions 156388 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156388 | tilghman | 2008-11-12 15:34:51 -0600 (Wed, 12 Nov 2008) | 12 lines Merged revisions 156386 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156386 | tilghman | 2008-11-12 15:18:57 -0600 (Wed, 12 Nov 2008) | 5 lines When using call limits under 1 second, infinite call lengths are allowed, instead. (closes issue #13851) Reported by: ruddy ........ ................ 2008-11-12 20:11 +0000 [r156354] Steve Murphy * main/pbx.c, /: Merged revisions 156299 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156299 | murf | 2008-11-12 12:47:29 -0700 (Wed, 12 Nov 2008) | 26 lines Merged revisions 156297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156297 | murf | 2008-11-12 12:36:16 -0700 (Wed, 12 Nov 2008) | 18 lines It turns out that the 0x0XX00 codes being returned for N, X, and Z are off by one, as per conversation with jsmith on #asterisk-dev; he was teaching a class and disconcerted that this published rule was not being followed, with patterns _NXX, _[1-8]22 and _[2-9]22... and NXX was winning, but [1-8] should have been. This change, tested on these 3 patterns now picks the proper one. However, this change may surprise users who set up dialplans based on previous behavior, which has been there for what, 2 and half years or so now. ........ ................ 2008-11-12 19:29 +0000 [r156296] Tilghman Lesher * /, apps/app_meetme.c: Merged revisions 156295 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156295 | tilghman | 2008-11-12 13:28:22 -0600 (Wed, 12 Nov 2008) | 13 lines Merged revisions 156294 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156294 | tilghman | 2008-11-12 13:26:45 -0600 (Wed, 12 Nov 2008) | 6 lines If the SLA thread is not started, then reload causes a memory leak. (closes issue #13889) Reported by: eliel Patches: app_meetme.c.patch uploaded by eliel (license 64) ........ ................ 2008-11-12 19:11 +0000 [r156291] Jeff Peeler * /, apps/app_meetme.c: Merged revisions 156290 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156290 | jpeeler | 2008-11-12 13:11:15 -0600 (Wed, 12 Nov 2008) | 11 lines Merged revisions 156289 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156289 | jpeeler | 2008-11-12 13:10:12 -0600 (Wed, 12 Nov 2008) | 3 lines For whatever reason, gcc only warned me about the possible use of an uninitialized variable when compiling 1.6.1. ........ ................ 2008-11-12 19:05 +0000 [r156284-156288] Tilghman Lesher * channels/chan_iax2.c, /: Merged revisions 156243 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156243 | tilghman | 2008-11-12 12:55:18 -0600 (Wed, 12 Nov 2008) | 18 lines Merged revisions 156229 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156229 | tilghman | 2008-11-12 12:39:21 -0600 (Wed, 12 Nov 2008) | 11 lines Revert revision 132506, since it occasionally caused IAX2 HANGUP packets not to be sent, and instead, schedule a task to destroy the iax2 pvt structure 10 seconds later. This allows the IAX2 HANGUP packet to be queued, transmitted, and ACKed before the pvt is destroyed. (closes issue #13645) Reported by: dzajro Patches: 20081111__bug13645__3.diff.txt uploaded by Corydon76 (license 14) Tested by: vazir Reviewed: http://reviewboard.digium.com/r/51/ ........ ................ * apps/app_meetme.c: Fix build (res possibly unused in this function, says gcc) 2008-11-12 18:55 +0000 [r156247] Jeff Peeler * /, apps/app_meetme.c: Merged revisions 156228 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156228 | jpeeler | 2008-11-12 12:32:46 -0600 (Wed, 12 Nov 2008) | 16 lines Merged revisions 156178 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156178 | jpeeler | 2008-11-12 11:53:44 -0600 (Wed, 12 Nov 2008) | 8 lines (closes issue #13173) Reported by: pep This change adds an announce_thread responsible for playing announcements to an existing conference. This allows all announcing to be immediately stopped if necessary but more importantly allows other threads that need to play something to not block. There are multiple benefits to this, but the actual bug is for solving the scenario for a channel to be unusable after hang up for the entire duration of the parting announcement. The parting announcement can be extremely long depending on what the user recorded upon joining the conference. Reviewed by Russell on Review Board: http://reviewboard.digium.com/r/25/ ........ ................ 2008-11-12 17:48 +0000 [r156171] Mark Michelson * apps/app_dial.c, /: Merged revisions 156169 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156169 | mmichelson | 2008-11-12 11:41:56 -0600 (Wed, 12 Nov 2008) | 15 lines Merged revisions 156167 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156167 | mmichelson | 2008-11-12 11:38:33 -0600 (Wed, 12 Nov 2008) | 7 lines When doing some tests, I was having a crash at the end of every call if an attended transfer occurred during the call. I traced the cause to the CDR on one of the channels being NULL. murf suggested a check in the end bridge callback to be sure the CDR is non-NULL before proceeding, so that's what I'm adding. ........ ................ 2008-11-12 17:38 +0000 [r156168] Russell Bryant * main/asterisk.c, /: Merged revisions 156166 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r156166 | russell | 2008-11-12 11:38:20 -0600 (Wed, 12 Nov 2008) | 15 lines Merged revisions 156164 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156164 | russell | 2008-11-12 11:29:52 -0600 (Wed, 12 Nov 2008) | 7 lines Move the sanity check that makes sure "always fork" is not set along with the console option to be after the code that reads options from asterisk.conf. This resolves a situation where Asterisk can start taking up 100% when misconfigured. (Thanks to Bryce Porter (x86 on IRC) for letting me log in to his system to figure out what was causing the 100% CPU problem.) ........ ................ 2008-11-12 15:34 +0000 [r156128] Mark Michelson * autoconf/ast_c_compile_check.m4, /, configure: Merged revisions 156127 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r156127 | mmichelson | 2008-11-12 09:33:11 -0600 (Wed, 12 Nov 2008) | 5 lines Add a couple of AC_SUBST calls to the AST_C_COMPILE_CHECK macro. These missing calls were discovered when working on timerfd support in a separate branch. ........ 2008-11-11 19:52 +0000 [r156005] Tilghman Lesher * /, res/res_realtime.c: Merged revisions 155862 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r155862 | tilghman | 2008-11-10 15:12:28 -0600 (Mon, 10 Nov 2008) | 5 lines Make documentation of update method match documentation and update update2 method to match. Reported by: atis, via -dev mailing list. Fixed by: me ........ 2008-11-10 21:15 +0000 [r155864] Mark Michelson * /, channels/chan_agent.c: Merged revisions 155863 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r155863 | mmichelson | 2008-11-10 15:14:44 -0600 (Mon, 10 Nov 2008) | 22 lines Merged revisions 155861 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155861 | mmichelson | 2008-11-10 15:07:39 -0600 (Mon, 10 Nov 2008) | 14 lines Channel drivers assume that when their indicate callback is invoked, that the channel on which the callback was called is locked. This patch corrects an instance in chan_agent where a channel's indicate callback is called directly without first locking the channel. This was leading to some observed locking issues in chan_local, but considering that all channel drivers operate under the same expectations, the generic fix in chan_agent is the right way to go. AST-126 ........ ................ 2008-11-10 20:56 +0000 [r155764-155826] Tilghman Lesher * doc/valgrind.txt, /: Merged revisions 155804 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r155803 | tilghman | 2008-11-10 14:49:59 -0600 (Mon, 10 Nov 2008) | 1 line I got tired of saying this in every single bugnote referring to this file. ........ * /, main/editline/readline.c: Merged revisions 155763 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r155763 | tilghman | 2008-11-10 12:04:30 -0600 (Mon, 10 Nov 2008) | 6 lines Fix memory leak when MALLOC_DEBUG is enabled. (closes issue #13864) Reported by: eliel Patches: readline.c.patch uploaded by eliel (license 64) ........ 2008-11-09 16:32 +0000 [r155556-155672] Sean Bright * configs/chan_dahdi.conf.sample, /: Merged revisions 155671 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r155671 | seanbright | 2008-11-09 11:30:29 -0500 (Sun, 09 Nov 2008) | 1 line Fix this as well. Pointed out by tzafrir. ........ * apps/app_followme.c, apps/app_queue.c, apps/app_dial.c, /, main/features.c, include/asterisk/channel.h: Merged revisions 155554 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r155554 | seanbright | 2008-11-08 20:27:00 -0500 (Sat, 08 Nov 2008) | 14 lines Merged revisions 155553 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155553 | seanbright | 2008-11-08 20:08:07 -0500 (Sat, 08 Nov 2008) | 6 lines Use static functions here instead of nested ones. This requires a small change to the ast_bridge_config struct as well. To understand the reason for this change, see the following post: http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html ........ ................ 2008-11-08 21:48 +0000 [r155515-155517] Russell Bryant * /, channels/chan_sip.c, include/asterisk/strings.h: Merged revisions 155516 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r155516 | russell | 2008-11-08 15:46:43 -0600 (Sat, 08 Nov 2008) | 3 lines - Check for failure when putting the packet in the ast_str - fix a spelling error in a header file ........ * /, channels/chan_sip.c: Merged revisions 155513 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r155513 | russell | 2008-11-08 15:34:36 -0600 (Sat, 08 Nov 2008) | 3 lines Remove some code that is basically a no-op. Code above this already ensures that the buffer is terminated. ........ 2008-11-07 23:42 +0000 [r155469] Mark Michelson * /, channels/chan_sip.c: Merged revisions 155467 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r155467 | mmichelson | 2008-11-07 17:41:44 -0600 (Fri, 07 Nov 2008) | 12 lines Set the invite state to INV_CANCELLED in a place that makes more sense. Where it was set before, it was impossible to actually delay sending a CANCEL if we had not yet received a provisional response to an INVITE. (closes issue #13626) Reported by: atis Patches: 13626.patch uploaded by putnopvut (license 60) Tested by: atis ........ 2008-11-07 22:29 +0000 [r155396-155400] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 155399 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r155399 | tilghman | 2008-11-07 16:28:58 -0600 (Fri, 07 Nov 2008) | 14 lines Merged revisions 155398 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155398 | tilghman | 2008-11-07 16:27:32 -0600 (Fri, 07 Nov 2008) | 7 lines Clarify error message. (closes issue #13809) Reported by: denke Patches: 20081104__bug13809.diff.txt uploaded by Corydon76 (license 14) Tested by: denke ........ ................ * /, funcs/func_odbc.c: Merged revisions 155395 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r155395 | tilghman | 2008-11-07 16:03:50 -0600 (Fri, 07 Nov 2008) | 2 lines Two bugs relating to colnames found by Marquis42 on #asterisk-dev ........ 2008-11-07 21:16 +0000 [r155362] Mark Michelson * /, configs/voicemail.conf.sample: Merged revisions 155360 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r155360 | mmichelson | 2008-11-07 15:14:49 -0600 (Fri, 07 Nov 2008) | 8 lines Remove one more instance of the sample configuration lying about what's possible. The tz cannot be set in a context like this. It can only be set in the general section or per-mailbox. Thanks to sasargen on #asterisk-dev for pointing this out ........ 2008-11-07 20:19 +0000 [r155325] Tilghman Lesher * channels/chan_dahdi.c, /: Merged revisions 155324 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r155324 | tilghman | 2008-11-07 14:13:32 -0600 (Fri, 07 Nov 2008) | 7 lines Send call release with unallocated cause instead of normal call clearing, when invalid extension is called. (closes issue #13408) Reported by: adomjan Patches: chan_dahdi.c-ss7-unallocated-2 uploaded by adomjan (license 487) ........ 2008-11-07 15:43 +0000 [r155242-155272] Russell Bryant * /, channels/chan_sip.c: Merged revisions 155264 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r155264 | russell | 2008-11-07 09:42:04 -0600 (Fri, 07 Nov 2008) | 3 lines Remove a bogus ast_free() that Kevin noticed. This was probably just left over from pre-astobj2ified chan_sip. ........ * /, include/asterisk/astobj2.h: Merged revisions 155244 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r155244 | russell | 2008-11-07 09:01:02 -0600 (Fri, 07 Nov 2008) | 4 lines Clarify which part of OBJ_MULTIPLE is not implemented, and under what case it is perfectly fine to use. (Inspired by a question I received about my last commit.) ........ * /, channels/chan_sip.c: Merged revisions 155241 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r155241 | russell | 2008-11-07 08:50:30 -0600 (Fri, 07 Nov 2008) | 4 lines Fix some code in chan_sip that was intended to unlink multiple objects from a container. The OBJ_MULTIPLE flag must be provided here. Otherwise, this would only remove a single object. ........ 2008-11-06 22:49 +0000 [r155117-155122] Kevin P. Fleming * res/ael/ael.flex, /, res/ael/ael_lex.c, utils/extconf.c: Merged revisions 155121 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r155121 | kpfleming | 2008-11-06 16:49:19 -0600 (Thu, 06 Nov 2008) | 3 lines don't blindly assume that Darwin and Cygwin need GLOB_ABORTED defined; only define it if it is not already defined ........ * configure, configure.ac: ensure that an adequately new version of libpri is in place so that chan_dahdi will compile with PRI support 2008-11-06 19:48 +0000 [r155014] Mark Michelson * /, configs/voicemail.conf.sample: Merged revisions 155012 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r155012 | mmichelson | 2008-11-06 13:46:53 -0600 (Thu, 06 Nov 2008) | 16 lines Merged revisions 155011 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155011 | mmichelson | 2008-11-06 13:45:52 -0600 (Thu, 06 Nov 2008) | 8 lines The documentation listed the ability to set 'maxmsg' per context. The truth is that you can only set this in the general section or per mailbox. Thus I am updating the sample config file to be more accurate. Thanks to sasargen on IRC for bringing up this issue. ........ ................ 2008-11-05 22:02 +0000 [r154920] Sean Bright * include/asterisk.h, /: Merged revisions 154919 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r154919 | seanbright | 2008-11-05 17:01:22 -0500 (Wed, 05 Nov 2008) | 2 lines Fix a problem found while building res_snmp. ........ 2008-11-05 22:00 +0000 [r154917] Tilghman Lesher * channels/chan_iax2.c, /: Merged revisions 154428 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r154428 | tilghman | 2008-11-04 17:03:00 -0600 (Tue, 04 Nov 2008) | 7 lines Switch to using a thread condition to signal that a child thread is ready for work, rather than a busy wait. (closes issue #13011) Reported by: jpgrayson Patches: chan_iax2_find_idle.patch uploaded by jpgrayson (license 492) ........ 2008-11-05 16:14 +0000 [r154690] Steve Murphy * main/channel.c, /: Merged revisions 154687 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r154687 | murf | 2008-11-05 09:11:11 -0700 (Wed, 05 Nov 2008) | 9 lines Merged revisions 154685 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154685 | murf | 2008-11-05 09:06:53 -0700 (Wed, 05 Nov 2008) | 1 line This fix was prompted by communication from user, who was seeing thousands of error logs... looks like EAGAIN. Made such uninteresting. ........ ................ 2008-11-04 20:52 +0000 [r154367] Tilghman Lesher * channels/chan_iax2.c, /: Merged revisions 154366 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r154366 | tilghman | 2008-11-04 14:51:18 -0600 (Tue, 04 Nov 2008) | 16 lines Merged revisions 154365 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154365 | tilghman | 2008-11-04 14:49:33 -0600 (Tue, 04 Nov 2008) | 9 lines On busy systems, it's possible for the values checked within a single line of code to change, unless the structure is locked to ensure a consistent state. (closes issue #13717) Reported by: kowalma Patches: 20081102__bug13717.diff.txt uploaded by Corydon76 (license 14) Tested by: kowalma ........ ................ 2008-11-04 19:09 +0000 [r154269] Richard Mudgett * channels/chan_misdn.c, /: Merged revisions 154268 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r154268 | rmudgett | 2008-11-04 13:07:26 -0600 (Tue, 04 Nov 2008) | 11 lines Merged revisions 154266 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154266 | rmudgett | 2008-11-04 13:01:08 -0600 (Tue, 04 Nov 2008) | 4 lines JIRA ABE-1703 mISDN sets the channel to the wrong state when it receives the indication AST_CONTROL_RINGING. ........ ................ 2008-11-04 19:02 +0000 [r154024-154267] Tilghman Lesher * /, channels/chan_h323.c: Merged revisions 154264 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r154264 | tilghman | 2008-11-04 12:59:48 -0600 (Tue, 04 Nov 2008) | 10 lines Recorded merge of revisions 154263 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154263 | tilghman | 2008-11-04 12:58:05 -0600 (Tue, 04 Nov 2008) | 3 lines Make the monitor thread non-detached, so it can be joined (suggested by Russell on -dev list). ........ ................ * apps/app_voicemail.c, /: Recorded merge of revisions 154072 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r154072 | tilghman | 2008-11-03 16:28:12 -0600 (Mon, 03 Nov 2008) | 12 lines Merged revisions 154066 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154066 | tilghman | 2008-11-03 16:27:10 -0600 (Mon, 03 Nov 2008) | 5 lines Attempting to expunge a mailbox when the mailstream is NULL will crash Asterisk. (Closes issue #13829) Reported by: jaroth Patch by: me (modified jaroth's patch) ........ ................ * main/rtp.c, /: Merged revisions 154060 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154060 | tilghman | 2008-11-03 15:48:21 -0600 (Mon, 03 Nov 2008) | 3 lines Remove the potential for a division by zero error. (Closes issue #13810) ........ * /, funcs/func_odbc.c: Recorded merge of revisions 154023 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r154023 | tilghman | 2008-11-03 15:01:30 -0600 (Mon, 03 Nov 2008) | 4 lines Should have passed the string pointer, not the ast_str structure. (closes issue #13830) Reported by: Marquis ........ 2008-11-03 00:21 +0000 [r153710-153711] Kevin P. Fleming * include/asterisk/compiler.h, apps/app_stack.c, include/asterisk/agi.h, configure, include/asterisk/autoconfig.h.in, autoconf/ast_gcc_attribute.m4, configure.ac: Merged revision 153709 from trunk ------------------------------------------------------------------------ r153709 | kpfleming | 2008-11-02 17:34:39 -0600 (Sun, 02 Nov 2008) | 3 lines instead of trying to forcibly load res_agi when app_stack is loaded (even if the administrator didn't want it loaded), use GCC weak symbols to determine whether it was loaded already or not; if it was loaded, then use it. ------------------------------------------------------------------------ * channels/chan_iax2.c, res/res_jabber.c, channels/chan_oss.c, utils/stereorize.c, main/channel.c, main/manager.c, res/ael/ael_lex.c, main/file.c, pbx/pbx_dundi.c, formats/format_gsm.c, main/asterisk.c, utils/muted.c, /, formats/format_wav.c, apps/app_authenticate.c, res/res_phoneprov.c, res/res_crypto.c, utils/astman.c, res/res_musiconhold.c, res/res_http_post.c, apps/app_queue.c, res/res_config_sqlite.c, agi/eagi-sphinx-test.c, utils/frame.c, channels/chan_dahdi.c, res/ael/ael.tab.c, funcs/func_odbc.c, main/ast_expr2f.c, res/res_agi.c, main/http.c, main/logger.c, channels/chan_h323.c, apps/app_sms.c, res/ael/ael.flex, pbx/pbx_config.c, apps/app_chanspy.c, apps/app_stack.c, utils/streamplayer.c, apps/app_adsiprog.c, apps/app_voicemail.c, apps/app_dial.c, channels/chan_sip.c, apps/app_festival.c, main/db1-ast/hash/hash_page.c, res/ael/ael.y, agi/eagi-test.c, pbx/pbx_lua.c, formats/format_ogg_vorbis.c, main/utils.c, utils/astcanary.c, formats/format_wav_gsm.c: import gcc 4.3.2 warning fixes from trunk, with a few changes specific to this branch 2008-11-02 20:07 +0000 [r153363-153653] Russell Bryant * include/asterisk/features.h, /: Merged revisions 153652 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r153652 | russell | 2008-11-02 14:06:03 -0600 (Sun, 02 Nov 2008) | 10 lines Merged revisions 153651 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r153651 | russell | 2008-11-02 13:51:17 -0600 (Sun, 02 Nov 2008) | 2 lines features.h depends on linkedlists.h, so include it ........ ................ * /, channels/chan_sip.c: Merged revisions 153362 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r153362 | russell | 2008-11-01 15:41:38 -0500 (Sat, 01 Nov 2008) | 3 lines Ensure that the sip_pvt properly has its refcount incremented when the scheduler holds a reference to it for session timer processing. ........ 2008-10-31 22:11 +0000 [r153266] Terry Wilson * apps/app_followme.c, apps/app_queue.c, apps/app_dial.c, /, main/features.c, include/asterisk/channel.h: Merged revisions 153181 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r153181 | twilson | 2008-10-31 13:55:33 -0500 (Fri, 31 Oct 2008) | 5 lines Recent CDR fixes moved execution of the 'h' exten into the bridging code, so variables that were set after ast_bridge_call was called would not show up in the 'h' exten. Added a callback function to handle setting variables, etc. from w/in the bridging code. Calls back into a nested function within the function calling ast_bridge_call (closes issue #13793) Reported by: greenfieldtech ........ 2008-10-31 20:10 +0000 [r153225] Mark Michelson * main/dial.c, include/asterisk/dial.h: This commit contains the bug fixes and documentation updates which were committed to trunk in revision 153223. I blocked that commit from 1.6.1 since it also contained a new feature. Note to self: Separate commits so that you don't end up with a situation where part of a commit should be merged but part should be blocked from stable branches. 2008-10-31 16:36 +0000 [r153123] Tilghman Lesher * channels/chan_sip.c: Turn off qualify on uncached realtime peers. (Closes issue #13383) 2008-10-30 21:01 +0000 [r152995] Sean Bright * bootstrap.sh: The -I argument to aclocal needs a space before the include directory name. 2008-10-30 20:36 +0000 [r152924-152974] Tilghman Lesher * channels/chan_h323.c: Cannot join detached threads. See http://www.opengroup.org/onlinepubs/000095399/functions/pthread_join.html (Closes issue #13400) * channels/chan_local.c: Unlock before returning, when extension doesn't exist. (closes issue #13807) Reported by: eliel Patches: chan_local.c.patch uploaded by eliel (license 64) 2008-10-30 19:41 +0000 [r152878-152921] Russell Bryant * channels/chan_sip.c: Fix the sip_peer reference count with respect to scheduler entries for scheduling peer pokes, and scheduling peer poke expirations. * channels/chan_sip.c: Fix the sip_peer reference count with respect to scheduler entries for registration expirations. * include/asterisk/sched.h: Fix a bug in AST_SCHED_REPLACE_UNREF(). The reference count of the object _must_ be increased before creating the scheduler entry. Otherwise, you create a race condition where the reference count may hit zero and the object can disappear out from under you. This could also would have incorrectly decreased the reference count in the case that the scheduler add failed. * channels/chan_sip.c: Modify the documentation of the sip_registry struct - Remove a comment that says that the monitor thread is the only one that ever touches these objects. This is no longer the case with TCP. Also, I would eventually like to get the scheduler in its own thread, so this is just a poor assumption to make. - Note that reference counting of these objects with respect to scheduler entries is not complete. There are some leaked references when deleting scheduler entries. 2008-10-30 16:55 +0000 [r152814] Kevin P. Fleming * main/cdr.c: instead of comparing the string pointer to 0, let's compare the value that was actually parsed out of the string (found by sparse) 2008-10-30 04:29 +0000 [r152690-152777] Tilghman Lesher * configs/extensions.conf.sample: Set up an example stdexten that preserves the original context and extension in the CDR. (Related to issue #13799) Reported by: davidw * main/pbx.c: Track down and fix annoying lock errors. These would occur when merging hints that resulted from a pattern matched hint during a 'dialplan reload'. 2008-10-29 20:55 +0000 [r152648] Mark Michelson * apps/app_directory.c: If there was no named defined in a voicemail.conf mailbox entry, then app_directory would crash when attempting to read that entry from the file. We now check for the NULL or empty string properly so that there will be no crash. (closes issue #13804) Reported by: bluecrow76 2008-10-29 20:16 +0000 [r152645] Terry Wilson * apps/app_queue.c: Small modification to putnopvut's patch to fix this issue. Thanks for all the help, putnopvut! (closes issue #12884) Reported by: bcnit Patches: 12884v4-1.6.0-branch.patch uploaded by otherwiseguy (license 396) Tested by: otherwiseguy 2008-10-29 05:52 +0000 [r152606] Steve Murphy * apps/app_queue.c, configs/features.conf.sample, apps/app_dial.c: A little documentation cross-ref between features and dial and queue... I wasted some time (stupidly) trying to get the one-touch parking stuff working, because it didn't occur to me that I had to also have the corresponding options in the dial command! Duh! (In all this time, I never set this up before!) So, to keep some poor fool from suffering the same fate, I made the features.conf.sample file mention the corresponding opts in dial/queue; and the docs for dial/app specifically mention the corresponding decls in the feature.conf file. I hope this doesn't spoil some vast, eternal plan... 2008-10-29 05:35 +0000 [r152573] Russell Bryant * channels/chan_sip.c: Fix an incorrect usage of sizeof() (closes issue #13795) Reported by: andrew53 Patches: chan_sip_sizeof.patch uploaded by andrew53 (license 519) 2008-10-29 05:09 +0000 [r152537] Steve Murphy * apps/app_queue.c, include/asterisk/features.h, apps/app_dial.c, main/features.c, include/asterisk/pbx.h: The magic trick to avoid this crash is not to try to find the channel by name in the list, which is slow and resource consuming, but rather to pay attention to the result codes from the ast_bridge_call, to which I added the AST_PBX_NO_HANGUP_PEER_PARKED value, which now are returned when a channel is parked. Why? because CDR's aren't generated via parking, so nothing is needed, but if a transfer occurred, there are critical things I need. If you get AST_PBX_KEEPALIVE, then don't touch the channel pointer. If you get AST_PBX_NO_HANGUP_PEER, or AST_PBX_NO_HANGUP_PEER_PARKED, then don't touch the peer pointer. Updated the several places where the results from a bridge were not being properly obeyed, and fixed some code I had introduced so that the results of the bridge were not overridden (in trunk). All the places that previously tested for AST_PBX_NO_HANGUP_PEER now have to check for both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED. I tested this against the 4 common parking scenarios: 1. A calls B; B answers; A parks B; B hangs up while A is getting the parking slot announcement, immediately after being put on hold. 2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but before the park times out. 3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold. 4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out. No crash. I also ran the scenarios above against valgrind, and accesses looked good. 2008-10-28 22:35 +0000 [r152370-152471] Tilghman Lesher * apps/app_voicemail.c: Quoting in the wrong direction (Fixes AST-107) * channels/chan_mgcp.c: Only re-add the io port if it was closed, otherwise reload causes a memory leak. (closes issue #13785) Reported by: eliel Patches: chan_mgcp.c.patch uploaded by eliel (license 64) * apps/app_dial.c: Reset all DIAL variables back to blank, in case Dial is called multiple times per call (which could otherwise lead to inconsistent status reports). (closes issue #13216) Reported by: ruddy Patches: 20081014__bug13216.diff.txt uploaded by Corydon76 (license 14) Tested by: ruddy 2008-10-27 23:32 +0000 [r152288] Jeff Peeler * channels/chan_dahdi.c: Buffer policy setting for half is not needed. 2008-10-27 21:53 +0000 [r152173-152217] Tilghman Lesher * channels/chan_local.c: Inherit ALL elements of CallerID across a local channel. (closes issue #13368) Reported by: Peter Schlaile Patches: 20080826__bug13368.diff.txt uploaded by Corydon76 (license 14) * apps/app_stack.c: Oops, only delete the ARG variables once upon release. The following section would have removed them again (removing variables from 2 stack frames, instead of just one). 2008-10-27 16:06 +0000 [r152133] Jason Parker * apps/app_transfer.c: Remove options argument parsing/syntax (it isn't used any longer) (closes issue #13789) Reported by: IgorG Patches: app_transfer.c.diff uploaded by IgorG (license 20) 2008-10-26 20:27 +0000 [r152068] Sean Bright * funcs/func_strings.c: Since passing \0 as the second argument to strchr is valid (and will match the trailing \0 of a string) we need to check that first, otherwise we end up with incorrect results. Fix suggested by reporter. (closes issue #13787) Reported by: meitinger 2008-10-25 11:11 +0000 [r151907] Russell Bryant * main/asterisk.c: Move AMI initialization to occur after loading modules. This prevents a deadlock when someone tries to initiate a module reload from the AMI just as Asterisk is starting. (closes issue #13778) Reported by: hotsblanc Fix suggested by hotsblanc 2008-10-22 20:08 +0000 [r151603] Tilghman Lesher * contrib/scripts/live_ast: Add a contributed script for running Asterisk without installing it, first. (closes issue #11680) Reported by: tzafrir Patches: live_ast_6 uploaded by tzafrir (license 46) 2008-10-22 20:05 +0000 [r151421-151602] Mark Michelson * channels/chan_dahdi.c: Change some logical ands to bitwise ands and add messages alerting that a channel is being ignored if the PROC_DAHDI_NOCHAN option is set in process_dahdi. (closes issue #13759) Reported by: smurfix Patches: dahdi.patch uploaded by smurfix (license 547) * channels/chan_sip.c: The logic of a strncasecmp call was reversed. (closes issue #13706) Reported by: andrew53 Patches: sip_notify_from_rfc3265.patch uploaded by andrew53 (license 519) * channels/chan_sip.c: Make the sip_standard_port function more granular by allowing separate type and port arguments. This is necessary because when building our From and Contact headers, we need to be absolutely sure that we are placing our source port there and not the peer's source port. (closes issue #12761) Reported by: asbestoshead Patches: patch-chan-sip-contact-port.txt uploaded by asbestoshead (license 455) * channels/chan_sip.c: Get this compiling in dev-mode * channels/chan_sip.c: If a peer uses any transport other than UDP, then MWI will fail for that peer since sip_alloc will allocate a sip_pvt with a default transport of UDP. This change resets the socket type immediately after allocating the sip_pvt in sip_send_mwi_from_peer, so that the proceeding call to create_addr_from_peer does not fail right away. The socket data from the peer is properly copied to the sip_pvt in create_addr_from_peer. (closes issue #13710) Reported by: andrew53 Patches: sip_notify_use_tcp.patch uploaded by andrew53 (license 519) * channels/chan_sip.c: When attempting to resolve hostnames, we need to be sure to remove any parameters from the string so that name resolution succeeds. (closes issue #13727) Reported by: fnordian Patches: resolvewithouturiparameter.patch uploaded by fnordian (license 110) 2008-10-21 15:21 +0000 [r151372] Tilghman Lesher * apps/app_mixmonitor.c: Default file modes should always be full read and write, to allow the system administrator to make the decision of what permissions will actually be given, through the use of the process umask. (Closes issue# 13751) 2008-10-21 11:03 +0000 [r151328] BJ Weschke * channels/chan_sip.c: Fix configuration parsing so type=friend still identifies "friend" as a peer even though it is now a legacy configuration verb. (closes issue #13705) reported by: blitzrage patched by: bweschke 2008-10-20 05:06 +0000 [r151135-151245] Kevin P. Fleming * autoconf (added), autoconf/ast_check_pwlib.m4, autoconf/acx_pthread.m4, autoconf/ast_func_fork.m4, configure, autoconf/ast_gcc_attribute.m4, bootstrap.sh, autoconf/ast_check_gnu_make.m4, autoconf/ast_ext_lib.m4, autoconf/ast_prog_ld.m4, autoconf/ast_c_compile_check.m4, autoconf/ast_c_define_check.m4, autoconf/ast_prog_egrep.m4, autoconf/ast_ext_tool_check.m4, autoconf/ast_check_mandatory.m4, autoconf/ast_check_openh323.m4, autoconf/ast_prog_ld_gnu.m4, configure.ac, acinclude.m4 (removed), autoconf/ast_prog_sed.m4: break up acinclude.m4 into individual files, which will make it easier to maintain, easier to add new macros (less patching) and will ease maintenance of these macros across Asterisk branches. Rename this macro to properly reflect what it does * main/tcptls.c, main/manager.c, channels/chan_sip.c, main/http.c, apps/app_externalivr.c, include/asterisk/tcptls.h: cleaup of the TCP/TLS socket API: 1) rename 'struct server_args' to 'struct ast_tcptls_session_args', to follow coding guidelines 2) make ast_make_file_from_fd() static and rename it to something that indicates what it really is for (again coding guidelines) 3) rename address variables inside 'struct ast_tcptls_session_args' to be more descriptive (dare i say it... coding guidelines) 4) change ast_tcptls_client_start() to use the new 'remote_address' field of the session args for the destination of the connection, and use the 'local_address' field to bind() the socket to the proper source address, if one is supplied 5) in chan_sip, ensure that we pass in the PP address we are bound to when creating outbound (client) connections, so that our connections will appear from the correct address 2008-10-18 02:29 +0000 [r150829] BJ Weschke * main/manager.c: Using the GetVar handler in AMI is potentially dangerous (insta-crash [tm]) when you use a dialplan function that requires a channel and then you don't provide one or provide an invalid one in the Channel: parameter. We'll handle this situation exactly the same way it was handled in pbx.c back on r61766. We'll create a bogus channel for the function call and destroy it when we're done. If we have trouble allocating the bogus channel then we're not going to try executing the function call at all and run the risk of crashing. (closes issue #13715) reported by: makoto patch by: bweschke 2008-10-17 17:10 +0000 [r150606-150636] Tilghman Lesher * channels/chan_iax2.c: Make helper call a little safer (suggested by Russell on IRC) * channels/chan_iax2.c, include/asterisk/sched.h: Fix the FRACK! warnings in chan_iax2 when POKE/LAGRQ packets are not answered. 2008-10-16 23:41 +0000 [r150208-150306] Mark Michelson * main/manager.c: Reverting changes from commits 150298 and 150301 since I was mistakenly under the assumption that dialplan functions *always* required that a channel be present. I need to go home earlier, I think :) * main/manager.c: Don't try to call a dialplan function's read callback from the manager's GetVar handler if an invalid channel has been specified. Several dialplan functions, including CHANNEL and SIP_HEADER, do not check for NULL-ness of the channel being passed in. (closes issue #13715) Reported by: makoto And don't forget to return on the error condition * apps/app_sms.c: Answer the channel prior to checking for the 'a' option in app_sms. (closes issue #13675) Reported by: alecdavis Patches: app_sms.bug13675.148985.diff.txt uploaded by alecdavis (license 585) * configure, configure.ac: Change configure script to search for openais in both /usr/lib and /usr/lib64 since some distros place 64-bit libraries only in the /usr/lib64 directory. (closes issue #13721) Reported by: jcollie Patches: 0007-Look-in-64bit-dirs-for-openais.patch uploaded by jcollie (license 412) * channels/chan_sip.c: INVITES with proxy auth were sent with a different branch than what was in the invite_branch of a sip_pvt, meaning that if a CANCEL were sent later, the branch in the CANCEL would not match the branch in the latest INVITE sent out, leading to some endpoints responding to the CANCEL with a 481. (closes issue #13714) Reported by: fnordian Patches: invite_branch.patch uploaded by fnordian (license 110) 2008-10-16 16:17 +0000 [r150127] Richard Mudgett * channels/chan_misdn.c: Fix memory leak found by customer 2008-10-16 13:32 +0000 [r149919-149995] Kevin P. Fleming * channels/chan_sip.c: return this logic to where it used to be, *after* the dialog->needdestroy flag has been determined to be set; otherwise, we generate these debug messages every time we inspect every active dialog * apps/app_stack.c: building this module depends on res_agi being built as well * res/res_phoneprov.c: inter-module dependencies should be included in the source code, not just in sample config files * res/res_phoneprov.c: correct file name in message 2008-10-15 21:00 +0000 [r149803] Mark Michelson * channels/chan_sip.c: Make the sip_proxy struct reference counted. This is necessary to allow for a sip_pvt to maintain a reference to a sip_peer's outboundproxy even after the peer has been freed. (closes issue #13700) Reported by: fnordian Patches: 13700.patch uploaded by putnopvut (license 60) Tested by: fnordian 2008-10-15 20:22 +0000 [r149758] BJ Weschke * configs/agents.conf.sample: An update to the documentation/example of agents.conf.sample with the correct parameter for this feature as defined in chan_agent.c (closes issue #13709) 2008-10-15 19:09 +0000 [r149589-149688] Tilghman Lesher * funcs/func_odbc.c: Permit data fields to contain more than 255 characters. (closes issue #13631) Reported by: seanbright Patches: 20081015__bug13631.diff.txt uploaded by Corydon76 (license 14) Tested by: blitzrage * funcs/func_odbc.c: Only set buf to blank before the goto. * codecs/lpc10/lpcini.c: When using MALLOC_DEBUG, codec_lpc10 leaks memory, because it matches a library malloc() with an ast_free (which, of course, doesn't match up with known allocated memory, so the free fails). (closes issue #13702) Reported by: eliel Patches: codec_lpc10_lpcini.c uploaded by eliel (license 64) * apps/app_echo.c: Minor spacing change (closes issue #13697) Reported by: alecdavis Patches: app_echo.bug13697.103249.diff.txt uploaded by alecdavis (license 585) 2008-10-15 11:32 +0000 [r149512] Kevin P. Fleming * channels/chan_sip.c: fix some problems when parsing SIP messages that have the maximum number of headers or body lines that we support 2008-10-14 23:58 +0000 [r149203-149280] Mark Michelson * CHANGES, apps/app_dial.c: When specifying an invalid timeout to Dial, take it to mean that no timeout is desired. (closes issue #13625) Reported by: atis * channels/chan_sip.c: Change this warning to an error message. Suggestion comes from Sean Bright. Thanks Sean! * channels/chan_sip.c: Call register_peer_exten even in the case that the peer's IP/port does not change. (closes issue #13309) Reported by: dimas Patches: v2-13309.patch uploaded by dimas (license 88) * include/asterisk/audiohook.h, main/audiohook.c: Add a tolerance period for sync-triggered audiohooks so that if packetization of audio is close (but not equal) we don't end up flushing the audiohooks over small inconsistencies in synchronization. Related to issue #13005, and solves the issue for most people who were experiencing the problem. However, a small number of people are still experiencing the problem on long calls, so I am not closing the issue yet * apps/app_queue.c: Update the queue with the correct number of calls and whether the call was completed within the service level when a transfer takes place. This way, we do not "break" the leastrecent and fewestcalls strategies by not logging a call until after the transferred call has ended. (closes issue #13395) Reported by: Marquis Patches: app_queue.c.transfer.patch uploaded by Marquis (license 32) 2008-10-14 22:42 +0000 [r149202] Tilghman Lesher * include/asterisk/hashtab.h, main/chanvars.c, main/config.c, main/hashtab.c, pbx/pbx_spool.c, channels/chan_sip.c, include/asterisk/chanvars.h, include/asterisk/config.h, include/asterisk/strings.h, res/res_indications.c: Add additional memory debugging to several core APIs, and fix several memory leaks found with these changes. (Closes issue #13505, closes issue #13543) Reported by: mav3rick, triccyx Patches: 20081001__bug13505.diff.txt uploaded by Corydon76 (license 14) Tested by: mav3rick, triccyx 2008-10-14 21:09 +0000 [r149132] Mark Michelson * channels/chan_sip.c: Don't allow reserved characters to be used in register lines in sip.conf. (closes issue #13570) Reported by: putnopvut 2008-10-14 20:17 +0000 [r149063] Tilghman Lesher * apps/app_waitforsilence.c: Check correct values in the return of ast_waitfor(); also, get rid of a possible memory leak. (closes issue #13658) Reported by: explidous Patch by: me 2008-10-14 19:42 +0000 [r149060] Leif Madsen * doc/manager_1_1.txt: Add missing documentation for SipShowRegistry action and RegistryEntry event. (closes issue #13342) Reported and patch by: Laureano 2008-10-14 18:59 +0000 [r148918-148986] Tilghman Lesher * apps/app_sms.c: App is ignoring 'p' parameter -- initial pause. (closes issue #13617) Reported by: alecdavis Patches: app_sms.13oct.diff.txt uploaded by alecdavis (license 585) * apps/app_voicemail.c: Ensure that mail headers are 7-bit clean, even when UTF-8 characters are used in headers like 'Subject' and 'To'. Closes AST-107. 2008-10-14 17:39 +0000 [r148915] Mark Michelson * channels/chan_local.c: Deadlock prevention in chan_local. (closes issue #13676) Reported by: tacvbo Patches: 13676.patch uploaded by putnopvut (license 60) Tested by: tacvbo 2008-10-14 15:18 +0000 [r148869] Tilghman Lesher * apps/app_fax.c: API differences in spandsp 0.0.6pre1 and higher (closes issue #13688) Reported by: irroot Patches: app_fax-span6.patch uploaded by irroot (license 52) with minor modifications by me 2008-10-14 11:35 +0000 [r148614-148763] Kevin P. Fleming * channels/chan_sip.c: fix some references to the owner of a private structure that may not be present * Makefile: on Ubuntu (at least), recent versions of ld in binutils delete all debugging symbols when -x is supplied; since the reasons why -x is being passed are lost in the mists of time, remove it so debugging will work properly * channels/chan_sip.c: ensure that *all* fields in the req structure are cleared out before reusing it; has_to_tag was not cleared, which caused the second incoming call over a TCP socket to fail if pedantic checking was enabled * main/translate.c: it would be nice if this message printing code had actually been tested before it was committed... 2008-10-13 17:56 +0000 [r148562] Steve Murphy * main/pbx.c: Hmmm. Nobody (but me) is interested in seeing the trie info when they do 'dialplan show ...' (even with debug set to non-zero); so I set up a 'dialplan debug [context]' cli command instead, to explicitly show just the trie info. I even added an extension_exists() call to make sure the trie info is built. I moved the explanatory header to above the extension loop to ensure it only prints once. And it will do this now, whether debug is set or not. I removed the trie printing from the 'dialplan show' command entirely. 2008-10-13 15:36 +0000 [r148472] Olle Johansson * channels/chan_sip.c: Sending a 403 after a 200 is considered very bad. (found at SIPit) 2008-10-10 21:22 +0000 [r148375-148377] Mark Michelson * channels/chan_sip.c: The logic used when checking a peer got changed subtly in the "kill the user" commit and caused calls relying on the insecure setting to not work properly. I changed for finding a peer back to how it was prior to that commit. (closes issue #13644) Reported by: pj Patches: 13644_trunkv2.patch uploaded by putnopvut (license 60) Tested by: pj * channels/chan_sip.c: Make sure that the inUse and inRinging fields for a sip peer cannot go below zero. This is a regression from 1.4 and so it will be applied to 1.6.0 as well. (closes issue #13668) Reported by: mjc 2008-10-10 16:37 +0000 [r148269] Tilghman Lesher * apps/app_voicemail.c: User not notified of temporary greeting, if ODBC storage is in use. (closes issue #13659) Reported by: moliveras Patches: 20081009__bug13659.diff.txt uploaded by Corydon76 (license 14) Tested by: moliveras 2008-10-10 01:33 +0000 [r148240] Sean Bright * res/res_config_sqlite.c, apps/app_voicemail.c, include/asterisk.h, main/tdd.c, main/cryptostub.c: Don't include logger.h in asterisk.h by default as it is causing problems building app_voicemail. Instead, include it where it is needed. This turned out to be a relatively minor issue because other headers include logger.h as well. Need to test -addons before merging this back to 1.6.0. (closes issue #13605) Reported by: tomo1657 Patches: 13605_seanbright.diff uploaded by seanbright (license 71) Tested by: mmichelson 2008-10-09 23:55 +0000 [r148151-148161] Mark Michelson * main/manager.c: The priority was unnecessary for the manager atxfer, so it has been removed. Furthermore, now we actually use the Context argument passed to set the transfer context and don't error out if no context is specified. This addresses the actual problems outlined in issue 12158. Regarding the other points brought up, regarding the inability to not transfer to extensions which cannot be represented by DTMF, it is not enough of a constraint that it is worth attempting to rework the feature. (closes issue #12158) Reported by: davidw * apps/app_voicemail.c: Read the callerid in the correct order and make sure to read the Urgent flag value from the IMAP headers. (closes issue #13652) Reported by: jaroth Patches: imapheaders.patch uploaded by jaroth (license 50) 2008-10-09 23:27 +0000 [r148128] Tilghman Lesher * configs/res_ldap.conf.sample: Fix example schema (closes issue #12860) Reported by: flyn Patches: res_ldap.conf.patch uploaded by flyn (license 503) 2008-10-09 23:20 +0000 [r148115] Mark Michelson * main/features.c: (closes issue #13579) Reported by: dwagner (closes issue #13584) Reported by: dwagner Tested by: murf, putnopvut The thought occurred to me that the res= from the extension spawn was ending up being returned from the bridge. "Thou shalt not poison the return value". Made the change and it appears to allow blind xfers to work as normal. If I'm wrong, reopen the bugs. But it looks good to me! Many thanks to putnopvut for helping me reproduce this! 2008-10-09 20:01 +0000 [r148006-148011] Tilghman Lesher * sounds/Makefile, sounds/sounds.xml: Publish MOH files in sln16 format * apps/app_voicemail.c: When blank, callerid name and number should display "unknown caller" in voicemail emails. (Closes issue #13643) 2008-10-09 19:28 +0000 [r147957] Jeff Peeler * main/features.c: (closes issue #13139) Reported by: krisk84 Tested by: krisk84 This change prevents a call that is placed in the parkinglot to be picked up before the PBX is finished. If another extension dials the parking extension before the PBX thread has completed at minimum warnings will occur about the PBX not properly being terminated. At worst, a crash could occur. 2008-10-09 17:54 +0000 [r147901] Michiel van Baak * include/asterisk/endian.h: only include this for OpenBSD. At least FreeBSD is borked when including it (closes issue #13649) Reported by: ys 2008-10-09 17:47 +0000 [r147898] Tilghman Lesher * configs/extensions.conf.sample: Remove "second form" of extensions, as it no longer applies. Also, cleanup the grammar, formatting, and introduce several clarifications to the text. (Closes issue #13654) 2008-10-09 15:06 +0000 [r147811] Steve Murphy * channels/chan_iax2.c, main/astobj2.c, channels/chan_oss.c, main/config.c, main/rtp.c, main/cli.c, channels/chan_usbradio.c, configure, channels/console_gui.c, utils/extconf.c, main/pbx.c, include/asterisk.h, doc/CODING-GUIDELINES, include/asterisk/autoconfig.h.in, main/translate.c, channels/vcodecs.c, configure.ac, channels/console_video.c: (closes issue #13557) Reported by: nickpeirson Patches: pbx.c.patch uploaded by nickpeirson (license 579) replace_bzero+bcopy.patch uploaded by nickpeirson (license 579) Tested by: nickpeirson, murf 1. replaced all refs to bzero and bcopy to memset and memmove instead. 2. added a note to the CODING-GUIDELINES 3. add two macros to asterisk.h to prevent bzero, bcopy from creeping back into the source 4. removed bzero from configure, configure.ac, autoconfig.h.in 2008-10-08 22:33 +0000 [r147719] Mark Michelson * apps/app_meetme.c: Some small tweaks regarding realtime conference announcements. (closes issue #13522) Reported by: DEA Patches: meetme-rt-fixes.txt uploaded by DEA (license 3) 2008-10-08 22:27 +0000 [r147692] Kevin P. Fleming * channels/chan_dahdi.c: when parsing a text configuration option, ensure that the buffer on the stack is actually large enough to hold the legal values of that option, and also ensure that sscanf() knows to stop parsing if it would overrun the buffer (without these changes, specifying "buffers=...,immediate" would overflow the buffer on the stack, and could not have worked as expected) 2008-10-08 19:09 +0000 [r147593] Tilghman Lesher * apps/app_sms.c: Correct a typo in the help; also, ensure that the date and time are correctly set, if not specified in the message. (Closes issue #13594, closes issue #13595) Reported by: alecdavis Patches: 20081001__bug13595.diff.txt uploaded by Corydon76 (license 14) Tested by: alecdavis 2008-10-08 15:10 +0000 [r147519] Mark Michelson * apps/app_speech_utils.c: If we receive DTMF make sure that the state of the speech structure goes back to being not ready. (issue #LUMENVOX-8) 2008-10-07 16:54 +0000 [r147196] Sean Bright * apps/app_voicemail.c: Make 'imapsecret' an alias to 'imappassword' in voicemail.conf. 2008-10-07 16:05 +0000 [r147147] Jeff Peeler * main/features.c: Explicitly setting these fields to NULL was done because I wasn't sure if they would be NULL otherwise. Since they will be set automatically, removing. 2008-10-07 15:06 +0000 [r147100] Richard Mudgett * funcs/func_callerid.c: Independent change from branch issue8824 that is not part of COLP. (-r142574 rmudgett) 2008-10-07 12:03 +0000 [r147052] Sean Bright * apps/app_dial.c: Make sure to compare the correct number of characters when special-casing our DAHDI operator mode stuff. Technically, it would work fine, as 'DAH' is currently unique amongst our channel technologies, but as Jared points out: <@jsmith> Sure... as long as the technology starts whith DAH.... but it could be DAHDOO! 2008-10-07 00:13 +0000 [r146972] Terry Wilson * channels/chan_sip.c: A blind transfer to the parking thread would cause a segfault because copy_request accesses dst->data w/o being able to tell whether it is proerly initialized 2008-10-06 23:22 +0000 [r146930] Tilghman Lesher * include/asterisk/threadstorage.h: Update documentation; AST_THREADSTORAGE() in trunk only takes a single argument. 2008-10-06 23:08 +0000 [r146876-146924] Jeff Peeler * include/asterisk/features.h, main/features.c, res/res_agi.c: Similar to r143204, masquerade the channel in the case of Park being called from AGI. ........ * include/asterisk/endian.h: Mvanbaak said this was needed to compile on OpenBSD, so put it in the OpenBSD section. * main/features.c: This commit squashes together three commits because the wrong approach was originally used. (One of the commits was only one line.) 1) r143204: The main change here was to masquerade the channel if the channel that was to be parked was running a PBX on it. The PBX thread can then maintain full control of the channel (the zombie) as it expects to while allowing the parking thread full control of the real (parked) channel. 2) r143270: Changed park_call_full to hold the parkinglot lock a little longer, which protects the parkeduser struct from being freed out from underneath. Made sure that the parking extension is added to the parking context while holding the lock thereby ensuring that there are no spurious warnings from removal attempts when a hangup occurs while the parking lot is being announced. 3) r143475: (the one liner) compare peer and chan instead of looking at the parked user (pu), which could have possibly already have been freed by the parking thread * main/features.c: fix some comment placement * main/features.c: Explicitly set args in park_call_exec NULL so in the case of no options being passed in, there is no garbage attempted to be used. Also, do not set args to unknown value again if there are no options passed in. 2008-10-06 21:53 +0000 [r146874] Michiel van Baak * include/asterisk/endian.h: make aescrypt.c compile on OpenBSD again 2008-10-06 21:32 +0000 [r146715-146838] Tilghman Lesher * channels/chan_iax2.c, funcs/func_callerid.c, apps/app_speech_utils.c, funcs/func_curl.c, funcs/func_groupcount.c, res/res_smdi.c, channels/chan_sip.c, funcs/func_timeout.c, funcs/func_odbc.c, funcs/func_cdr.c, funcs/func_math.c: Dialplan functions should not actually return 0, unless they have modified the workspace. To signal an error (and no change to the workspace), -1 should be returned instead. (closes issue #13340) Reported by: kryptolus Patches: 20080827__bug13340__2.diff.txt uploaded by Corydon76 (license 14) * channels/chan_local.c: Check whether an extension exists in the _call method, rather than the _alloc method, because we need to evaluate the callerid (since that data affects whether an extension exists). (closes issue #13343) Reported by: efutch Patches: 20080915__bug13343.diff.txt uploaded by Corydon76 (license 14) Tested by: efutch 2008-10-06 16:39 +0000 [r146698] Kevin P. Fleming * channels/chan_dahdi.c: ensure that the private structure for pseudo channels is created without 'leaking' configuration data from other configured channels (closes issue #13555) Reported by: jeffg Patches: issue_13555.patch uploaded by kpfleming (license 421) Tested by: jeffg 2008-10-06 00:23 +0000 [r146557] Sean Bright * utils/Makefile: Quote arguments to cp so we can handle spaces in our paths. 2008-10-05 21:24 +0000 [r146451] Jason Parker * channels/chan_sip.c: Fix silly formatting. 2008-10-04 01:57 +0000 [r146314] Sean Bright * configs/sip_notify.conf.sample: Add ability to remotely reboot snom phones. Also cleaned up and reorganized sip_notify.conf.sample a bit as well. Tested snom reboot on snom 360 and verified snom-check-cfg worked as well. (closes issue #13601) Reported by: mjc Tested by: seanbright 2008-10-03 22:42 +0000 [r146243] Jeff Peeler * main/features.c: remove superfluous reference counting operations in manage_parkinglot since ao2_interator_next increments the ref count automatically 2008-10-03 22:13 +0000 [r146200] Sean Bright * main/cli.c: Resolve a subtle bug where we would never successfully be able to get the first item in the CLI entry list. This was preventing '!' from showing up in either 'help' or in tab completion. (closes issue #13578) Reported by: mvanbaak 2008-10-02 19:31 +0000 [r145960-145964] Russell Bryant * CHANGES: The 'P' command for ExternalIVR was also added in 1.6.0 * CHANGES: TCP support for ExternalIVR went in to 1.6.1, not 1.6.0 2008-10-02 15:30 +0000 [r145781] Sean Bright * configure, configure.ac: This is much cleaner, methinks. 2008-10-02 15:19 +0000 [r145754] Tilghman Lesher * res/res_odbc.c: Some sanity checks that may have led to prior crashes, found by codefreeze-lap (murf) on IRC. Also some cleanup of incorrectly-used constants. 2008-10-01 23:54 +0000 [r145694] Sean Bright * configure, configure.ac: Try a test compile using the GMime library. Some distros install gmime-config in the base package instead of the -devel package. Now we print a notice and disable GMime support instead of bombing during the main compilation. (closes issue #13583) Reported by: arkadia 2008-10-01 22:24 +0000 [r145557-145609] Mark Michelson * main/features.c: Okay, this should really do it now. While I did manage to fix blind transfers with my last commit here, I also caused an unwanted side-effect. That is, only the first priority of the 'h' extension would be executed when a blind transfer occurred instead of all priorities. Essentially, my last commit corrected the return value of ast_bridge_call. However, the implementation still was not 100% correct. Now it is. * main/features.c: if (!(x) == 0) is the same as if (x). * main/features.c: The logic surrounding the return value of ast_spawn_extension within ast_bridge_call was reversed. This problem was observed when a blind transfer placed from the callee channel of a test call failed. While the problem I am solving here is exactly the same as what was reported in issue #13584, the difference is that this fix I am applying is trunk-only. Issue #13584 was reported against the 1.4 branch, and my tests of 1.4's blind transfers appear to work fine. 2008-10-01 Russell Bryant * Asterisk 1.6.0 released. 2008-09-09 Russell Bryant * Asterisk 1.6.0-rc6 released. 2008-09-09 15:44 +0000 [r142065] Russell Bryant * /, main/features.c: Merged revisions 142064 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r142064 | russell | 2008-09-09 10:44:10 -0500 (Tue, 09 Sep 2008) | 13 lines Merged revisions 142063 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142063 | russell | 2008-09-09 10:40:24 -0500 (Tue, 09 Sep 2008) | 5 lines Ensure that the stored CDR reference is still valid after the bridge before poking at it. Also, keep the channel locked while messing with this CDR. (fixes crashes reported in issue #13409) ........ ................ 2008-09-09 12:34 +0000 [r141996-141999] Mark Michelson * channels/chan_oss.c, /: Merged revisions 141995 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r141995 | mmichelson | 2008-09-09 05:20:58 -0500 (Tue, 09 Sep 2008) | 8 lines Fix a memory leak in chan_oss (closes issue #13311) Reported by: eliel Patches: chan_oss.c.patch uploaded by eliel (license 64) ........ 2008-09-09 01:49 +0000 [r141950] Russell Bryant * main/channel.c, /: Merged revisions 141949 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r141949 | russell | 2008-09-08 20:47:56 -0500 (Mon, 08 Sep 2008) | 9 lines Modify ast_answer() to not hold the channel lock while calling ast_safe_sleep() or when calling ast_waitfor(). These are inappropriate times to hold the channel lock. This is what has caused "could not get the channel lock" messages from chan_sip and has likely caused a negative impact on performance results of SIP in Asterisk 1.6. Thanks to file for pointing out this section of code. (closes issue #13287) (closes issue #13115) ........ 2008-09-08 21:07 +0000 [r141808] Russell Bryant * main/pbx.c, /: Merged revisions 141807 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r141807 | russell | 2008-09-08 16:05:01 -0500 (Mon, 08 Sep 2008) | 15 lines Merged revisions 141806 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141806 | russell | 2008-09-08 16:02:36 -0500 (Mon, 08 Sep 2008) | 7 lines When doing an async goto, detect if the channel is already in the middle of a masquerade. This can happen when chan_local is trying to optimize itself out. If this happens, fail the async goto instead of bursting into flames. (closes issue #13435) Reported by: geoff2010 ........ ................ 2008-09-08 Russell Bryant * Asterisk 1.6.0-rc5 released. 2008-09-08 20:19 +0000 [r141746] Jason Parker * Makefile, /, redhat (removed): Merged revisions 141745 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r141745 | qwell | 2008-09-08 15:18:17 -0500 (Mon, 08 Sep 2008) | 16 lines Merged revisions 141741 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141741 | qwell | 2008-09-08 15:15:42 -0500 (Mon, 08 Sep 2008) | 8 lines Remove RPM package targets from Makefile (and all associated parts). This has never worked in 1.4, and we decided that it makes no sense to be done here. There are many distros out there that already have "proper" spec files that can be (re)used. Closes issue #13113 Closes issue #10950 Closes issue #10952 ........ ................ 2008-09-08 17:14 +0000 [r141683] Sean Bright * /, build_tools/make_buildopts_h: Merged revisions 141682 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r141682 | seanbright | 2008-09-08 13:13:04 -0400 (Mon, 08 Sep 2008) | 9 lines Quote the arguments to grep so that sh on various platforms doesn't choke on the special characters (like ^). (closes issue #13417) Reported by: dougm Patches: 13417.make_buildopts_h.patch uploaded by seanbright (license 71) Tested by: dougm ........ 2008-09-06 20:21 +0000 [r141567] Steve Murphy * /, channels/chan_sip.c: Merged revisions 141566 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r141566 | murf | 2008-09-06 14:19:50 -0600 (Sat, 06 Sep 2008) | 9 lines Merged revisions 141565 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141565 | murf | 2008-09-06 14:13:16 -0600 (Sat, 06 Sep 2008) | 1 line This fix comes from Joshua Colp The Brilliant, who, given the trace, came up with a solution. This will most likely will close 13235 and 13409. I'll wait till Monday to verify, and then close these bugs. ........ ................ 2008-09-06 15:40 +0000 [r141505-141508] Tilghman Lesher * /, res/res_agi.c: Merged revisions 141504 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r141504 | tilghman | 2008-09-06 10:26:45 -0500 (Sat, 06 Sep 2008) | 12 lines Merged revisions 141503 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141503 | tilghman | 2008-09-06 10:23:42 -0500 (Sat, 06 Sep 2008) | 4 lines Reverting behavior change (AGI should not exit non-zero on SUCCESS) (closes issue #13434) Reported by: francesco_r ........ ................ 2008-09-05 22:06 +0000 [r141368-141426] Mark Michelson * /, channels/chan_agent.c: Merged revisions 141367 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r141367 | mmichelson | 2008-09-05 16:12:09 -0500 (Fri, 05 Sep 2008) | 15 lines Merged revisions 141366 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141366 | mmichelson | 2008-09-05 16:10:32 -0500 (Fri, 05 Sep 2008) | 7 lines Agent's should not try to call a channel's indicate callback if the channel has been hung up. It will likely crash otherwise ABE-1159 ........ ................ 2008-09-05 14:24 +0000 [r141116-141158] Steve Murphy * main/channel.c, /: Merged revisions 141157 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r141157 | murf | 2008-09-05 08:18:43 -0600 (Fri, 05 Sep 2008) | 9 lines Merged revisions 141156 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141156 | murf | 2008-09-05 08:15:43 -0600 (Fri, 05 Sep 2008) | 1 line A small change to prevent double-posting of CDR's; thanks to Daniel Ferrer for bringing it to our attention ........ ................ * pbx/ael/ael-test/ref.ael-vtest25 (added), /, pbx/ael/ael-test/ael-vtest25/extensions.ael, pbx/ael/ael-test/ael-vtest25 (added), res/ael/ael_lex.c, pbx/ael/ael-test/ref.ael-test6, res/ael/ael.flex: Merged revisions 141115 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r141115 | murf | 2008-09-04 17:31:41 -0600 (Thu, 04 Sep 2008) | 78 lines Merged revisions 141094 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141094 | murf | 2008-09-04 17:15:07 -0600 (Thu, 04 Sep 2008) | 70 lines (closes issue #13357) Reported by: pj Tested by: murf (closes issue #13416) Reported by: yarns Tested by: murf If you find this message overly verbose, relax, it's probably not meant for you. This message is meant for probably only two people in the whole world: me, or the poor schnook that has to maintain this code because I'm either dead or unavailable at the moment. This fix solves two reports, both having to do with embedding a function call in a ${} construct. It was tricky because the funccall syntax has parenthesis () in it. And up till now, the 'word' token in the flex stuff didn't allow that, because it would tend to steal the LP and RP tokens. To be truthful, the "word" token was the trickiest, most unstable thing in the whole lexer. I was lucky it made this long without complaints. I had to choose every character in the pattern with extreme care, and I knew that someday I'd have to revisit it. Well, the day has come. So, my brilliant idea (and I'm being modest), was to use the surrounding ${} construct to make a state machine and capture everything in it, no matter what it contains. But, I have to now treat the word token like I did with comments, in that I turn the whole thing into a state-machine sort of spec, with new contexts "curlystate", "wordstate", and "brackstate". Wait a minute, "brackstate"? Yes, well, it didn't take very many regression tests to point out if I do this for ${} constructs, I also have to do it with the $[] constructs, too. I had to create a separate pcbstack2 and pcbstack3 because these constructs can occur inside macro argument lists, and when we have two state machines operating on the same structures we'd get problems otherwise. I guess I could have stopped at pcbstack2 and had the brackstate stuff share it, but it doesn't hurt to be safe. So, the pcbpush and pcbpop routines also now have versions for "2" and "3". I had to add the {KEYWORD} construct to the initial pattern for "word", because previously word would match stuff like "default7", because it was a longer match than the keyword "default". But, not any more, because the word pattern only matches only one or two characters now, and it will always lose. So, I made it the winner again by making an optional match on any of the keywords before it's normal pattern. I added another regression test to make sure we don't lose this in future edits, and had to fix just one regression, where it no longer reports a 'cascaded' error, which I guess is a plus. I've given some thought as to whether to apply these fixes to 1.4 and the 1.6.x releases, vs trunk; I decided to put it in 1.4 because one of the bug reports was against 1.4; and it is unexpected that AEL cannot handle this situation. It actually reduced the amount of useless "cascade" error messages that appeared in the regressions (by one line, ehhem). There is a possible side-effect in that it does now do more careful checking of what's in those ${} constructs, as far as matching parens, and brackets are concerned. Some users may find a an insidious problem and correct it this way. This should be exceedingly rare, I hope. ........ ................ 2008-09-04 18:35 +0000 [r141086] Jeff Peeler * /, main/features.c, res/res_agi.c: Merged revisions 141039 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r141039 | jpeeler | 2008-09-04 12:27:56 -0500 (Thu, 04 Sep 2008) | 15 lines Merged revisions 141028 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141028 | jpeeler | 2008-09-04 12:00:29 -0500 (Thu, 04 Sep 2008) | 7 lines (closes issue #11979) Fixes multiple parking problems: Crash when executing a park on an extension dialed by AGI due to not returning the proper return code. Crash when using a builtin feature that was a subset of a enabled dynamic feature. Crash due to always hanging up the peer despite the fact that the peer was supposed to be parked. ........ ................ 2008-09-03 20:18 +0000 [r140976] Mark Michelson * /, apps/app_queue.c: Merged revisions 140975 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r140975 | mmichelson | 2008-09-03 15:16:12 -0500 (Wed, 03 Sep 2008) | 4 lines Fix some locking order issues in app_queue. This was brought up by atis on IRC a while ago. ........ 2008-09-03 Russell Bryant * Asterisk 1.6.0-rc4 released. 2008-09-03 14:17 +0000 [r140825-140827] Steve Murphy * main/cdr.c, /: Merged revisions 140749 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r140749 | murf | 2008-09-02 17:44:04 -0600 (Tue, 02 Sep 2008) | 11 lines Merged revisions 140747 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140747 | murf | 2008-09-02 17:36:56 -0600 (Tue, 02 Sep 2008) | 1 line I am turning the warnings generated in ast_cdr_free and post_cdr into verbose level 2 messages. Really, they matter little to end users. You either get the CDR's you wanted, or you don't, and it is a bug. For trunk, I am going one step further. These messages were pretty worthless even for debug, so I'm completely removing them. ........ ................ * main/channel.c, /: Merged revisions 140692 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r140692 | murf | 2008-09-02 16:55:12 -0600 (Tue, 02 Sep 2008) | 13 lines Merged revisions 140690 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140690 | murf | 2008-09-02 16:40:13 -0600 (Tue, 02 Sep 2008) | 1 line After reconsidering, with respect to 13409, ast_cdr_detach should be OK, better in fact, than ast_cdr_free, which generates lots of useless warnings that will undoubtably generate complaints. Hmmm. It doesn't hush the useless warnings, but it does allow control of posting via the detach and post routines, for those possible situations, where you'd want to post single-channel cdrs. ........ ................ * main/channel.c, main/pbx.c, /: Merged revisions 140691 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r140691 | murf | 2008-09-02 16:50:59 -0600 (Tue, 02 Sep 2008) | 22 lines Merged revisions 140670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140670 | murf | 2008-09-02 16:15:57 -0600 (Tue, 02 Sep 2008) | 14 lines (closes issue #13409) Reported by: tomaso Patches: asterisk-1.6.0-rc2-cdrmemleak.patch uploaded by tomaso (license 564) I basically spent the day, verifying that this patch solves the problem, and doesn't hurt in non-problem cases. Why valgrind did not plainly reveal this leak absolutely mystifies and stuns me. Many, many thanks to tomaso for finding and providing the fix. ........ ................ 2008-09-03 13:27 +0000 [r140818] Russell Bryant * main/poll.c, /: Merged revisions 140817 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r140817 | russell | 2008-09-03 08:26:43 -0500 (Wed, 03 Sep 2008) | 12 lines Merged revisions 140816 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140816 | russell | 2008-09-03 08:24:35 -0500 (Wed, 03 Sep 2008) | 4 lines Don't freak out if the poll emulation receives NULL for the pollfds array (closes issue #13307) Reported by: jcovert ........ ................ 2008-09-02 18:17 +0000 [r140607] Sean Bright * /, channels/chan_iax2.c: Merged revisions 140606 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r140606 | seanbright | 2008-09-02 14:15:54 -0400 (Tue, 02 Sep 2008) | 16 lines Merged revisions 140605 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140605 | seanbright | 2008-09-02 14:14:57 -0400 (Tue, 02 Sep 2008) | 8 lines Make sure to use the correct length of the mohinterpret and mohsuggest buffers when copying configuration values. (closes issue #13336) Reported by: decryptus_proformatique Patches: chan_iax2_mohinterpret_mohsuggest_general_settings.patch uploaded by decryptus (license 555) ........ ................ 2008-09-02 15:12 +0000 [r140564-140567] Russell Bryant * apps/app_jack.c, /, codecs/codec_resample.c: Merged revisions 140566 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r140566 | russell | 2008-09-02 10:11:53 -0500 (Tue, 02 Sep 2008) | 2 lines Update instructions for getting libresample ........ 2008-08-27 20:15 +0000 [r140302-140304] Mark Michelson * channels/chan_sip.c: Revert commit 140302. Should not be merging changes like that into a release-candidate branch * channels/chan_sip.c: Merged revisions 140301 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r140301 | mmichelson | 2008-08-27 15:11:22 -0500 (Wed, 27 Aug 2008) | 19 lines Merged revisions 140299 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140299 | mmichelson | 2008-08-27 14:49:20 -0500 (Wed, 27 Aug 2008) | 11 lines Fix tag checking in get_sip_pvt_byid_locked when in pedantic mode. The problem was that the wrong tags would be compared depending on the direction of the call. (closes issue #13353) Reported by: flefoll Patches: chan_sip.c.br14.139015.patch-refer-pedantic uploaded by flefoll (license 244) ........ ................ 2008-08-26 18:12 +0000 [r140170] Russell Bryant * Makefile, /: Merged revisions 140169 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r140169 | russell | 2008-08-26 13:11:49 -0500 (Tue, 26 Aug 2008) | 4 lines Fix building menuselect-tree with PRINT_DIR set. We _must_ use the --quiet flag here, or else some arbitrary text will end up in the resulting menuselect-tree file and things will explode. ........ 2008-08-25 21:33 +0000 [r139918] Sean Bright * build_tools/get_moduleinfo, /, build_tools/get_makeopts: Merged revisions 139915 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139915 | seanbright | 2008-08-25 17:32:10 -0400 (Mon, 25 Aug 2008) | 17 lines Merged revisions 139909 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139909 | seanbright | 2008-08-25 17:31:03 -0400 (Mon, 25 Aug 2008) | 9 lines Some versions of awk (nawk, for example) don't like empty regular expressions so be slightly more verbose. (closes issue #13374) Reported by: dougm Patches: 13374.diff uploaded by seanbright (license 71) Tested by: dougm ........ ................ 2008-08-25 21:05 +0000 [r139872] Terry Wilson * /, channels/chan_sip.c: Merged revisions 139870 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139870 | twilson | 2008-08-25 15:59:58 -0500 (Mon, 25 Aug 2008) | 10 lines Merged revisions 139869 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139869 | twilson | 2008-08-25 15:46:10 -0500 (Mon, 25 Aug 2008) | 2 lines Make SIPADDHEADER() propagate indefinitely ........ ................ 2008-08-25 16:00 +0000 [r139774] Steve Murphy * main/pbx.c, /, main/features.c: Merged revisions 139770 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139770 | murf | 2008-08-25 09:54:18 -0600 (Mon, 25 Aug 2008) | 17 lines Merged revisions 139764 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139764 | murf | 2008-08-25 09:33:14 -0600 (Mon, 25 Aug 2008) | 9 lines This patch reverts the changes made via 139347, and 139635, as users are seeing adverse difference. I will un-close 13251. Back to the drawing board/ concept/ beginning/ whatever! ........ ................ 2008-08-24 16:30 +0000 [r139705-139708] Tilghman Lesher * /, cdr/cdr_pgsql.c: Merged revisions 139707 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r139707 | tilghman | 2008-08-24 11:26:48 -0500 (Sun, 24 Aug 2008) | 2 lines Memory leak ........ 2008-08-22 22:35 +0000 [r139628-139671] Steve Murphy * /, main/features.c: Merged revisions 139662 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139662 | murf | 2008-08-22 16:32:35 -0600 (Fri, 22 Aug 2008) | 14 lines Merged revisions 139635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139635 | murf | 2008-08-22 16:24:02 -0600 (Fri, 22 Aug 2008) | 6 lines I found some problems with the code I committed earlier, when I merged them into trunk, so I'm coming back to clean up. And, in the process, I found an error in the code I added to trunk and 1.6.x, that I'll fix using this patch also. ........ ................ * apps/app_dial.c, main/pbx.c, /, main/features.c: Merged revisions 139627 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139627 | murf | 2008-08-22 16:03:13 -0600 (Fri, 22 Aug 2008) | 59 lines Merged revisions 139347 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139347 | murf | 2008-08-21 17:03:50 -0600 (Thu, 21 Aug 2008) | 47 lines (closes issue #13251) Reported by: sergee Tested by: murf THis is a bold move for a static release fix, but I wouldn't have made it if I didn't feel confident (at least a *bit* confident) that it wouldn't mess everyone up. The reasoning goes something like this: 1. We simply cannot do anything with CDR's at the current point (in pbx.c, after the __ast_pbx_run loop). It's way too late to have any affect on the CDRs. The CDR is already posted and gone, and the remnants have been cleared. 2. I was very much afraid that moving the running of the 'h' extension down into the bridge code (where it would be now practical to do it), would result in a lot more calls to the 'h' exten, so I implemented it as another exten under another name, but found, to my pleasant surprise, that there was a 1:1 correspondence to the running of the 'h' exten in the pbx_run loop, and the new spot at the end of the bridge. So, I ifdef'd out the current 'h' loop, and moved it into the bridge code. The only difference I can see is the stuff about the AST_PBX_KEEPALIVE, and hopefully, if this is still an important decision point, I can replicate it if there are complaints. To be perfectly honest, the KEEPALIVE situation is not totally clear to me, and how it relates to a post-bridge situation is less clear. I suspect the users will point out everything in total clarity if this steps on anyone's toes! 3. I temporarily swap the bridge_cdr into the channel before running the 'h' exten, which makes it possible for users to edit the cdr before it goes out the door. And, of course, with the endbeforehexten config var set, the users can also get at the billsec/duration vals. After the h exten finishes, the cdr is swapped back and processing continues as normal. Please, all who deal with CDR's, please test this version of Asterisk, and file bug reports as appropriate! ........ I also made a little fix to the app_dial's 'e' option, that is related to my updates. ................ 2008-08-22 20:21 +0000 [r139458-139564] Mark Michelson * include/asterisk/threadstorage.h, /: Merged revisions 139554 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139554 | mmichelson | 2008-08-22 14:45:41 -0500 (Fri, 22 Aug 2008) | 16 lines Merged revisions 139553 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139553 | mmichelson | 2008-08-22 14:45:19 -0500 (Fri, 22 Aug 2008) | 8 lines Fix compilation when DEBUG_THREAD_LOCALS is selected (closes issue #13298) Reported by: snuffy Patches: bug13298_20080822.diff uploaded by snuffy (license 35) ........ ................ * /, channels/chan_iax2.c: Merged revisions 139469 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139469 | mmichelson | 2008-08-22 12:25:12 -0500 (Fri, 22 Aug 2008) | 11 lines Merged revisions 139466 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139466 | mmichelson | 2008-08-22 12:24:47 -0500 (Fri, 22 Aug 2008) | 3 lines Fix the build. Thanks, mvanbaak! ........ ................ * /, channels/chan_iax2.c: Merged revisions 139457 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139457 | mmichelson | 2008-08-22 11:58:21 -0500 (Fri, 22 Aug 2008) | 15 lines Merged revisions 139456 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139456 | mmichelson | 2008-08-22 11:57:38 -0500 (Fri, 22 Aug 2008) | 7 lines Prevent a deadlock in chan_iax2 resulting from incorrect locking order between iax2_pvt and ast_channel structures. AST-13 ........ ................ 2008-08-21 23:46 +0000 [r139400] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 139391 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139391 | jpeeler | 2008-08-21 18:41:50 -0500 (Thu, 21 Aug 2008) | 11 lines Merged revisions 139387 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139387 | jpeeler | 2008-08-21 18:39:31 -0500 (Thu, 21 Aug 2008) | 3 lines Fixes loop that could possibly never exit in the event of a channel never being able to be opened or specify after a restart. (closes issue #11017) ........ ................ 2008-08-21 10:02 +0000 [r139282] Philippe Sultan * /, channels/chan_gtalk.c: Merged revisions 139281 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r139281 | phsultan | 2008-08-21 11:55:31 +0200 (Thu, 21 Aug 2008) | 5 lines Fix two memory leaks in chan_gtalk, thanks Eliel! (closes issue #13310) Reported by: eliel Patches: chan_gtalk.c.patch uploaded by eliel (license 64) ........ 2008-08-20 Kevin P. Fleming * Asterisk 1.6.0-rc3 released. 2008-08-20 22:17 +0000 [r139216] Russell Bryant * apps/app_chanspy.c, /: Merged revisions 139215 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139215 | russell | 2008-08-20 17:16:36 -0500 (Wed, 20 Aug 2008) | 19 lines Merged revisions 139213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139213 | russell | 2008-08-20 17:14:35 -0500 (Wed, 20 Aug 2008) | 11 lines Fix a crash in the ChanSpy application. The issue here is that if you call ChanSpy and specify a spy group, and sit in the application long enough looping through the channel list, you will eventually run out of stack space and the application with exit with a seg fault. The backtrace was always inside of a harmless snprintf() call, so it was tricky to track down. However, it turned out that the call to snprintf() was just the biggest stack consumer in this code path, so it would always be the first one to hit the boundary. (closes issue #13338) Reported by: ruddy ........ ................ 2008-08-20 20:12 +0000 [r139155] Shaun Ruffell * codecs/codec_dahdi.c: Fix bug where the samples were not accurate when in G723 mode, which would cause the timestamp field of the RTP header to be invalid. 2008-08-20 17:30 +0000 [r139104] Steve Murphy * main/cdr.c, /: Merged revisions 139083 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139083 | murf | 2008-08-20 11:25:07 -0600 (Wed, 20 Aug 2008) | 20 lines Merged revisions 139074 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139074 | murf | 2008-08-20 11:14:55 -0600 (Wed, 20 Aug 2008) | 12 lines (closes issue #13263) Reported by: brainy Tested by: murf The specialized reset routine is tromping on the flags field of the CDR. I made a change to not reset the DISABLED bit. This should get rid of this problem. ........ ................ 2008-08-20 15:39 +0000 [r138889-139017] Mark Michelson * /, channels/chan_sip.c: Merged revisions 139016 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139016 | mmichelson | 2008-08-20 10:38:47 -0500 (Wed, 20 Aug 2008) | 14 lines Merged revisions 139015 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139015 | mmichelson | 2008-08-20 10:37:56 -0500 (Wed, 20 Aug 2008) | 6 lines sip_read should properly handle a NULL return from sip_rtp_read. (closes issue #13257) Reported by: travishein ........ ................ * apps/app_chanspy.c: Manually add revision 138887 from trunk to the 1.6.0 branch. I had misunderstood the policy for when to merge to 1.6.0 since it moved to rc status. 2008-08-19 16:38 +0000 [r138846-138847] Steve Murphy * utils/conf2ael.c, /, res/ael/ael.tab.c, res/ael/ael.y, res/ael/ael.tab.h, utils/ael_main.c: Merged revisions 138845 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138845 | murf | 2008-08-19 10:31:24 -0600 (Tue, 19 Aug 2008) | 1 line Oops. put a decl in a generated file. My bad, but fixed now. ........ * main/pbx.c, /, res/ael/ael.tab.c, res/ael/ael.y, res/ael/ael.tab.h: Merged revisions 138815 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138815 | murf | 2008-08-19 09:59:12 -0600 (Tue, 19 Aug 2008) | 19 lines These changes are in regards to bug 13249, where users are being surprised by the changes made to the Set app in trunk/1.6.x, as they come from the 1.4 world. They are only bitten if they write their AEL dialplan in the 1.4 world, and then carry it over to a trunk/1.6.x installation where a "make samples" was executed, or where they hand-edited the asterisk.conf file and added the [compat] category with app_set = 1.6 (or higher). (this commit does not totally solve 13249, at least not yet) The change involves issueing a single warning while the AEL file is loading, if: 1. app_set is present in the config file, and set to 1.6 or higher. 2. there are double quotes in an assignment statement (eg x = "hi there";) 3. the warning was not already issued. The standalone app, aelparse, does not (yet) issue this warning. I'd have to have it read in the asterisk.conf file, and that's a bit of hassle. I'll add it if users request it, tho. ........ 2008-08-19 00:15 +0000 [r138776-138781] Sean Bright * /, channels/chan_sip.c: Merged revisions 138778-138780 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138778 | seanbright | 2008-08-18 20:08:27 -0400 (Mon, 18 Aug 2008) | 1 line While we're at it, make this machine parseable too. ........ r138779 | seanbright | 2008-08-18 20:09:38 -0400 (Mon, 18 Aug 2008) | 1 line And remove code we don't need anymore. ........ r138780 | seanbright | 2008-08-18 20:10:56 -0400 (Mon, 18 Aug 2008) | 1 line Let it compile now, too (woops) ........ * /, channels/chan_sip.c: Merged revisions 138775 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138775 | seanbright | 2008-08-18 19:42:36 -0400 (Mon, 18 Aug 2008) | 3 lines Change event header to RegistrationTime to be more consistent (and avoid breaking existing frameworks). Pointed out by Laureano on #asterisk-dev. ........ 2008-08-18 20:23 +0000 [r138688-138695] Mark Michelson * /, apps/app_queue.c: Merged revisions 138687 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r138687 | mmichelson | 2008-08-18 15:04:10 -0500 (Mon, 18 Aug 2008) | 18 lines Merged revisions 138685 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138685 | mmichelson | 2008-08-18 15:01:14 -0500 (Mon, 18 Aug 2008) | 10 lines Change the inequalities used in app_queue with regards to timeouts from being strict to non-strict for more accuracy. (closes issue #13239) Reported by: atis Patches: app_queue_timeouts_v2.patch uploaded by atis (license 242) ........ ................ 2008-08-18 15:54 +0000 [r138632] Jason Parker * Makefile, /: Merged revisions 138631 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138631 | qwell | 2008-08-18 10:54:07 -0500 (Mon, 18 Aug 2008) | 1 line Remove option that isn't valid here. ........ 2008-08-18 02:14 +0000 [r138519] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 138518 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138518 | jpeeler | 2008-08-17 21:13:04 -0500 (Sun, 17 Aug 2008) | 1 line add missing define for SS7 in dahdi_restart ........ 2008-08-17 14:14 +0000 [r138443-138483] Sean Bright * /, main/features.c: Merged revisions 138482 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138482 | seanbright | 2008-08-17 10:12:11 -0400 (Sun, 17 Aug 2008) | 6 lines Move Uniqueid to the end of the event for those that rely on the position of the name/value pairs, pointed out by snuffy-home on #asterisk-commits. For those of you who rely on the position of name/value pairs in manager events... stop... that is why associative arrays were invented. ........ * /, main/features.c: Merged revisions 138479 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138479 | seanbright | 2008-08-17 09:51:08 -0400 (Sun, 17 Aug 2008) | 7 lines Add Uniqueid header to ParkedCall manager event. (closes issue #13323) Reported by: srt Patches: 13323_unique_id_for_parkedcalls_event.diff uploaded by srt (license 378) ........ * main/rtp.c, /: Merged revisions 138476 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138476 | seanbright | 2008-08-17 09:40:36 -0400 (Sun, 17 Aug 2008) | 7 lines Add missing colons to RTCPReceived and RTCPSent manager events. (closes issue #13319) Reported by: srt Patches: 13319_rtcp_manager_event_headers.diff uploaded by srt (license 378) ........ * /, channels/chan_iax2.c: Merged revisions 138473 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138473 | seanbright | 2008-08-17 09:31:54 -0400 (Sun, 17 Aug 2008) | 7 lines Fix the output of the JitterBufStats manager event. (closes issue #13324) Reported by: srt Patches: 13324_missing_nl_in_jitterbufstats_event_2.diff uploaded by srt (license 378) ........ * configs/cdr_tds.conf.sample, /: Merged revisions 138442 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138442 | seanbright | 2008-08-16 12:40:43 -0400 (Sat, 16 Aug 2008) | 4 lines Since it's introduction in revision 3497, cdr_tds has *never* read the port configuration option from cdr_tds.conf. So go ahead and remove it from the sample config. ........ 2008-08-16 13:07 +0000 [r138410-138413] Tilghman Lesher * channels/chan_dahdi.c, /: Merged revisions 138412 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138412 | tilghman | 2008-08-16 08:07:08 -0500 (Sat, 16 Aug 2008) | 2 lines Fix compilation warnings (found with dev-mode) ........ 2008-08-16 01:14 +0000 [r138333-138362] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 138361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r138361 | jpeeler | 2008-08-15 20:13:26 -0500 (Fri, 15 Aug 2008) | 9 lines Merged revisions 138360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138360 | jpeeler | 2008-08-15 20:12:18 -0500 (Fri, 15 Aug 2008) | 1 line fixes use count to properly decrement if an active dahdi channel is destroyed allowing module to be unloaded ........ ................ * channels/chan_dahdi.c, /: Merged revisions 138311 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r138311 | jpeeler | 2008-08-15 18:46:09 -0500 (Fri, 15 Aug 2008) | 20 lines Merged revisions 138119,138151,138238 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138119 | jpeeler | 2008-08-15 14:21:51 -0500 (Fri, 15 Aug 2008) | 4 lines Fixes the dahdi restart functionality. Dahdi restart allows one to restart all DAHDI channels, even if they are currently in use. This is different from unloading and then loading the module since unloading requires the use count to be zero. Reloading the module is different in that the signalling is not changed from what it was originally configured. Also, this fixes not closing all the file descriptors for D-channels upon module unload (which would prevent loading the module afterwards). (closes issue #11017) ........ r138151 | jpeeler | 2008-08-15 14:41:29 -0500 (Fri, 15 Aug 2008) | 1 line declared static mutexes using AST_MUTEX_DEFINE_STATIC macro ........ r138238 | jpeeler | 2008-08-15 16:28:26 -0500 (Fri, 15 Aug 2008) | 1 line initialize condition variable ss_thread_complete using ast_cond_init ........ ................ 2008-08-15 23:03 +0000 [r138207-138262] Tilghman Lesher * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 138260 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r138260 | tilghman | 2008-08-15 17:54:57 -0500 (Fri, 15 Aug 2008) | 16 lines Merged revisions 138258 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138258 | tilghman | 2008-08-15 17:33:42 -0500 (Fri, 15 Aug 2008) | 8 lines More fixes for realtime peers. (closes issue #12921) Reported by: Nuitari Patches: 20080804__bug12921.diff.txt uploaded by Corydon76 (license 14) 20080815__bug12921.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ ................ * configs/extensions.conf.sample, main/pbx.c, /: Merged revisions 138206 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138206 | tilghman | 2008-08-15 15:35:24 -0500 (Fri, 15 Aug 2008) | 4 lines Remove deprecated syntax from sample config file (closes issue #13314) Reported by: kue ........ 2008-08-15 20:20 +0000 [r138156-138157] Jeff Peeler * channels/chan_dahdi.c: rename all zfd instances in chan_dahdi to dfd to match 1.4 (left over from DAHDI transition) 2008-08-15 15:12 +0000 [r138029] Russell Bryant * main/autoservice.c, /: Merged revisions 138028 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r138028 | russell | 2008-08-15 10:09:46 -0500 (Fri, 15 Aug 2008) | 17 lines Merged revisions 138027 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138027 | russell | 2008-08-15 10:07:16 -0500 (Fri, 15 Aug 2008) | 9 lines Ensure that when a hangup occurs in autoservice, that a hangup frame gets properly deferred to be read from the channel owner when it gets taken out of autoservice. (closes issue #12874) Reported by: dimas Patches: v1-12874.patch uploaded by dimas (license 88) ........ ................ 2008-08-15 15:04 +0000 [r138025] Tilghman Lesher * /, funcs/func_strings.c: Merged revisions 138024 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r138024 | tilghman | 2008-08-15 10:03:32 -0500 (Fri, 15 Aug 2008) | 16 lines Merged revisions 138023 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138023 | tilghman | 2008-08-15 09:51:12 -0500 (Fri, 15 Aug 2008) | 8 lines Additional check for more string specifiers than arguments. (closes issue #13299) Reported by: adomjan Patches: 20080813__bug13299.diff.txt uploaded by Corydon76 (license 14) func_strings.c-sprintf.patch uploaded by adomjan (license 487) Tested by: adomjan ........ ................ 2008-08-14 22:43 +0000 [r137988] Russell Bryant * /, doc/tex/Makefile: Merged revisions 137987 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137987 | russell | 2008-08-14 17:43:15 -0500 (Thu, 14 Aug 2008) | 2 lines Fix a bashism that causes an error when trying to build the pdf on ubuntu ........ 2008-08-14 18:48 +0000 [r137934] Sean Bright * cdr/cdr_sqlite3_custom.c, /: Merged revisions 137933 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137933 | seanbright | 2008-08-14 14:47:28 -0400 (Thu, 14 Aug 2008) | 8 lines Fix memory leak in cdr_sqlite3_custom. (closes issue #13304) Reported by: eliel Patches: sqlite.patch uploaded by eliel (license 64) (Slightly modified by me) ........ 2008-08-14 17:01 +0000 [r137849-137852] Tilghman Lesher * channels/chan_dahdi.c, /: Merged revisions 137848 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r137848 | tilghman | 2008-08-14 11:52:43 -0500 (Thu, 14 Aug 2008) | 17 lines Merged revisions 137847 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r137847 | tilghman | 2008-08-14 11:47:30 -0500 (Thu, 14 Aug 2008) | 9 lines When creating the secondary subchannel name, it is necessary to compare to the existing channel name without the "Zap/" or "DAHDI/" prefix, since our test string is also without that prefix. (closes issue #13027) Reported by: dferrer Patches: chan_zap-1.4.21.1_fix2.patch uploaded by dferrer (license 525) (Slightly modified by me, to compensate for both names) ........ ................ 2008-08-14 Jason Parker * Asterisk 1.6.0-rc2 released. 2008-08-14 15:37 +0000 [r137814] Jason Parker * /, channels/chan_sip.c: Merged revisions 137812 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137812 | qwell | 2008-08-14 10:32:16 -0500 (Thu, 14 Aug 2008) | 8 lines Make sure we set the socket port, so we don't try to use :0. (closes issue #13255) Reported by: falves11 Patches: 13255-socketport.diff uploaded by qwell (license 4) Tested by: falves11 ........ 2008-08-14 15:20 +0000 [r137783] Russell Bryant * /, configs/sip.conf.sample: Merged revisions 137732 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r137732 | russell | 2008-08-14 09:15:50 -0500 (Thu, 14 Aug 2008) | 12 lines Merged revisions 137731 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r137731 | russell | 2008-08-14 09:05:23 -0500 (Thu, 14 Aug 2008) | 4 lines Comments in this config file were aligned only if your tab size was set to 8. So, convert tabs to spaces so that things should be aligned regardless of what tab size you use in your editor. ........ ................ 2008-08-14 15:05 +0000 [r137781] Sean Bright * cdr/cdr_tds.c, /: Merged revisions 137780 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137780 | seanbright | 2008-08-14 11:03:03 -0400 (Thu, 14 Aug 2008) | 8 lines If we detect that we are no longer connected, try to reconnect a few times before giving up. This relies on the timeout settings in the freetds.conf file and, unfortunately, on a recent version of FreeTDS (0.82 or newer). I either need to change the current execs to be non-blocking (which I do not want to do) or we have to force people to run with the latest and greatest of FreeTDS. I'm on the fence... ........ 2008-08-14 02:04 +0000 [r137681] Kevin P. Fleming * /, Zaptel-to-DAHDI.txt: Merged revisions 137680 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r137680 | kpfleming | 2008-08-13 21:03:47 -0500 (Wed, 13 Aug 2008) | 9 lines Merged revisions 137679 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r137679 | kpfleming | 2008-08-13 21:03:04 -0500 (Wed, 13 Aug 2008) | 1 line forgot one module name that changed ........ ................ 2008-08-13 Kevin P. Fleming * Asterisk 1.6.0-rc1 released. 2008-08-13 23:00 +0000 [r137631-137641] Kevin P. Fleming * /, build_tools/prep_tarball: Merged revisions 137640 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137640 | kpfleming | 2008-08-13 18:00:37 -0500 (Wed, 13 Aug 2008) | 1 line make this script actually work ........ * /, Zaptel-to-DAHDI.txt (added), UPGRADE.txt: Merged revisions 137627 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r137627 | kpfleming | 2008-08-13 17:33:32 -0500 (Wed, 13 Aug 2008) | 9 lines Merged revisions 137530 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r137530 | kpfleming | 2008-08-13 15:46:19 -0500 (Wed, 13 Aug 2008) | 1 line add document describing what users will need to be aware of when upgrading to this version and using DAHDI ........ ................ 2008-08-13 21:09 +0000 [r137497-137533] Jason Parker * /, channels/chan_sip.c: Merged revisions 137532 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137532 | qwell | 2008-08-13 16:08:58 -0500 (Wed, 13 Aug 2008) | 8 lines Correctly end locally ended calls. (closes issue #12170) Reported by: pj Patches: 20080702__issue12170_clear_pendinginvite.diff uploaded by bbryant (license 36) Tested by: bbryant, pabelanger ........ * /, apps/app_fax.c: Merged revisions 137496 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137496 | qwell | 2008-08-13 15:05:50 -0500 (Wed, 13 Aug 2008) | 6 lines Add FAXMODE variable with what fax transport was used. (closes issue #13252) Patches: v1-13252.patch uploaded by dimas (license 88) ........ 2008-08-13 14:47 +0000 [r137350-137407] Sean Bright * /, doc/tex/cdrdriver.tex: Merged revisions 137406 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r137406 | seanbright | 2008-08-13 10:41:49 -0400 (Wed, 13 Aug 2008) | 9 lines Merged revisions 137405 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r137405 | seanbright | 2008-08-13 10:33:49 -0400 (Wed, 13 Aug 2008) | 1 line Update docs to reflect the change to cdr_tds ........ ................ * cdr/cdr_tds.c, /: Merged revisions 137403 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137403 | seanbright | 2008-08-13 10:22:47 -0400 (Wed, 13 Aug 2008) | 1 line Use the ast_vasprintf macro instead of vasprintf directly. ........ 2008-08-12 19:48 +0000 [r137300-137302] Russell Bryant * doc/tex/asterisk.tex, /: Merged revisions 137301 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137301 | russell | 2008-08-12 14:48:38 -0500 (Tue, 12 Aug 2008) | 2 lines Grammar hax from Qwell ........ * doc/tex/asterisk.tex, /: Merged revisions 137299 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137299 | russell | 2008-08-12 14:40:35 -0500 (Tue, 12 Aug 2008) | 3 lines Note that developer documentation belongs in doxygen, and not integrated with the user manual stuff in doc/tex/. ........ 2008-08-11 16:15 +0000 [r137240] Russell Bryant * Makefile, /: Merged revisions 137239 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137239 | russell | 2008-08-11 11:14:29 -0500 (Mon, 11 Aug 2008) | 2 lines Make PRINT_DIR work as advertised. ........ 2008-08-11 14:31 +0000 [r137217] Sean Bright * cdr/cdr_tds.c, /, UPGRADE.txt: Merged revisions 137203 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137203 | seanbright | 2008-08-11 10:25:15 -0400 (Mon, 11 Aug 2008) | 7 lines Log the userfield CDR variable like the other CDR backends, assuming the column is actually there. If it's not, we still log everything else as before. (closes issue #13281) Reported by: falves11 ........ 2008-08-11 00:27 +0000 [r137160] Tilghman Lesher * res/res_odbc.c, /: Merged revisions 137150 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r137150 | tilghman | 2008-08-10 19:25:28 -0500 (Sun, 10 Aug 2008) | 13 lines Merged revisions 137138 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r137138 | tilghman | 2008-08-10 19:20:38 -0500 (Sun, 10 Aug 2008) | 5 lines Deallocate database connection handle on disconnect, as we allocate another one on connect. (closes issue #13271) Reported by: dveiga ........ ................ 2008-08-09 15:27 +0000 [r136948] Tilghman Lesher * /, include/asterisk/compat.h, include/asterisk/astobj2.h: Merged revisions 136947 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r136947 | tilghman | 2008-08-09 10:26:27 -0500 (Sat, 09 Aug 2008) | 18 lines Merged revisions 136946 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r136946 | tilghman | 2008-08-09 10:25:36 -0500 (Sat, 09 Aug 2008) | 10 lines Merged revisions 136945 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r136945 | tilghman | 2008-08-09 10:24:36 -0500 (Sat, 09 Aug 2008) | 2 lines Regression fixes for Solaris ........ ................ ................ 2008-08-09 01:16 +0000 [r136860] Tilghman Lesher * /, res/res_agi.c: Merged revisions 136859 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136859 | tilghman | 2008-08-08 20:15:38 -0500 (Fri, 08 Aug 2008) | 4 lines Update documentation as to the behavior of AGI in 1.6.0 and higher. Also, add an OOB message that answers the question of, if AGI no longer shuts down the connection on hangup, how will FastAGI know when to stop processing the call? ........ 2008-08-08 15:33 +0000 [r136785] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 136784 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136784 | mmichelson | 2008-08-08 10:31:31 -0500 (Fri, 08 Aug 2008) | 3 lines Fix compilation for ODBC voicemail ........ 2008-08-08 06:45 +0000 [r136778] Steve Murphy * pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18, pbx/ael/ael-test/ref.ael-test19, pbx/ael/ael-test/ref.ael-vtest13, res/ael/pval.c, /, pbx/ael/ael-test/ref.ael-ntest10, include/asterisk/ael_structs.h, utils/ael_main.c: Merged revisions 136746 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r136746 | murf | 2008-08-07 18:48:35 -0600 (Thu, 07 Aug 2008) | 40 lines Merged revisions 136726 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r136726 | murf | 2008-08-07 18:15:34 -0600 (Thu, 07 Aug 2008) | 32 lines (closes issue #13236) Reported by: korihor Wow, this one was a challenge! I regrouped and ran a new strategy for setting the ~~MACRO~~ value; I set it once per extension, up near the top. It is only set if there is a switch in the extension. So, I had to put in a chunk of code to detect a switch in the pval tree. I moved the code to insert the set of ~~exten~~ up to the beginning of the gen_prios routine, instead of down in the switch code. I learned that I have to push the detection of the switches down into the code, so everywhere I create a new exten in gen_prios, I make sure to pass onto it the values of the mother_exten first, and the exten next. I had to add a couple fields to the exten struct to accomplish this, in the ael_structs.h file. The checked field makes it so we don't repeat the switch search if it's been done. I also updated the regressions. ........ ................ 2008-08-08 02:36 +0000 [r136753] Tilghman Lesher * /: Merged revisions 136751 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136751 | tilghman | 2008-08-07 21:34:17 -0500 (Thu, 07 Aug 2008) | 2 lines Removing bad properties ........ 2008-08-07 23:42 +0000 [r136719-136724] Mark Michelson * apps/app_voicemail.c: This is weird. Either SVN or vim tabbed a bunch of functions over one level during a merge. * apps/app_voicemail.c, /: Merged revisions 136722 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136722 | mmichelson | 2008-08-07 18:39:50 -0500 (Thu, 07 Aug 2008) | 3 lines Remove one last batch of debug messages ........ * apps/app_voicemail.c, /: Merged revisions 136715 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136715 | mmichelson | 2008-08-07 17:25:50 -0500 (Thu, 07 Aug 2008) | 18 lines Merging the imap_consistency_trunk branch to trunk. For an explanation of what "imap_consistency" is, please see svn revision 134223 to the 1.4 branch. Coincidentally, this also fixes a recent bug report regarding the inability to save messages to the new folder when using IMAP storage since they will would be flagged as "seen" and not be recognized as new messages. (closes issue #13234) Reported by: jaroth ........ 2008-08-07 20:41 +0000 [r136672-136674] Shaun Ruffell * codecs/codec_dahdi.c: Removing code that was commented out. * codecs/codec_dahdi.c: Updated codec_dahdi to use the transcoder interface in the DAHDI. (Issue: DAHDI-42) 2008-08-07 20:26 +0000 [r136632-136663] Mark Michelson * /, main/features.c: Merged revisions 136660 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136660 | mmichelson | 2008-08-07 15:25:43 -0500 (Thu, 07 Aug 2008) | 4 lines Bump a LOG_NOTICE message to LOG_DEBUG since it appears once for every bridged call ........ * main/pbx.c, /: Merged revisions 136635 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136635 | mmichelson | 2008-08-07 14:58:32 -0500 (Thu, 07 Aug 2008) | 5 lines Don't allow Answer() to accept a negative argument. Negative argument means an infinite delay and we don't want that. ........ * main/channel.c, /: Merged revisions 136633 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136633 | mmichelson | 2008-08-07 14:54:27 -0500 (Thu, 07 Aug 2008) | 7 lines Fix a calculation error I had made in the poll. The poll would reset to 500 ms every time a non-voice frame was received. The total time we poll should be 500 ms, so now we save the amount of time left after the poll returned and use that as our argument for the next call to poll ........ * main/channel.c, /: Merged revisions 136631 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136631 | mmichelson | 2008-08-07 14:36:46 -0500 (Thu, 07 Aug 2008) | 13 lines Scrap the 500 ms delay when Asterisk auto-answers a channel. Instead, poll the channel until receiving a voice frame. The cap on this poll is 500 ms. The optional delay is still allowable in the Answer() application, but the delay has been moved back to its original position, after the call to the channel's answer callback. The poll for the voice frame will not happen if a delay is specified when calling Answer(). (closes issue #12708) Reported by: kactus ........ 2008-08-07 19:19 +0000 [r136598] Richard Mudgett * channels/misdn_config.c, channels/chan_misdn.c, /, configs/misdn.conf.sample: Merged revisions 136594 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r136594 | rmudgett | 2008-08-07 14:01:03 -0500 (Thu, 07 Aug 2008) | 13 lines Merged revisions 136241 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r136241 | rmudgett | 2008-08-06 16:18:53 -0500 (Wed, 06 Aug 2008) | 5 lines * The allowed_bearers setting in misdn.conf misspelled one of its options: digital_restricted. * Fixed some other spelling errors and typos. ........ ................ 2008-08-07 17:44 +0000 [r136506-136543] Kevin P. Fleming * include/asterisk/doxyref.h, /: Merged revisions 136542 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r136542 | kpfleming | 2008-08-07 12:44:20 -0500 (Thu, 07 Aug 2008) | 6 lines Merged revisions 136541 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ ........ ................ 2008-08-07 16:57 +0000 [r136490] Tilghman Lesher * /, apps/app_queue.c: Merged revisions 136489 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r136489 | tilghman | 2008-08-07 11:55:57 -0500 (Thu, 07 Aug 2008) | 15 lines Merged revisions 136488 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r136488 | tilghman | 2008-08-07 11:50:47 -0500 (Thu, 07 Aug 2008) | 7 lines Update persistent state on all exit conditions. (closes issue #12916) Reported by: sgenyuk Patches: app_queue.patch.txt uploaded by neutrino88 (license 297) Tested by: sgenyuk, aragon ........ ................ 2008-08-06 20:16 +0000 [r136113-136192] Tilghman Lesher * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 136191 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r136191 | tilghman | 2008-08-06 15:15:34 -0500 (Wed, 06 Aug 2008) | 12 lines Merged revisions 136190 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r136190 | tilghman | 2008-08-06 15:14:54 -0500 (Wed, 06 Aug 2008) | 4 lines -C option takes a filename, not a directory path. (closes issue #13007) Reported by: klaus3000 ........ ................ * /, funcs/func_dialgroup.c: Merged revisions 136112 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136112 | tilghman | 2008-08-06 11:58:42 -0500 (Wed, 06 Aug 2008) | 7 lines Persist DIALGROUP() values in astdb (closes issue #13138) Reported by: Corydon76 Patches: 20080725__bug13138.diff.txt uploaded by Corydon76 (license 14) Tested by: pj ........ 2008-08-06 16:00 +0000 [r136064] Mark Michelson * main/rtp.c, /, channels/chan_skinny.c: Merged revisions 136063 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r136063 | mmichelson | 2008-08-06 10:59:29 -0500 (Wed, 06 Aug 2008) | 24 lines Merged revisions 136062 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r136062 | mmichelson | 2008-08-06 10:58:40 -0500 (Wed, 06 Aug 2008) | 16 lines Since adding the AST_CONTROL_SRCUPDATE frame type, there are places where ast_rtp_new_source may be called where the tech_pvt of a channel may not yet have an rtp structure allocated. This caused a crash in chan_skinny, which was fixed earlier, but now the same crash has been reported against chan_h323 as well. It seems that the best solution is to modify ast_rtp_new_source to not attempt to set the marker bit if the rtp structure passed in is NULL. This change to ast_rtp_new_source also allows the removal of what is now a redundant pointer check from chan_skinny. (closes issue #13247) Reported by: pj ........ ................ 2008-08-06 13:59 +0000 [r136006] Olle Johansson * /, res/res_jabber.c: Merged revisions 136005 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136005 | oej | 2008-08-06 15:34:08 +0200 (Ons, 06 Aug 2008) | 6 lines - Formatting - Changing debug messages from VERBOSE to DEBUG channel - Adding a few todo's - Adding a few more "XMPP"'s to compliment Jabber... ........ 2008-08-06 03:56 +0000 [r135901-135951] Tilghman Lesher * main/channel.c, /: Merged revisions 135950 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135950 | tilghman | 2008-08-05 22:55:49 -0500 (Tue, 05 Aug 2008) | 12 lines Merged revisions 135949 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135949 | tilghman | 2008-08-05 22:53:36 -0500 (Tue, 05 Aug 2008) | 4 lines Fix a longstanding bug in channel walking logic, and fix the explanation to make sense. (Closes issue #13124) ........ ................ * /, main/translate.c: Merged revisions 135938 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135938 | tilghman | 2008-08-05 22:29:42 -0500 (Tue, 05 Aug 2008) | 12 lines Merged revisions 135915 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135915 | tilghman | 2008-08-05 22:24:56 -0500 (Tue, 05 Aug 2008) | 4 lines Since powerof() can return an error condition, it's foolhardy not to detect and deal with that condition. (Related to issue #13240) ........ ................ * include/asterisk/threadstorage.h, include/asterisk/utils.h, /: Merged revisions 135900 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135900 | tilghman | 2008-08-05 22:04:01 -0500 (Tue, 05 Aug 2008) | 12 lines Merged revisions 135899 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135899 | tilghman | 2008-08-05 22:02:59 -0500 (Tue, 05 Aug 2008) | 4 lines 1) Bugfix for debugging code 2) Reduce compiler warnings for another section of debugging code (Closes issue #13237) ........ ................ 2008-08-06 00:31 +0000 [r135852] Mark Michelson * include/asterisk/abstract_jb.h, main/channel.c, /, main/abstract_jb.c, main/fixedjitterbuf.h: Merged revisions 135851 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135851 | mmichelson | 2008-08-05 19:30:53 -0500 (Tue, 05 Aug 2008) | 48 lines Merged revisions 135841,135847,135850 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135841 | mmichelson | 2008-08-05 19:25:10 -0500 (Tue, 05 Aug 2008) | 27 lines Merging the issue11259 branch. The purpose of this branch was to take into account "burps" which could cause jitterbuffers to misbehave. One such example is if the L option to Dial() were used to inject audio into a bridged conversation at regular intervals. Since the audio here was not passed through the jitterbuffer, it would cause a gap in the jitterbuffer's timestamps which would cause a frames to be dropped for a brief period. Now ast_generic_bridge will empty and reset the jitterbuffer each time it is called. This causes injected audio to be handled properly. ast_generic_bridge also will empty and reset the jitterbuffer if it receives an AST_CONTROL_SRCUPDATE frame since the change in audio source could negatively affect the jitterbuffer. All of this was made possible by adding a new public API call to the abstract_jb called ast_jb_empty_and_reset. (closes issue #11259) Reported by: plack Tested by: putnopvut ........ r135847 | mmichelson | 2008-08-05 19:27:54 -0500 (Tue, 05 Aug 2008) | 4 lines Revert inadvertent changes to app_skel that occurred when I was testing for a memory leak ........ r135850 | mmichelson | 2008-08-05 19:29:54 -0500 (Tue, 05 Aug 2008) | 3 lines Remove properties that should not be here ........ ................ 2008-08-05 23:52 +0000 [r135822] Steve Murphy * apps/app_dial.c, main/cdr.c, main/channel.c, /, main/features.c, include/asterisk/cdr.h: Merged revisions 135821 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135821 | murf | 2008-08-05 17:45:32 -0600 (Tue, 05 Aug 2008) | 42 lines Merged revisions 135799 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135799 | murf | 2008-08-05 17:13:20 -0600 (Tue, 05 Aug 2008) | 34 lines (closes issue #12982) Reported by: bcnit Tested by: murf I discovered that also, in the previous bug fixes and changes, the cdr.conf 'unanswered' option is not being obeyed, so I fixed this. And, yes, there are two 'answer' times involved in this scenario, and I would agree with you, that the first answer time is the time that should appear in the CDR. (the second 'answer' time is the time that the bridge was begun). I made the necessary adjustments, recording the first answer time into the peer cdr, and then using that to override the bridge cdr's value. To get the 'unanswered' CDRs to appear, I purposely output them, using the dial cmd to mark them as DIALED (with a new flag), and outputting them if they bear that flag, and you are in the right mode. I also corrected one small mention of the Zap device to equally consider the dahdi device. I heavily tested 10-sec-wait macros in dial, and without the macro call; I tested hangups while the macro was running vs. letting the macro complete and the bridge form. Looks OK. Removed all the instrumentation and debug. ........ ................ 2008-08-05 21:38 +0000 [r135749] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 135748 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135748 | tilghman | 2008-08-05 16:37:35 -0500 (Tue, 05 Aug 2008) | 17 lines Merged revisions 135747 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135747 | tilghman | 2008-08-05 16:34:46 -0500 (Tue, 05 Aug 2008) | 9 lines In a conversion to use ast_strlen_zero, the meaning of the flag IAX_HASCALLERID was perverted. This change reverts IAX2 to the original meaning, which was, that the callerid set on the client should be overridden on the server, even if that means the resulting callerid is blank. In other words, if you set "callerid=" in the IAX config, then the callerid should be overridden to blank, even if set on the client. Note that there's a distinction, even on realtime, between the field not existing (NULL in databases) and the field existing, but set to blank (override callerid to blank). ........ ................ 2008-08-05 13:27 +0000 [r135599] Sean Bright * main/cli.c, /: Merged revisions 135598 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135598 | seanbright | 2008-08-05 09:26:34 -0400 (Tue, 05 Aug 2008) | 9 lines Merged revisions 135597 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135597 | seanbright | 2008-08-05 09:25:00 -0400 (Tue, 05 Aug 2008) | 1 line Use PATH_MAX for filenames ........ ................ 2008-08-04 20:15 +0000 [r135538] Russell Bryant * configs/chan_dahdi.conf.sample, /: Merged revisions 135537 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135537 | russell | 2008-08-04 15:15:27 -0500 (Mon, 04 Aug 2008) | 10 lines Merged revisions 135536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135536 | russell | 2008-08-04 15:15:03 -0500 (Mon, 04 Aug 2008) | 2 lines fix a config sample typo ........ ................ 2008-08-04 17:12 +0000 [r135478-135486] Tilghman Lesher * contrib/init.d/rc.mandriva.asterisk (added), Makefile, contrib/init.d/rc.mandrake.asterisk (removed), /, contrib/init.d/rc.mandriva.zaptel (added), contrib/init.d/rc.mandrake.zaptel (removed): Merged revisions 135485 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r135485 | tilghman | 2008-08-04 12:12:15 -0500 (Mon, 04 Aug 2008) | 3 lines Rename Mandrake scripts to Mandriva (Closes issue #13221) ........ * contrib/init.d/rc.mandrake.asterisk, /: Merged revisions 135483 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135483 | tilghman | 2008-08-04 12:08:42 -0500 (Mon, 04 Aug 2008) | 11 lines Merged revisions 135482 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135482 | tilghman | 2008-08-04 12:07:52 -0500 (Mon, 04 Aug 2008) | 2 lines Define ASTSBINDIR for script (Closes issue #13221) ........ ................ * apps/app_voicemail.c, /: Merged revisions 135480 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135480 | tilghman | 2008-08-04 11:58:29 -0500 (Mon, 04 Aug 2008) | 14 lines Merged revisions 135479 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135479 | tilghman | 2008-08-04 11:56:19 -0500 (Mon, 04 Aug 2008) | 6 lines Memory leak on unload (closes issue #13231) Reported by: eliel Patches: app_voicemail.leak.patch uploaded by eliel (license 64) ........ ................ 2008-08-04 16:28 +0000 [r135440-135475] Russell Bryant * configs/chan_dahdi.conf.sample, /: Merged revisions 135474 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135474 | russell | 2008-08-04 11:28:07 -0500 (Mon, 04 Aug 2008) | 10 lines Merged revisions 135473 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135473 | russell | 2008-08-04 11:26:17 -0500 (Mon, 04 Aug 2008) | 2 lines Add a minor clarification to the documentation of mohinterpret and mohsuggest ........ ................ * /, channels/chan_console.c: Merged revisions 135439 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r135439 | russell | 2008-08-04 10:02:12 -0500 (Mon, 04 Aug 2008) | 4 lines Be explicit that we don't want a result from this callback. The callback would never indicate a match, so nothing would have been returned anyway, but it was still a poor example of proper usage. ........ 2008-08-02 05:15 +0000 [r135266] Steve Murphy * main/pbx.c, /: Merged revisions 135265 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r135265 | murf | 2008-08-01 22:51:29 -0600 (Fri, 01 Aug 2008) | 31 lines (closes issue #13202) Reported by: falves11 Tested by: murf falves11 == The changes I introduce here seem to clear up the problem for me. However, if they do not for you, please reopen this bug, and we'll keep digging. The root of this problem seems to be a subtle memory corruption introduced when creating an extension with an empty extension name. While valgrind cannot detect it outside of DEBUG_MALLOC mode, when compiled with DEBUG_MALLOC, this is certain death. The code in main/features.c is a puzzle to me. On the initial module load, the code is attempting to add the parking extension before the features.conf file has even been opened! I just wrapped the offending call with an if() that will not try to add the extension if the extension name is empty. THis seems to solve the corruption, and let the "memory show allocations" work as one would expect. But, really, adding an extension with an empty name is a seriously bad thing to allow, as it will mess up all the pattern matching algorithms, etc. So, I added a statement to the add_extension2 code to return a -1 if this is attempted. in 1.6.0, the changes to only main/pbx.c were applicable, as apparently the code added to main/features by jpeeler were not included in 1.6.0. ........ 2008-08-01 19:30 +0000 [r135198] Sean Bright * channels/chan_mgcp.c, /: Merged revisions 135197 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r135197 | seanbright | 2008-08-01 15:29:26 -0400 (Fri, 01 Aug 2008) | 6 lines Remove some code that used to do something but does not anymore, mainly to get rid of a shadow warning (but this seemed legitimate enough to fix here instead of in my branch). Thanks to putnopvut for taking a look as well. ........ 2008-08-01 17:10 +0000 [r135127-135129] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 135128 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r135128 | tilghman | 2008-08-01 12:09:50 -0500 (Fri, 01 Aug 2008) | 2 lines Picky, picky, buildbot ........ * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 135126 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r135126 | tilghman | 2008-08-01 11:39:51 -0500 (Fri, 01 Aug 2008) | 9 lines SIP should use the transport type set in the Moved Temporarily for the next invite. (closes issue #11843) Reported by: pestermann Patches: 20080723__issue11843_302_ignores_transport_16branch.diff uploaded by bbryant (license 36) 20080723__issue11843_302_ignores_transport_trunk.diff uploaded by bbryant (license 36) Tested by: pabelanger ........ 2008-08-01 14:43 +0000 [r135070] Mark Michelson * apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged revisions 135067-135068 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r135067 | mmichelson | 2008-08-01 09:29:48 -0500 (Fri, 01 Aug 2008) | 13 lines IMAP storage functioned under the assumption that folders such as "Work" and "Family" would be subfolders of the INBOX. This is an invalid assumption to make, but it could be desirable to set up folders in this manner, so a new option for voicemail.conf, "imapparentfolder" has been added to allow for this. (closes issue #13142) Reported by: jaroth Patches: parentfolder.patch uploaded by jaroth (license 50) ........ r135068 | mmichelson | 2008-08-01 09:42:24 -0500 (Fri, 01 Aug 2008) | 3 lines IMAP-specific items must go in IMAP_STORAGE defines... ........ 2008-08-01 12:18 +0000 [r135057-135062] Michiel van Baak * /, apps/app_ices.c: Merged revisions 135059 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135059 | mvanbaak | 2008-08-01 13:47:34 +0200 (Fri, 01 Aug 2008) | 10 lines Merged revisions 135058 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135058 | mvanbaak | 2008-08-01 13:43:46 +0200 (Fri, 01 Aug 2008) | 2 lines make app_ices compile on OpenBSD. ........ ................ * /, channels/chan_skinny.c: Merged revisions 135056 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135056 | mvanbaak | 2008-08-01 13:00:13 +0200 (Fri, 01 Aug 2008) | 16 lines Merged revisions 135055 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135055 | mvanbaak | 2008-08-01 12:55:27 +0200 (Fri, 01 Aug 2008) | 8 lines fix some potential deadlocks in chan_skinny (closes issue #13215) Reported by: qwell Patches: 2008080100_bug13215.diff.txt uploaded by mvanbaak (license 7) Tested by: mvanbaak ........ ................ 2008-07-31 22:34 +0000 [r135034] Kevin P. Fleming * /, main/http.c: Merged revisions 135016 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135016 | kpfleming | 2008-07-31 17:28:42 -0500 (Thu, 31 Jul 2008) | 11 lines Merged revisions 134983 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134983 | kpfleming | 2008-07-31 17:18:11 -0500 (Thu, 31 Jul 2008) | 3 lines accomodate users who seem to lack a sense of humor :-) ........ ................ 2008-07-31 21:58 +0000 [r134926-134981] Tilghman Lesher * sample.call, main/manager.c, pbx/pbx_spool.c, /: Merged revisions 134980 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134980 | tilghman | 2008-07-31 16:55:42 -0500 (Thu, 31 Jul 2008) | 16 lines Blocked revisions 134976 via svnmerge ........ r134976 | tilghman | 2008-07-31 16:53:19 -0500 (Thu, 31 Jul 2008) | 9 lines Specify codecs in callfiles and manager, to allow video calls to be set up from callfiles and AMI. (closes issue #9531) Reported by: Geisj Patches: 20080715__bug9531__1.4.diff.txt uploaded by Corydon76 (license 14) 20080715__bug9531__1.6.0.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ ................ * res/res_config_sqlite.c, /: Merged revisions 134977 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r134977 | tilghman | 2008-07-31 16:53:59 -0500 (Thu, 31 Jul 2008) | 2 lines Switch command order, to meet with current specs ........ 2008-07-31 19:54 +0000 [r134923] Steve Murphy * /, main/features.c: Merged revisions 134922 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134922 | murf | 2008-07-31 13:48:08 -0600 (Thu, 31 Jul 2008) | 63 lines Merged revisions 134883 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134883 | murf | 2008-07-31 13:23:42 -0600 (Thu, 31 Jul 2008) | 51 lines (closes issue #11849) Reported by: greyvoip Tested by: murf OK, a few days of debugging, a bunch of instrumentation in chan_sip, main/channel.c, main/pbx.c, etc. and 5 solid notebook pages of notes later, I have made the small tweek necc. to get the start time right on the second CDR when: A Calls B B answ. A hits Xfer button on sip phone, A dials C and hits the OK button, A hangs up C answers ringing phone B and C converse B and/or C hangs up But does not harm the scenario where: A Calls B B answ. B hits xfer button on sip phone, B dials C and hits the OK button, B hangs up C answers ringing phone A and C converse A and/or C hangs up The difference in start times on the second CDR is because of a Masquerade on the B channel when the xfer number is sent. It ends up replacing the CDR on the B channel with a duplicate, which ends up getting tossed out. We keep a pointer to the first CDR, and update *that* after the bridge closes. But, only if the CDR has changed. I hope this change is specific enough not to muck up any current CDR-based apps. In my defence, I assert that the previous information was wrong, and this change fixes it, and possibly other similar scenarios. I wonder if I should be doing the same thing for the channel, as I did for the peer, but I can't think of a scenario this might affect. I leave it, then, as an exersize for the users, to find the scenario where the chan's CDR changes and loses the proper start time. ........ ................ 2008-07-31 19:41 +0000 [r134918] Russell Bryant * /, apps/app_ices.c: Merged revisions 134917 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134917 | russell | 2008-07-31 14:39:50 -0500 (Thu, 31 Jul 2008) | 17 lines Merged revisions 134915 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134915 | russell | 2008-07-31 14:37:26 -0500 (Thu, 31 Jul 2008) | 9 lines Get app_ices working again (closes issue #12981) Reported by: dlogan Patches: 20080709__app_ices_v2_update_trunk.diff uploaded by bbryant (license 36) 20080709__app_ices_v2_update_14.diff uploaded by bbryant (license 36) Tested by: bbryant ........ ................ 2008-07-31 16:53 +0000 [r134816] Russell Bryant * channels/iax2-parser.c: Merged revisions 134815 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134815 | russell | 2008-07-31 11:50:10 -0500 (Thu, 31 Jul 2008) | 15 lines Merged revisions 134814 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134814 | russell | 2008-07-31 11:45:31 -0500 (Thu, 31 Jul 2008) | 7 lines In case we have some processing threads that free more frames than they allocate, do not let the frame cache grow forever. (closes issue #13160) Reported by: tavius Tested by: tavius, russell ........ ................ 2008-07-31 16:07 +0000 [r134760] Mark Michelson * /, apps/app_queue.c: Merged revisions 134759 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134759 | mmichelson | 2008-07-31 11:05:12 -0500 (Thu, 31 Jul 2008) | 24 lines Merged revisions 134758 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134758 | mmichelson | 2008-07-31 10:56:18 -0500 (Thu, 31 Jul 2008) | 16 lines Add more timeout checks into app_queue, specifically targeting areas where an unknown and potentially long time has just elapsed. Also added a check to try_calling() to return early if the timeout has elapsed instead of potentially setting a negative timeout for the call (thus making it have *no* timeout at all). (closes issue #13186) Reported by: miquel_cabrespina Patches: 13186.diff uploaded by putnopvut (license 60) Tested by: miquel_cabrespina ........ ................ 2008-07-30 22:41 +0000 [r134651-134707] Tilghman Lesher * main/sched.c, /, include/asterisk/sched.h: Merged revisions 134703 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r134703 | tilghman | 2008-07-30 17:38:58 -0500 (Wed, 30 Jul 2008) | 2 lines Oops, wrong define ........ * /, configure, configure.ac: Merged revisions 134650 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134650 | tilghman | 2008-07-30 16:40:08 -0500 (Wed, 30 Jul 2008) | 12 lines Merged revisions 134649 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134649 | tilghman | 2008-07-30 16:38:50 -0500 (Wed, 30 Jul 2008) | 4 lines Qwell pointed out, via IRC, that the previous fix only worked when explicitly set. When nothing is set, and the option is implied, it breaks, because configure sets the prefix to 'NONE'. Fixing. ........ ................ 2008-07-30 21:06 +0000 [r134599] Mark Michelson * /, apps/app_queue.c: Merged revisions 134598 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134598 | mmichelson | 2008-07-30 16:05:37 -0500 (Wed, 30 Jul 2008) | 15 lines Merged revisions 134556 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r134556 | mmichelson | 2008-07-30 15:24:40 -0500 (Wed, 30 Jul 2008) | 7 lines Fix the parsing of the "reason" parameter in the Diversion: header. (closes issue #13195) Reported by: woodsfsg ........ ................ 2008-07-30 20:39 +0000 [r134597] Russell Bryant * /, pbx/pbx_dundi.c: Merged revisions 134596 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134596 | russell | 2008-07-30 15:38:35 -0500 (Wed, 30 Jul 2008) | 14 lines Merged revisions 134595 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134595 | russell | 2008-07-30 15:37:17 -0500 (Wed, 30 Jul 2008) | 6 lines Reduce stack consumption by 12.5% of the max stack size to fix a crash when compiled with LOW_MEMORY. (closes issue #13154) Reported by: edantie ........ ................ 2008-07-30 20:25 +0000 [r134561] Mark Michelson * /, channels/chan_sip.c: Merged revisions 134556 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r134556 | mmichelson | 2008-07-30 15:24:40 -0500 (Wed, 30 Jul 2008) | 7 lines Fix the parsing of the "reason" parameter in the Diversion: header. (closes issue #13195) Reported by: woodsfsg ........ 2008-07-30 19:56 +0000 [r134542] Russell Bryant * funcs/func_curl.c, /: Merged revisions 134541 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134541 | russell | 2008-07-30 14:55:31 -0500 (Wed, 30 Jul 2008) | 12 lines Merged revisions 134540 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134540 | russell | 2008-07-30 14:52:53 -0500 (Wed, 30 Jul 2008) | 4 lines Fix a memory leak in func_curl. Every thread that used this function leaked an allocation the size of a pointer. (reported by jmls in #asterisk-dev) ........ ................ 2008-07-30 19:49 +0000 [r134482-134539] Tilghman Lesher * /, configure, configure.ac: Merged revisions 134538 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134538 | tilghman | 2008-07-30 14:48:37 -0500 (Wed, 30 Jul 2008) | 12 lines Merged revisions 134536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134536 | tilghman | 2008-07-30 14:47:16 -0500 (Wed, 30 Jul 2008) | 4 lines Only override sysconfdir and mandir when prefix=/usr (closes issue #13093) Reported by: pabelanger ........ ................ * /, apps/app_queue.c: Merged revisions 134483 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r134483 | tilghman | 2008-07-30 14:17:38 -0500 (Wed, 30 Jul 2008) | 4 lines Let "roundrobin" also reference rrmemory, for the 1.6 release (as described in UPGRADE-1.4.txt) (Closes issue #13181) ........ * /, res/res_agi.c: Merged revisions 134481 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134481 | tilghman | 2008-07-30 14:05:35 -0500 (Wed, 30 Jul 2008) | 13 lines Merged revisions 134480 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134480 | tilghman | 2008-07-30 14:03:44 -0500 (Wed, 30 Jul 2008) | 5 lines launch_netscript sometimes returns -1, which fails to set AGISTATUS. Map failure to -1, so that AGISTATUS is always set. (closes issue #13199) Reported by: smw1218 ........ ................ 2008-07-30 18:33 +0000 [r134477] Mark Michelson * /, main/app.c: Merged revisions 134476 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134476 | mmichelson | 2008-07-30 13:33:12 -0500 (Wed, 30 Jul 2008) | 12 lines Merged revisions 134475 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134475 | mmichelson | 2008-07-30 13:31:47 -0500 (Wed, 30 Jul 2008) | 4 lines Fix a spot where a function could return without bringing a channel out of autoservice. ........ ................ 2008-07-30 15:34 +0000 [r134356] Kevin P. Fleming * Makefile, /: Merged revisions 134355 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134355 | kpfleming | 2008-07-30 10:32:14 -0500 (Wed, 30 Jul 2008) | 10 lines Merged revisions 134352 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134352 | kpfleming | 2008-07-30 10:29:17 -0500 (Wed, 30 Jul 2008) | 2 lines use the proper method for building version.h ........ ................ 2008-07-29 22:29 +0000 [r134283] Kevin P. Fleming * apps/app_rpt.c, apps/app_dahdibarge.c, channels/chan_dahdi.c, /, apps/app_meetme.c, apps/app_dahdiscan.c, apps/app_dahdiras.c: Merged revisions 134260 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r134260 | kpfleming | 2008-07-29 17:22:13 -0500 (Tue, 29 Jul 2008) | 2 lines build against the now-typedef-free dahdi/user.h, and remove some #ifdefs for features that will always be present in DAHDI ........ 2008-07-28 22:16 +0000 [r134164] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 134163 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134163 | tilghman | 2008-07-28 17:07:12 -0500 (Mon, 28 Jul 2008) | 15 lines Merged revisions 134161 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134161 | tilghman | 2008-07-28 16:50:50 -0500 (Mon, 28 Jul 2008) | 7 lines Detect when sox fails to raise the volume, because sox can't read the file. (closes issue #12939) Reported by: rickbradley Patches: 20080728__bug12939.diff.txt uploaded by Corydon76 (license 14) Tested by: rickbradley ........ ................ 2008-07-28 19:55 +0000 [r134126] Mark Michelson * /, configure, main/Makefile, configure.ac, CHANGES: Merged revisions 134125 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r134125 | mmichelson | 2008-07-28 14:53:56 -0500 (Mon, 28 Jul 2008) | 27 lines This commit compensates for buggy poll(2) implementations. Asterisk has, for a long time, had its own implementation of poll(2) which just used the input arguments to call select(2). In 1.4, this internal implementation was used for Darwin systems. This was removed in Asterisk trunk at some point, but it seems as though this was not the right move to make. On Mac OS X, it appears as though the poll used to gather CLI input does not respond properly when connecting via a remote Asterisk console. Reverting to the use of Asterisk's poll fixed the issue. Also, there is now an option for the configure script, --enable-internal-poll, which will allow for anyone to use Asterisk's internal poll implementation in case they suspect that their system's poll implementation is buggy. closes issue #11928) Reported by: adriavidal Patches: 1.6.0-configurev2.patch uploaded by putnopvut (license 60) ........ 2008-07-28 16:49 +0000 [r134087] Kevin P. Fleming * apps/app_parkandannounce.c, main/loader.c, sample.call, contrib/scripts/autosupport, build_tools/cflags.xml, main/channel.c, apps/app_dahdibarge.c, channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, doc/ss7.txt, /, main/features.c, doc/osp.txt, main/file.c, pbx/pbx_config.c: Merged revisions 134086 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r134086 | kpfleming | 2008-07-28 11:42:00 -0500 (Mon, 28 Jul 2008) | 3 lines remove remaining Zaptel references in various places ........ 2008-07-28 16:13 +0000 [r134052] Mark Michelson * apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c, /, apps/app_meetme.c, apps/app_dahdiscan.c: Merged revisions 134050 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r134050 | mmichelson | 2008-07-28 11:00:19 -0500 (Mon, 28 Jul 2008) | 3 lines merging the zap_and_dahdi_trunk branch up to trunk ........ 2008-07-26 15:34 +0000 [r133942-133982] Russell Bryant * main/asterisk.c, include/asterisk/doxyref.h, /: Include the licensing page in 1.6.0 as well. Now, this page exists in 1.4, trunk, and 1.6.0. * /: unblock 133575 * /, main/devicestate.c: Merged revisions 133945-133946 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r133945 | russell | 2008-07-26 10:15:14 -0500 (Sat, 26 Jul 2008) | 6 lines ast_device_state() gets called in two different ways. The first way is when called from elsewhere in Asterisk to find the current state of a device. In that case, we want to use the cached value if it exists. The other way is when processing a device state change. In that case, we do not want to check the cache because returning the last known state is counter productive. ........ r133946 | russell | 2008-07-26 10:16:20 -0500 (Sat, 26 Jul 2008) | 1 line actually use the cache_cache argument ........ 2008-07-25 22:09 +0000 [r133863-133905] Tilghman Lesher * contrib/scripts/asterisk.ldif, /: Merged revisions 133902 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r133902 | tilghman | 2008-07-25 16:59:39 -0500 (Fri, 25 Jul 2008) | 6 lines Update version (closes issue #13163) Reported by: suretec Patches: asterisk.ldif uploaded by suretec (license 70) ........ 2008-07-25 19:37 +0000 [r133804-133806] Brandon Kruse * /: Blocking revert of code changes in r133770 * main/http.c: Include the http_decode function from trunk to replace the + with a space. 2008-07-25 17:33 +0000 [r133694] Brandon Kruse * /: Blocking a fix from trunk for the function http_decode. 1.6.0 does not have this function. 2008-07-25 17:26 +0000 [r133671] Tilghman Lesher * main/channel.c, /, channels/chan_agent.c, main/devicestate.c: Merged revisions 133665 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r133665 | tilghman | 2008-07-25 12:24:43 -0500 (Fri, 25 Jul 2008) | 16 lines Merged revisions 133649 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133649 | tilghman | 2008-07-25 12:19:39 -0500 (Fri, 25 Jul 2008) | 8 lines Fix some errant device states by making the devicestate API more strict in terms of the device argument (only without the unique identifier appended). (closes issue #12771) Reported by: davidw Patches: 20080717__bug12771.diff.txt uploaded by Corydon76 (license 14) Tested by: davidw, jvandal, murf ........ ................ 2008-07-25 15:01 +0000 [r133576-133580] Russell Bryant * /, LICENSE: Merged revisions 133579 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r133579 | russell | 2008-07-25 10:00:49 -0500 (Fri, 25 Jul 2008) | 18 lines Merged revisions 133578 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r133578 | russell | 2008-07-25 10:00:31 -0500 (Fri, 25 Jul 2008) | 10 lines Merged revisions 133577 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r133577 | russell | 2008-07-25 10:00:13 -0500 (Fri, 25 Jul 2008) | 2 lines Fix the IAX2 URI for calling Digium ........ ................ ................ 2008-07-25 14:41 +0000 [r133571-133574] Mark Michelson * /, channels/chan_sip.c: Merged revisions 133573 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r133573 | mmichelson | 2008-07-25 09:40:52 -0500 (Fri, 25 Jul 2008) | 15 lines Merged revisions 133572 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133572 | mmichelson | 2008-07-25 09:40:10 -0500 (Fri, 25 Jul 2008) | 7 lines We need to make sure to null-terminate the "name" portion of SIP URI parameters so that there are no bogus comparisons. Thanks to bbryant for pointing this out. ........ ................ 2008-07-25 13:25 +0000 [r133567-133569] Russell Bryant * /, channels/chan_sip.c: Merged revisions 133568 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r133568 | russell | 2008-07-25 08:01:59 -0500 (Fri, 25 Jul 2008) | 4 lines Minor coding guidelines tweaks ... - Use ast_strlen_zero in one place - check for successful string comparison the way most of Asterisk code does it ........ 2008-07-24 21:31 +0000 [r133524] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 133509 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r133509 | tilghman | 2008-07-24 16:27:06 -0500 (Thu, 24 Jul 2008) | 11 lines Merged revisions 133488 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133488 | tilghman | 2008-07-24 16:17:55 -0500 (Thu, 24 Jul 2008) | 3 lines Fix rtautoclear and rtcachefriends (Closes issue #12707) ........ ................ 2008-07-24 20:41 +0000 [r133487] Russell Bryant * /, channels/chan_agent.c: Merged revisions 133486 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r133486 | russell | 2008-07-24 15:40:15 -0500 (Thu, 24 Jul 2008) | 3 lines I made this change from DEVICE_STATE to DEVICE_STATE_CHANGE, but I had it backwards, this is the right event to subscribe to ... ........ 2008-07-24 19:55 +0000 [r133449] Mark Michelson * /, main/logger.c: Merged revisions 133448 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r133448 | mmichelson | 2008-07-24 14:53:37 -0500 (Thu, 24 Jul 2008) | 12 lines Print the correct PID in log messages. Prior to this commit, only the logger thread's PID would be printed. (closes issue #13150) Reported by: atis Patches: log_pid.diff uploaded by putnopvut (license 60) Tested by: eliel ........ 2008-07-24 05:21 +0000 [r133392-133405] Tilghman Lesher * contrib/scripts/asterisk.logrotate, Makefile, /: Merged revisions 133400 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r133400 | tilghman | 2008-07-24 00:21:00 -0500 (Thu, 24 Jul 2008) | 3 lines Build the logrotate script according to paths (Closes issue #13147) ........ * Makefile, /: Merged revisions 133391 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r133391 | tilghman | 2008-07-23 23:51:42 -0500 (Wed, 23 Jul 2008) | 3 lines Optionally install logrotate file (Closes issue #13148) ........ 2008-07-23 22:07 +0000 [r133300] Steve Murphy * main/pbx.c, /: Merged revisions 133299 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r133299 | murf | 2008-07-23 16:03:48 -0600 (Wed, 23 Jul 2008) | 27 lines (closes issue #13144) Reported by: murf Tested by: murf For: J. Geis The 'data' field in the ast_exten struct was being 'moved' from the current dialplan to the replacement dialplan. This was not good, as the current dialplan could have problems in the time between the change and when the new dialplan is swapped in. So, I modified the merge_and_delete code to strdup the 'data' field (the args to the app call), and then it's freed as normal. I improved a few messages; I added code to limit the number of calls to the context_merge_incls_swits_igps_other_registrars() to one per context. I don't think having it called multiple times per context was doing anything bad, but it was inefficient. I hope this fixes the problems Mr. Geiss was noting in asterisk-users, see http://lists.digium.com/pipermail/asterisk-users/2008-July/215634.html ........ 2008-07-23 21:50 +0000 [r133297] Jason Parker * channels/chan_dahdi.c, /: Merged revisions 133296 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r133296 | qwell | 2008-07-23 16:50:20 -0500 (Wed, 23 Jul 2008) | 9 lines Merged revisions 133295 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133295 | qwell | 2008-07-23 16:49:03 -0500 (Wed, 23 Jul 2008) | 1 line inbandrelease is gone - it's now inbanddisconnect ........ ................ 2008-07-23 20:39 +0000 [r133218] Brett Bryant * /, channels/chan_sip.c: Merged revisions 133197 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r133197 | bbryant | 2008-07-23 15:33:22 -0500 (Wed, 23 Jul 2008) | 2 lines Fix issue where tcp in sip is enabled by default, despite what it says in the config sample file. Also fix "sip show settings" for tcp connections. ........ 2008-07-23 19:50 +0000 [r133042-133172] Mark Michelson * apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c, /: Merged revisions 133171 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r133171 | mmichelson | 2008-07-23 14:48:03 -0500 (Wed, 23 Jul 2008) | 20 lines Merged revisions 133169 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133169 | mmichelson | 2008-07-23 14:39:47 -0500 (Wed, 23 Jul 2008) | 12 lines As suggested by seanbright, the PSEUDO_CHAN_LEN in app_chanspy should be set at load time, not at compile time, since dahdi_chan_name is determined at load time. Also changed the next_unique_id_to_use to have the static qualifier. Also added the dahdi_chan_name_len variable so that strlen(dahdi_chan_name) isn't necessary. Thanks to seanbright for the suggestion. ........ ................ * apps/app_chanspy.c, /: Merged revisions 133106 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r133106 | mmichelson | 2008-07-23 14:07:56 -0500 (Wed, 23 Jul 2008) | 13 lines Merged revisions 133104 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133104 | mmichelson | 2008-07-23 14:06:16 -0500 (Wed, 23 Jul 2008) | 5 lines Zap/pseudo is ten characters, but DAHDI/pseudo is twelve. The strncmp call in next_channel should account for this. ........ ................ * apps/app_chanspy.c, /: Merged revisions 133102 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r133102 | mmichelson | 2008-07-23 13:58:37 -0500 (Wed, 23 Jul 2008) | 14 lines Merged revisions 133101 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133101 | mmichelson | 2008-07-23 13:57:17 -0500 (Wed, 23 Jul 2008) | 6 lines Update the "last" channel in next_channel in app_chanspy so that the same pseudo channel isn't constantly returned. related to issue #13124 ........ ................ * channels/chan_dahdi.c, /: Merged revisions 133041 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r133041 | mmichelson | 2008-07-23 12:54:03 -0500 (Wed, 23 Jul 2008) | 15 lines Merged revisions 133038 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133038 | mmichelson | 2008-07-23 12:50:01 -0500 (Wed, 23 Jul 2008) | 7 lines Small cleanup. Move the declaration of the DAHDI_SPANINFO variable to the block where it is used. This allows one less #ifdef HAVE_PRI to clutter things up. Thanks to Tzafrir for pointing this out on #asterisk-dev ........ ................ 2008-07-23 17:21 +0000 [r132978-132983] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 132981 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132981 | tilghman | 2008-07-23 12:20:43 -0500 (Wed, 23 Jul 2008) | 6 lines Yet another conversion of '|' to ',' (closes issue #13137) Reported by: eliel Patches: chan_iax2trunk-IAXPEER.patch uploaded by eliel (license 64) ........ * contrib/scripts/asterisk.logrotate (added), /: Merged revisions 132977 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132977 | tilghman | 2008-07-23 12:14:56 -0500 (Wed, 23 Jul 2008) | 6 lines Add logrotate script for Asterisk (closes issue #13085) Reported by: pabelanger Patches: logrotate uploaded by pabelanger (license 224) ........ 2008-07-23 16:42 +0000 [r132965-132967] Kevin P. Fleming * channels/misdn/isdn_lib.c, /: Merged revisions 132883,132966 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132883 | crichter | 2008-07-23 07:07:15 -0500 (Wed, 23 Jul 2008) | 9 lines Merged revisions 132826 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132826 | crichter | 2008-07-23 13:37:50 +0200 (Mi, 23 Jul 2008) | 1 line another Fix because of r119585, this commit has broken high frequented BRI Ports, there was a possibility that a channel, that was marked as in_use would be reused later, the corresponding port could got stuck then. So it is recommended to upgrade for chan_misdn users. ........ ................ r132966 | kpfleming | 2008-07-23 11:38:28 -0500 (Wed, 23 Jul 2008) | 2 lines use correct function name... please compile with --enable-dev-mode ................ * include/asterisk/stringfields.h, /, main/utils.c: Merged revisions 132964 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132964 | kpfleming | 2008-07-23 11:30:18 -0500 (Wed, 23 Jul 2008) | 10 lines Merged revisions 132872 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132872 | kpfleming | 2008-07-23 06:52:18 -0500 (Wed, 23 Jul 2008) | 2 lines minor optimization for stringfields: when a field is being set to a larger value than it currently contains and it happens to be the most recent field allocated from the currentl pool, it is possible to 'grow' it without having to waste the space it is currently using (or potentially even allocate a new pool) ........ ................ 2008-07-23 08:18 +0000 [r132824] Olle Johansson * /, channels/chan_sip.c: Merged revisions 132823 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132823 | oej | 2008-07-23 10:13:07 +0200 (Ons, 23 Jul 2008) | 8 lines Well, the content of a channel variable may be longer than the size of a pointer... Thanks, eliel! Reported by: eliel Patches: chan_siptrunk.SIPPEER.patch uploaded by eliel (license 64) (closes issue #13135) ........ 2008-07-22 22:20 +0000 [r132797] Mark Michelson * /, channels/chan_sip.c: Merged revisions 132795 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132795 | mmichelson | 2008-07-22 17:17:09 -0500 (Tue, 22 Jul 2008) | 11 lines Merged revisions 132777 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ Allow Spiraled INVITEs to work correctly within Asterisk. Prior to this change, a spiraled INVITE would cause a 482 Loop Detected to be sent to the caller. With this change, if a potential loop is detected, the Request-URI is inspected to see if it has changed from what was originally received. If pedantic mode is on, then this inspection is fully RFC 3261 compliant. If pedantic mode is not on, then a string comparison is used to test the equality of the two R-URIs. This has been tested by using OpenSER to rewrite the R-URI and send the INVITE back to Asterisk. (closes issue #7403) Reported by: stephen_dredge Modified: branches/1.4/channels/chan_sip.c ........ ................ 2008-07-22 22:15 +0000 [r132793] Kevin P. Fleming * channels/chan_dahdi.c, /: Merged revisions 132791 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132791 | kpfleming | 2008-07-22 17:14:37 -0500 (Tue, 22 Jul 2008) | 2 lines correct fix made in r132777... the code *did* compile in dev-mode, as long as libpri was installed and enabled ........ 2008-07-22 21:59 +0000 [r132782] Olle Johansson * /, channels/chan_sip.c, doc/sip-retransmit.txt (added): Merged revisions 132703 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132703 | oej | 2008-07-22 22:46:11 +0200 (Tis, 22 Jul 2008) | 17 lines Merged revisions 132645 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132645 | oej | 2008-07-22 22:10:26 +0200 (Tis, 22 Jul 2008) | 9 lines The most common question on the #asterisk iRC channel and on mailing lists seems to be in regards to an error message when retransmit fails. This is frequently misunderstood as a failure of Asterisk, not a failure of the network to reach the other party. This document tries to assist the Asterisk user in sorting out these issues by explaining the logic and pointing at some possible causes. Hopefully, we will get other questions now :-) ........ ................ 2008-07-22 21:55 +0000 [r132780] Tilghman Lesher * configs/iax.conf.sample, /, channels/chan_iax2.c: Merged revisions 132778 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132778 | tilghman | 2008-07-22 16:53:40 -0500 (Tue, 22 Jul 2008) | 18 lines Merged revisions 132713 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r132713 | tilghman | 2008-07-22 16:19:39 -0500 (Tue, 22 Jul 2008) | 10 lines Merged revisions 132711 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r132711 | tilghman | 2008-07-22 16:14:10 -0500 (Tue, 22 Jul 2008) | 2 lines Fixes for AST-2008-010 and AST-2008-011 ........ ................ ................ 2008-07-22 21:54 +0000 [r132779] Mark Michelson * channels/chan_dahdi.c, /: Merged revisions 132777 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132777 | mmichelson | 2008-07-22 16:52:24 -0500 (Tue, 22 Jul 2008) | 3 lines Get chan_dahdi to compile in devmode ........ 2008-07-22 21:23 +0000 [r132574-132729] Kevin P. Fleming * channels/chan_dahdi.c, /: Merged revisions 132721 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132721 | kpfleming | 2008-07-22 16:21:56 -0500 (Tue, 22 Jul 2008) | 14 lines Merged revisions 132712 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132712 | kpfleming | 2008-07-22 16:17:23 -0500 (Tue, 22 Jul 2008) | 6 lines ensure that if any alarms exist at channel creation time, they are handled identically to if they occurred later, so that later alarm clearing will work properly and 'make sense' (closes issue #12160) Reported by: tzafrir ........ ................ * /, configure, configure.ac, acinclude.m4: Merged revisions 132705 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132705 | kpfleming | 2008-07-22 15:54:07 -0500 (Tue, 22 Jul 2008) | 10 lines Merged revisions 132704 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132704 | kpfleming | 2008-07-22 15:49:41 -0500 (Tue, 22 Jul 2008) | 2 lines make AST_C_COMPILE_CHECK able to print a 'pretty' description of what it is doing ........ ................ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 132643 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132643 | kpfleming | 2008-07-22 14:59:10 -0500 (Tue, 22 Jul 2008) | 10 lines Merged revisions 132641 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132641 | kpfleming | 2008-07-22 14:49:11 -0500 (Tue, 22 Jul 2008) | 2 lines use renamed libpri API call for controlling this feature (was improperly named before) ........ ................ * channels/chan_dahdi.c, /: Merged revisions 132573 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132573 | kpfleming | 2008-07-21 17:51:16 -0500 (Mon, 21 Jul 2008) | 10 lines Merged revisions 132571 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132571 | kpfleming | 2008-07-21 17:45:16 -0500 (Mon, 21 Jul 2008) | 2 lines teach chan_dahdi how to find the D-channel on BRI spans, and don't attempt to use channel 24 as a D-channel on spans of unexpected sizes ........ ................ 2008-07-21 21:13 +0000 [r132515] Brett Bryant * configs/features.conf.sample, configs/gtalk.conf.sample, /, configs/jingle.conf.sample, configs/manager.conf.sample: Merged revisions 132514 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132514 | bbryant | 2008-07-21 16:12:51 -0500 (Mon, 21 Jul 2008) | 8 lines Update configuration files to add missing options for jingle, gtalk, manager.conf, and features.conf. (closes issue #13128) Reported by: caio1982 Patches: missing_options1.diff uploaded by caio1982 (license 22) ........ 2008-07-21 21:02 +0000 [r132512-132513] Tilghman Lesher * main/fskmodem.c (added), /, include/asterisk/fskmodem.h (added): Merged revisions 132511 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132511 | tilghman | 2008-07-21 16:00:47 -0500 (Mon, 21 Jul 2008) | 2 lines (Step 2 of 2) ........ * main/fskmodem.c (removed), include/asterisk/fskmodem_int.h (added), build_tools/cflags.xml, main/fskmodem_float.c (added), /, main/tdd.c, include/asterisk/fskmodem.h (removed), main/fskmodem_int.c (added), main/callerid.c, include/asterisk/fskmodem_float.h (added): Merged revisions 132510 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132510 | tilghman | 2008-07-21 15:59:03 -0500 (Mon, 21 Jul 2008) | 5 lines Optionally build integer-based routines for FSK tone decoding (but default to the more accurate float-based routines). (Closes issue #11679) (Step 1 of 2) ........ 2008-07-21 20:55 +0000 [r132467-132509] Brett Bryant * /, apps/app_sendtext.c: Merged revisions 132508 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132508 | bbryant | 2008-07-21 15:54:09 -0500 (Mon, 21 Jul 2008) | 9 lines Fix a bug where SENDTEXTSTATUS isn't set properly when it isn't supported on a channel (yet _another_ useful patch by eliel). (closes issue #13081) Reported by: eliel Patches: app_sendtext.c.patch uploaded by eliel (license 64) Tested by: eliel ........ * /, channels/chan_sip.c: Merged revisions 132468 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132468 | bbryant | 2008-07-21 12:42:45 -0500 (Mon, 21 Jul 2008) | 5 lines Fix bug where ast_parse_arg would inadvertantly enable sip tcp when parsing a tcpbindaddr if it was disabled. (closes issue #13117) Reported by: pj ........ * /, channels/chan_iax2.c: Merged revisions 132466 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132466 | bbryant | 2008-07-21 12:22:02 -0500 (Mon, 21 Jul 2008) | 3 lines Fix an issue in iax2 where a call that's been rejected still kept an open channel on the side that attempted to make the call (not the side of the call that rejected the call). Changes were load tested and also approved by Russell. ........ 2008-07-21 15:34 +0000 [r132426] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 132425 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132425 | jpeeler | 2008-07-21 10:33:13 -0500 (Mon, 21 Jul 2008) | 2 lines make buffers config option (chan_dahdi.conf) parsing safer and added logging in case of failure ........ 2008-07-21 14:48 +0000 [r132389-132391] Russell Bryant * apps/app_jack.c, include/asterisk/libresample.h (removed), /, build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, main/Makefile, main/libresample (removed), configure.ac, codecs/codec_resample.c, makeopts.in: Merged revisions 132390 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132390 | russell | 2008-07-21 09:47:41 -0500 (Mon, 21 Jul 2008) | 16 lines Remove libresample from the Asterisk source tree. It is now available in its own repository, and must be installed like any other library for Asterisk to use. The two modules that require it are codec_resample and app_jack. To install libresample: $ svn co http://svn.digium.com/svn/libresample/trunk libresample $ cd libresample $ ./configure $ make $ sudo make install This code is currently in our own repository because the build system did not include the appropriate targets for building a dynamic library or for installing the library. ........ * apps/app_jack.c, /, codecs/codec_resample.c: Merged revisions 132388 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132388 | russell | 2008-07-21 08:51:05 -0500 (Mon, 21 Jul 2008) | 3 lines Enable higher quality resampling, as it doesn't have a noticeable performance impact on my machine .. ........ 2008-07-19 16:47 +0000 [r132313] Kevin P. Fleming * /, LICENSE: Merged revisions 132312 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132312 | kpfleming | 2008-07-19 11:46:33 -0500 (Sat, 19 Jul 2008) | 10 lines Merged revisions 132311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132311 | kpfleming | 2008-07-19 11:45:52 -0500 (Sat, 19 Jul 2008) | 2 lines grant a license exception to allow distribution of Asterisk binaries that use the UW IMAP Toolkit (which is licensed under a non-GPL-compatible license) ........ ................ 2008-07-19 10:47 +0000 [r132278] Michiel van Baak * res/res_config_sqlite.c, /: Merged revisions 132277 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132277 | mvanbaak | 2008-07-19 12:46:12 +0200 (Sat, 19 Jul 2008) | 7 lines fix a couple of comments in sqlite resource driver. (closes issue #13110) Reported by: gknispel_proformatique Patches: res_config_sqlite_comments.patch uploaded by gknispel (license 261) ........ 2008-07-18 22:20 +0000 [r132245] Brett Bryant * main/manager.c, /: Merged revisions 132242 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132242 | bbryant | 2008-07-18 17:19:56 -0500 (Fri, 18 Jul 2008) | 4 lines Fixes problem where manager users loaded from users.conf would be removed early (before the routine to load the configuration was finished) because a variable wasn't initialized. ........ 2008-07-18 20:58 +0000 [r132114-132207] Tilghman Lesher * /, main/say.c: Merged revisions 132113 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132113 | tilghman | 2008-07-18 14:09:39 -0500 (Fri, 18 Jul 2008) | 14 lines Merged revisions 132112 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132112 | tilghman | 2008-07-18 14:06:10 -0500 (Fri, 18 Jul 2008) | 6 lines Fix for Taiwanese number syntax (closes issue #12319) Reported by: CharlesWang Patches: saynumber-tw-1.4.18.1.patch uploaded by CharlesWang (license 444) ........ ................ 2008-07-18 18:53 +0000 [r132111] Matthew Fredrickson * channels/chan_dahdi.c, /: Merged revisions 132108 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132108 | mattf | 2008-07-18 13:50:00 -0500 (Fri, 18 Jul 2008) | 1 line Make sure we break the poll so that messages queued will be sent on the SS7 when using CLI commands for blocking and blocking of CICs and linksets. ........ 2008-07-18 18:51 +0000 [r132110] Tilghman Lesher * main/config.c, /: Merged revisions 132109 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132109 | tilghman | 2008-07-18 13:50:37 -0500 (Fri, 18 Jul 2008) | 14 lines Merged revisions 132107 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132107 | tilghman | 2008-07-18 13:47:50 -0500 (Fri, 18 Jul 2008) | 6 lines Textual clarification (closes issue #13106) Reported by: flefoll Patches: config.c.br14.120173.patch-unknown-directive uploaded by flefoll (license 244) ........ ................ 2008-07-18 17:56 +0000 [r132051] Brett Bryant * main/hashtab.c, /, cdr/cdr_radius.c: Merged revisions 132050 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132050 | bbryant | 2008-07-18 12:55:41 -0500 (Fri, 18 Jul 2008) | 8 lines Fix magic Revision keywords in hashtab.c and change cdr_radius.c to use the same keyword as the other files (patch by eliel). (closes issue #13104) Reported by: eliel Patches: revision.patch uploaded by eliel (license 64) ........ 2008-07-18 17:40 +0000 [r131984-132047] Tilghman Lesher * main/sched.c, /: Merged revisions 131989 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131989 | tilghman | 2008-07-18 12:10:34 -0500 (Fri, 18 Jul 2008) | 10 lines Merged revisions 131988 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131988 | tilghman | 2008-07-18 12:10:01 -0500 (Fri, 18 Jul 2008) | 2 lines Oops ........ ................ * main/sched.c, /, include/asterisk/sched.h: Merged revisions 131986 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131986 | tilghman | 2008-07-18 11:48:18 -0500 (Fri, 18 Jul 2008) | 10 lines Merged revisions 131985 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131985 | tilghman | 2008-07-18 11:46:23 -0500 (Fri, 18 Jul 2008) | 2 lines Preserve ABI compatibility with last change ........ ................ * main/sched.c, /, include/asterisk/sched.h, channels/chan_iax2.c: Merged revisions 131982 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131982 | tilghman | 2008-07-18 11:33:56 -0500 (Fri, 18 Jul 2008) | 10 lines Merged revisions 131970 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131970 | tilghman | 2008-07-18 11:30:31 -0500 (Fri, 18 Jul 2008) | 2 lines Make the ast_assert call within ast_sched_del report something useful. ........ ................ 2008-07-18 16:16 +0000 [r131924] Kevin P. Fleming * main/dlfcn.c (removed), main/loader.c, /, main/Makefile, include/asterisk/dlfcn-compat.h (removed): Merged revisions 131923 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131923 | kpfleming | 2008-07-18 11:16:12 -0500 (Fri, 18 Jul 2008) | 10 lines Merged revisions 131921 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131921 | kpfleming | 2008-07-18 11:15:41 -0500 (Fri, 18 Jul 2008) | 2 lines remove the dlfcn compatibility stuff, because no platforms that Asterisk currently runs on it use it, and it doesn't build anyway ........ ................ 2008-07-18 15:39 +0000 [r131917] Brett Bryant * /, main/features.c: Merged revisions 131916 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131916 | bbryant | 2008-07-18 10:38:22 -0500 (Fri, 18 Jul 2008) | 12 lines Merged revisions 131915 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131915 | bbryant | 2008-07-18 10:34:42 -0500 (Fri, 18 Jul 2008) | 4 lines Fix a bug in blind transfers where the BLINDTRANSFER variable isn't always set to the other end of the blind transfer. (closes issue #12586) ........ ................ 2008-07-17 22:45 +0000 [r131869] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 131868 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r131868 | jpeeler | 2008-07-17 17:40:00 -0500 (Thu, 17 Jul 2008) | 6 lines Add configuration option to chan_dahdi.conf to allow buffering policy and number of buffers to be configured per channel. Syntax: buffers=, Where the number of buffers is some non-negative integer and the policy is either "full", "half", or "immediate". ........ 2008-07-17 21:27 +0000 [r131830] Mark Michelson * /, apps/app_senddtmf.c: Merged revisions 131824 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r131824 | mmichelson | 2008-07-17 16:26:41 -0500 (Thu, 17 Jul 2008) | 10 lines Document that the duration of dtmf may be passed to the SendDTMF application. Also correct the default pause between digits. (closes issue #13102) Reported by: eliel Patches: app_senddtmf.c.patch uploaded by eliel (license 64) ........ 2008-07-17 20:38 +0000 [r131754-131792] Tilghman Lesher * channels/chan_dahdi.c, /: Merged revisions 131791 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131791 | tilghman | 2008-07-17 15:37:14 -0500 (Thu, 17 Jul 2008) | 15 lines Merged revisions 131790 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131790 | tilghman | 2008-07-17 15:35:44 -0500 (Thu, 17 Jul 2008) | 7 lines Revert part of issue #5620 (revision 6965) as it appears that it was in error. This should fix talk call progress on analog lines. (closes issue #12178) Reported by: michael-fig Patches: 20080717__bug12178.diff.txt uploaded by Corydon76 (license 14) ........ ................ * res/res_config_sqlite.c, /: Merged revisions 131753 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r131753 | tilghman | 2008-07-17 13:36:34 -0500 (Thu, 17 Jul 2008) | 6 lines Fix memory leaks (closes issue #13099) Reported by: gknispel_proformatique Patches: res_config_sqlite_leak_on_error.patch uploaded by gknispel (license 261) ........ 2008-07-17 18:15 +0000 [r131718] Brett Bryant * /, main/features.c: Merged revisions 131717 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r131717 | bbryant | 2008-07-17 13:14:42 -0500 (Thu, 17 Jul 2008) | 8 lines Fix a memory leak in register_group_feature when attempting to register a feature without specifying a group or feature to register. (closes issue #13101) Reported by: eliel Patches: features.c.patch uploaded by eliel (license 64) ........ 2008-07-17 15:46 +0000 [r131682] Tilghman Lesher * res/res_config_sqlite.c, /: Merged revisions 131681 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r131681 | tilghman | 2008-07-17 10:45:25 -0500 (Thu, 17 Jul 2008) | 4 lines Fix memory leak. (Closes issue #13096) Reported by gknispel_proformatique ........ 2008-07-16 23:56 +0000 [r131571] Steve Murphy * /: The commit from 131570 should not be applied to 1.6.0, as it is not as necessary, because log_show_lock in trunk is not available in 1.6.0, and is estimated to be the only function that might care about the lock_addr values. 2008-07-16 22:18 +0000 [r131493] Brett Bryant * /, channels/chan_iax2.c: Merged revisions 131492 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131492 | bbryant | 2008-07-16 17:17:36 -0500 (Wed, 16 Jul 2008) | 14 lines Merged revisions 131491 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131491 | bbryant | 2008-07-16 17:17:07 -0500 (Wed, 16 Jul 2008) | 6 lines Fix a bug in iax2 registration that allowed peers to register with case-insensitive names (user_cmp_cb and peer_cmp_cb are now both case-sensitive). (closes issue #13091) ........ ................ 2008-07-16 21:54 +0000 [r131455-131486] Brett Bryant * /, funcs/func_sysinfo.c: Merged revisions 131484 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r131484 | bbryant | 2008-07-16 16:54:08 -0500 (Wed, 16 Jul 2008) | 4 lines Fixes sysinfo operator issue also fixed elsewhere in r131445. (issue #13057) ........ * main/asterisk.c, /: Merged revisions 131445 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r131445 | bbryant | 2008-07-16 16:24:18 -0500 (Wed, 16 Jul 2008) | 9 lines Fixes an issue with "core show sysinfo" that used the wrong operator to calculate the number of bytes from a sysinfo structure. unsigned long. (closes issue #13057) Reported by: eliel Patches: asterisk.c.patch uploaded by eliel (license 64) ........ 2008-07-16 20:48 +0000 [r131423] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 131422 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131422 | russell | 2008-07-16 15:48:27 -0500 (Wed, 16 Jul 2008) | 15 lines Merged revisions 131421 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131421 | russell | 2008-07-16 15:47:53 -0500 (Wed, 16 Jul 2008) | 7 lines Always ensure that the channel's tech_pvt reference is NULL after calling the destroy callback. (closes issue #13060) Reported by: jpgrayson Patches: chan_iax2_tech_pvt_crash.patch uploaded by jpgrayson (license 492) ........ ................ 2008-07-16 20:24 +0000 [r131301-131378] Mark Michelson * /, apps/app_queue.c: Merged revisions 131375 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131375 | mmichelson | 2008-07-16 15:24:12 -0500 (Wed, 16 Jul 2008) | 22 lines Merged revisions 131369 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131369 | mmichelson | 2008-07-16 15:23:02 -0500 (Wed, 16 Jul 2008) | 14 lines Move the init_queue call back to where it used to be (changed Sept 12 last year). It was moved then to prevent a memory leak. Since then, the same memory leak recurred and was fixed in a better way. Now it has been found that the placement of this init_queue call can cause problems if a realtime queue has values changed to an empty string. The problem is that the default value for that queue parameter would not be set. (closes issue #13084) Reported by: elbriga ........ ................ * res/res_config_sqlite.c, /: Merged revisions 131361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r131361 | mmichelson | 2008-07-16 14:57:02 -0500 (Wed, 16 Jul 2008) | 9 lines Don't try to dereference the dbfile pointer if we know that it's NULL. (closes issue #13092) Reported by: gknispel_proformatique Patches: trunk_sqlite_check_vars_null.patch uploaded by gknispel (license 261) ........ * /, apps/app_queue.c: Merged revisions 131358 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131358 | mmichelson | 2008-07-16 14:37:42 -0500 (Wed, 16 Jul 2008) | 14 lines Merged revisions 131357 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131357 | mmichelson | 2008-07-16 14:37:08 -0500 (Wed, 16 Jul 2008) | 6 lines Apparently, "thread safety" is important, whatever that means. :P (Thanks Russell!) ........ ................ * /, apps/app_queue.c: Merged revisions 131300 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131300 | mmichelson | 2008-07-16 13:59:27 -0500 (Wed, 16 Jul 2008) | 21 lines Merged revisions 131299 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131299 | mmichelson | 2008-07-16 13:57:34 -0500 (Wed, 16 Jul 2008) | 13 lines Make absolutely certain that the transfer datastore is removed from the calling channel once the caller is finished in the queue. This could have weird con- sequences when dialing local queue members when multiple transfers occur on a single call. Also fixed a memory leak that would occur when an attended transfer occurred from a queue member. (closes issue #13047) Reported by: festr ........ ................ 2008-07-16 18:20 +0000 [r131248] Steve Murphy * res/ael/pval.c, /: Merged revisions 131243 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131243 | murf | 2008-07-16 11:59:33 -0600 (Wed, 16 Jul 2008) | 27 lines Merged revisions 131242 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131242 | murf | 2008-07-16 11:53:43 -0600 (Wed, 16 Jul 2008) | 19 lines (closes issue #13090) Reported by: murf The problem was that, esoteric as it is, because the hangerupper context immediately preceded the std-priv-extent macro, that the checking code accidentally would fall from traversing hangerupper into the std-priv-exten macro, where it would hit the hangerupper in the 'includes', and proceed into an infinite recursion. A small fix to traverse into the statements of the context instead of the context solves this issue. I also added some commented out printfs for debug, which were pretty handy in the face of a dorky gdb. This was a problem around since the package was first written; but evidently pretty rare in turning up in the field. ........ ................ 2008-07-16 15:04 +0000 [r131206] Luigi Rizzo * channels/chan_agent.c: add missing terminator argument for ast_event_subscribe(). Without it the function will randomly walk on the stack possibly causing a panic 2008-07-16 00:54 +0000 [r131168] Tilghman Lesher * /, main/logger.c: Merged revisions 131166 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r131166 | tilghman | 2008-07-15 19:52:48 -0500 (Tue, 15 Jul 2008) | 3 lines Fix rotate strategy (Closes issue #13086) ........ 2008-07-15 23:41 +0000 [r131131] Steve Murphy * main/pbx.c, /: Merged revisions 131129 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r131129 | murf | 2008-07-15 17:36:19 -0600 (Tue, 15 Jul 2008) | 21 lines (closes issue #12960) Reported by: mnicholson Spent most of the day on this bug, and the solution was so simple. Just had to find and understand the problem. The problem was, that the routine to copy the existing switches, includes, and ignorepats from the old context to the new one, wasn't getting called when the context is already existent. (In other words, if AEL is adding a new context to the mix, they get copied, but if pbx_config already defined a context, then the copy wasn't happening. This made no sense, so I moved the call to copy the includes & etc, no matter the case. ........ 2008-07-15 18:47 +0000 [r131073] Russell Bryant * /, res/res_agi.c: Merged revisions 131072 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r131072 | russell | 2008-07-15 13:46:40 -0500 (Tue, 15 Jul 2008) | 5 lines Fix a couple of places in res_agi where the agi_commands lock would not be released, causing a deadlock. (Reported by mvanbaak in #asterisk-dev, discovered by bbryant's change to the lock tracking code to yell at you if a thread exits with a lock still held) ........ 2008-07-15 18:29 +0000 [r131060] Tilghman Lesher * main/pbx.c, main/manager.c, /, channels/chan_sip.c: Merged revisions 131044 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131044 | tilghman | 2008-07-15 13:25:34 -0500 (Tue, 15 Jul 2008) | 16 lines Merged revisions 130959 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130959 | tilghman | 2008-07-15 12:19:13 -0500 (Tue, 15 Jul 2008) | 8 lines astman_send_error does not need a newline appended -- the API takes care of that for us. (closes issue #13068) Reported by: gknispel_proformatique Patches: asterisk_1_4_astman_send.patch uploaded by gknispel (license 261) asterisk_trunk_astman_send.patch uploaded by gknispel (license 261) ........ ................ 2008-07-15 18:00 +0000 [r131014] Michiel van Baak * main/cdr.c, /: Merged revisions 131013 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131013 | mvanbaak | 2008-07-15 19:49:48 +0200 (Tue, 15 Jul 2008) | 15 lines Merged revisions 131012 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131012 | mvanbaak | 2008-07-15 19:47:15 +0200 (Tue, 15 Jul 2008) | 7 lines remove 4 lines of redundant code. (closes issue #13080) Reported by: gknispel_proformatique Patches: trunk_ast_cdr_setapp.patch uploaded by gknispel (license 261) ........ ................ 2008-07-15 13:14 +0000 [r130946] Steve Murphy * utils/conf2ael.c, utils/Makefile, res/ael/pval.c, channels/chan_skinny.c, res/ael/ael.tab.c, main/features.c, pbx/pbx_ael.c, res/ael/ael_lex.c, res/ael/ael.tab.h, utils/ael_main.c, include/asterisk/pbx.h, utils/extconf.c, res/ael/ael.flex, pbx/pbx_config.c, apps/app_stack.c, apps/app_dial.c, main/pbx.c, include/asterisk/pval.h, /, channels/chan_sip.c, apps/app_meetme.c, res/ael/ael.y, channels/chan_iax2.c, apps/app_queue.c: Merged revisions 130145 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk Merging this rev from trunk to 1.6.0 was not simple. Why? Because we've enhanced trunk to do a [fast] merge-and-delete operation which also solved problems with contexts having entries from different registrars. Fast as in the amount of time the contexts are locked down. That *is* fast, but traversing the entire dialplan looking for priorities to delete takes more time overall. This particular fix involved pulling in those enhancements from trunk, along with all the various fixes and refinements made along the way. Merging all this from trunk into 1.6 involved: a. mergetrunk6 in the stuff from 130145; b. revert all but the prop changes c. catalog all revisions to pbx.c since 1.6.0 was forked (at rev 105596). d. catalog all revisions to pbx.c in trunk since 1.6.0 was forked, making special note of all revs that were not merged into 1.6.0. e. study each rev in trunk not applied to 1.6.0, and determine if it was involved in the merge_and_delete enhancements in trunk. 25 commits were done in 1.6.0, all but one (106306) was a merge from trunk. Trunk had 22 additional changes, of which 7 were involved in the merge_and_delete enhancements: 106757 108894 109169 116461 123358 130145 130297 f. Go to trunk and collect patches, one by one, of the changes made by each rev across the entire source tree, using svn diff -c > pfile g. Apply each patch in order to 1.6.0, and resolve all failures and compilation problems before proceding to the next patch. h. test the stuff. i. profit! ........ r130145 | murf | 2008-07-11 12:24:31 -0600 (Fri, 11 Jul 2008) | 40 lines (closes issue #13041) Reported by: eliel Tested by: murf (closes issue #12960) Reported by: mnicholson In this 'omnibus' fix, I **think** I solved both the problem in 13041, where unloading pbx_ael.so caused crashes, or incomplete removal of previous registrar'ed entries. And I added code to completely remove all includes, switches, and ignorepats that had a matching registrar entry, which should appease 12960. I also added a lot of seemingly useless brackets around single statement if's, which helped debug so much that I'm leaving them there. I added a routine to check the correlation between the extension tree lists and the hashtab tables. It can be amazingly helpful when you have lots of dialplan stuff, and need to narrow down where a problem is occurring. It's ifdef'd out by default. I cleaned up the code around the new CIDmatch code. It was leaving hanging extens with bad ptrs, getting confused over which objects to remove, etc. I tightened up the code and changed the call to remove_exten in the merge_and_delete code. I added more conditions to check for empty context worthy of deletion. It's not empty if there are any includes, switches, or ignorepats present. If I've missed anything, please re-open this bug, and be prepared to supply example dialplan code. ........ 2008-07-15 00:00 +0000 [r130891] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 130890 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r130890 | tilghman | 2008-07-14 18:59:54 -0500 (Mon, 14 Jul 2008) | 16 lines Merged revisions 130889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130889 | tilghman | 2008-07-14 18:59:13 -0500 (Mon, 14 Jul 2008) | 8 lines Override the callerid in all cases when the callerid is set in the user, not just when a remote callerid is set. Also, if not set in the user, allow the remote CallerID to pass through. (closes issue #12875) Reported by: dimas Patches: 20080714__bug12875.diff.txt uploaded by Corydon76 (license 14) ........ ................ 2008-07-14 22:24 +0000 [r130795-130855] Mark Michelson * main/asterisk.c, /: Merged revisions 130854 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r130854 | mmichelson | 2008-07-14 17:22:57 -0500 (Mon, 14 Jul 2008) | 9 lines Fix a memory leak in the case that /dev/null cannot be opened when running startup commands from cli.conf (closes issue #13066) Reported by: eliel Patches: asterisk.c.patch uploaded by eliel (license 64) ........ * apps/app_dial.c, /: Merged revisions 130794 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r130794 | mmichelson | 2008-07-14 12:54:11 -0500 (Mon, 14 Jul 2008) | 16 lines Merged revisions 130792 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130792 | mmichelson | 2008-07-14 12:50:21 -0500 (Mon, 14 Jul 2008) | 8 lines Add a check to the CAN_EARLY_BRIDGE macro in app_dial to be sure there are no audiohooks present on the channels involved. This fixed a one-way audio situation I had in my test setup. I couldn't find any open issues that suggested one-way audio with regards to mixmonitor (or other audiohook) usage, though. ........ ................ 2008-07-14 17:22 +0000 [r130752] Michiel van Baak * main/dnsmgr.c, /: Merged revisions 130744 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r130744 | mvanbaak | 2008-07-14 19:21:18 +0200 (Mon, 14 Jul 2008) | 18 lines Merged revisions 130735 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130735 | mvanbaak | 2008-07-14 19:10:21 +0200 (Mon, 14 Jul 2008) | 10 lines notify the user that dnsmgr refresh wont work when dnsmgr is not enabled. Previously this command would automagically appear and disappear. This was confusing. (closes issue #12796) Reported by: chappell Patches: dnsmgr_refresh_3.diff uploaded by chappell (license 8) Tested by: russell, chappell, mvanbaak ........ ................ 2008-07-14 10:40 +0000 [r130636-130637] Russell Bryant * /, include/asterisk/astobj.h: Merged revisions 129987 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r129987 | russell | 2008-07-11 09:22:44 -0500 (Fri, 11 Jul 2008) | 10 lines Merged revisions 129970 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129970 | russell | 2008-07-11 09:18:43 -0500 (Fri, 11 Jul 2008) | 2 lines add a simple ASTOBJ_TRYWRLOCK macro ... ........ ................ * /, main/audiohook.c: Merged revisions 130635 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r130635 | russell | 2008-07-14 05:39:23 -0500 (Mon, 14 Jul 2008) | 10 lines Merged revisions 130634 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130634 | russell | 2008-07-14 05:38:14 -0500 (Mon, 14 Jul 2008) | 2 lines Bump up the debug level for a message. ........ ................ 2008-07-13 23:20 +0000 [r130575-130582] Michiel van Baak * /, doc/tex/Makefile, build_tools/prep_tarball, res/Makefile: Merged revisions 130578 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r130578 | mvanbaak | 2008-07-14 01:14:03 +0200 (Mon, 14 Jul 2008) | 15 lines Make all sed calls Posix sed compatible. To make sure nobody commits script-modified files we first make a backup of asterisk.tex, run the script, generate the pdf and / or html, and put the original asterisk.tex back. This will guard us for the stuff that happened before that someone committed a locally modified asterisk.tex, with changes done by this script. (closes issue #13062) Reported by: mvanbaak Patches: sed_without-i-v3.diff uploaded by mvanbaak (license 7) Tested by: mvanbaak Feedback from Corydon. Thanks for taking the time to go through this. ........ * main/manager.c, /: Merged revisions 130574 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r130574 | mvanbaak | 2008-07-14 00:50:31 +0200 (Mon, 14 Jul 2008) | 16 lines Merged revisions 130573 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130573 | mvanbaak | 2008-07-14 00:48:51 +0200 (Mon, 14 Jul 2008) | 8 lines fix memory leak when originate from manager cannot create a thread (closes issue #13069) Reported by: gknispel_proformatique Patches: asterisk_trunk_action_originate.patch uploaded by gknispel (license 261) Tested by: gknispel_proformatique, mvanbaak ........ ................ 2008-07-13 17:59 +0000 [r130516] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 130515 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r130515 | tilghman | 2008-07-13 12:58:47 -0500 (Sun, 13 Jul 2008) | 12 lines Merged revisions 130514 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130514 | tilghman | 2008-07-13 12:56:10 -0500 (Sun, 13 Jul 2008) | 4 lines Reverting 2 changesets, as it breaks incoming IAX2 calls (Related to issue #12963) Reported by: mvanbaak ........ ................ 2008-07-13 15:00 +0000 [r130480] Michiel van Baak * doc/tex/asterisk.tex, /: Merged revisions 130479 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r130479 | mvanbaak | 2008-07-13 16:58:40 +0200 (Sun, 13 Jul 2008) | 3 lines restore ASTERISKVERSION marker to asterisk.tex. This got lost in commit 97634 ........ 2008-07-13 02:35 +0000 [r130445] Tilghman Lesher * /, channels/chan_agent.c: Merged revisions 130444 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r130444 | tilghman | 2008-07-12 21:34:32 -0500 (Sat, 12 Jul 2008) | 2 lines Unlock list before returning ........ 2008-07-11 21:39 +0000 [r130294] Matthew Fredrickson * channels/chan_dahdi.c, /: Merged revisions 130293 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r130293 | mattf | 2008-07-11 16:36:26 -0500 (Fri, 11 Jul 2008) | 1 line Support new TRANSPORT definitions in libss7 ........ 2008-07-11 20:04 +0000 [r130238] Mark Michelson * /, main/audiohook.c: Merged revisions 130237 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r130237 | mmichelson | 2008-07-11 15:03:55 -0500 (Fri, 11 Jul 2008) | 11 lines Merged revisions 130236 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130236 | mmichelson | 2008-07-11 15:03:23 -0500 (Fri, 11 Jul 2008) | 3 lines Remove redundant logic ........ ................ 2008-07-11 19:57 +0000 [r130231-130235] Tilghman Lesher * channels/chan_dahdi.c, /, channels/chan_agent.c, utils/astman.c: Merged revisions 130230 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r130230 | tilghman | 2008-07-11 14:40:55 -0500 (Fri, 11 Jul 2008) | 2 lines Fix trunk breakage ........ 2008-07-11 19:14 +0000 [r130175] Mark Michelson * /, main/audiohook.c: Merged revisions 130174 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r130174 | mmichelson | 2008-07-11 14:14:15 -0500 (Fri, 11 Jul 2008) | 15 lines Merged revisions 130173 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130173 | mmichelson | 2008-07-11 14:13:29 -0500 (Fri, 11 Jul 2008) | 7 lines Fix a typo in audiohook_read_frame_both. While this change has not been proven to fix any specific issue, it is incorrect and could cause unforeseen problems. ........ ................ 2008-07-11 18:53 +0000 [r130171] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 130170 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r130170 | tilghman | 2008-07-11 13:52:42 -0500 (Fri, 11 Jul 2008) | 15 lines Merged revisions 130169 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130169 | tilghman | 2008-07-11 13:51:56 -0500 (Fri, 11 Jul 2008) | 7 lines Ensure that a destination callno of 0 will not match for frames that do not start a dialog (new, lagrq, and ping). (closes issue #12963) Reported by: russellb Patches: chan_iax2_dup_new_fix4.patch uploaded by jpgrayson (license 492) ........ ................ 2008-07-11 18:33 +0000 [r130168] Sean Bright * /, channels/chan_sip.c: Merged revisions 130167 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r130167 | seanbright | 2008-07-11 14:32:26 -0400 (Fri, 11 Jul 2008) | 1 line Missed one. Formatting only. ........ 2008-07-11 18:14 +0000 [r130130] Brett Bryant * main/cli.c, channels/chan_jingle.c, channels/chan_dahdi.c, channels/chan_skinny.c, main/abstract_jb.c, apps/app_minivm.c, codecs/codec_resample.c, codecs/codec_dahdi.c, apps/app_chanspy.c, main/asterisk.c, apps/app_milliwatt.c, main/dsp.c, codecs/codec_g722.c, /, channels/chan_sip.c, main/threadstorage.c, utils/astman.c, main/utils.c, channels/chan_gtalk.c, pbx/dundi-parser.c: Merged revisions 130129 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r130129 | bbryant | 2008-07-11 13:09:35 -0500 (Fri, 11 Jul 2008) | 8 lines Janitor patch to change uses of sizeof to ARRAY_LEN (closes issue #13054) Reported by: pabelanger Patches: ARRAY_LEN.patch2 uploaded by pabelanger (license 224) Tested by: seanbright ........ 2008-07-11 17:30 +0000 [r130127] Tilghman Lesher * /, channels/chan_agent.c: Merged revisions 130126 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r130126 | tilghman | 2008-07-11 12:29:24 -0500 (Fri, 11 Jul 2008) | 17 lines Merged revisions 130102 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130102 | tilghman | 2008-07-11 11:50:42 -0500 (Fri, 11 Jul 2008) | 9 lines Pass the devicestate from an underlying channel up through the Agent channel. This should make the Agent always report the correct device state, even when the underlying channel is used for other purposes. (closes issue #12773) Reported by: davidw Patches: 20080710__bug12773.diff.txt uploaded by Corydon76 (license 14) Tested by: davidw ........ ................ 2008-07-11 16:18 +0000 [r129936-130045] Kevin P. Fleming * doc/ss7.txt, /, contrib/utils/zones2indications.c, CHANGES: Merged revisions 130044 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r130044 | kpfleming | 2008-07-11 11:18:01 -0500 (Fri, 11 Jul 2008) | 2 lines clean up a bunch more Zaptel-related references ........ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 130040 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r130040 | kpfleming | 2008-07-11 10:57:17 -0500 (Fri, 11 Jul 2008) | 12 lines Merged revisions 130039 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130039 | kpfleming | 2008-07-11 10:41:56 -0500 (Fri, 11 Jul 2008) | 4 lines add support for a configuration parameter for 'inband audio during RELEASE', which is currently mandatory in libpri-1.4.4 but will become configurable in libpri-1.4.5 later today (related to issue #13042) ........ ................ * /, main/astmm.c: Merged revisions 129968 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r129968 | kpfleming | 2008-07-11 09:16:15 -0500 (Fri, 11 Jul 2008) | 18 lines Merged revisions 129966 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129966 | kpfleming | 2008-07-11 09:03:52 -0500 (Fri, 11 Jul 2008) | 5 lines fix a flaw found while experimenting with structure alignment and padding; low-fence checking would not work properly on 64-bit platforms, because the compiler was putting 4 bytes of padding between the fence field and the allocation memory block added a very obvious runtime warning if this condition reoccurs, so the developer who broke it can be chastised into fixing it :-) ........ r129967 | kpfleming | 2008-07-11 09:03:52 -0500 (Fri, 11 Jul 2008) | 5 lines simplify calculation ........ ................ * /, sounds/Makefile: Merged revisions 129916 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r129916 | kpfleming | 2008-07-11 07:21:29 -0500 (Fri, 11 Jul 2008) | 10 lines Merged revisions 129907 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129907 | kpfleming | 2008-07-11 07:15:42 -0500 (Fri, 11 Jul 2008) | 2 lines don't attempt to set user/group ownership of extracted sound files (reported on asterisk-users) ........ ................ 2008-07-11 01:01 +0000 [r129865] Sean Bright * res/res_config_pgsql.c, /, res/res_config_ldap.c: Merged revisions 129864 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r129864 | seanbright | 2008-07-10 20:55:06 -0400 (Thu, 10 Jul 2008) | 1 line Fix some usages of snprintf, and clarify a couple variable names. ........ 2008-07-10 22:07 +0000 [r129764-129805] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 129804 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r129804 | tilghman | 2008-07-10 17:06:07 -0500 (Thu, 10 Jul 2008) | 16 lines Merged revisions 129803 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129803 | tilghman | 2008-07-10 16:57:05 -0500 (Thu, 10 Jul 2008) | 8 lines Correctly deal with duplicate NEW frames (due to retransmission). Also, fixup the destination call number matching to be more strict and reliable. (closes issue #12963) Reported by: jpgrayson Patches: chan_iax2_dup_new_fix3.patch uploaded by jpgrayson (license 492) Tested by: jpgrayson, Corydon76 ........ ................ * res/res_config_odbc.c, /: Merged revisions 129758 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r129758 | tilghman | 2008-07-10 16:23:23 -0500 (Thu, 10 Jul 2008) | 10 lines Merged revisions 129741 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129741 | tilghman | 2008-07-10 16:19:48 -0500 (Thu, 10 Jul 2008) | 2 lines Oops ........ ................ 2008-07-10 21:05 +0000 [r129739] Terry Wilson * Makefile, /: Merged revisions 129738 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r129738 | twilson | 2008-07-10 15:56:20 -0500 (Thu, 10 Jul 2008) | 2 lines Move phoneprov config files to be installed with 'make samples' so changes aren't inadvertently lost on a 'make install' ........ 2008-07-10 19:14 +0000 [r129685] Brett Bryant * /, apps/app_queue.c: Merged revisions 129684 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r129684 | bbryant | 2008-07-10 14:13:12 -0500 (Thu, 10 Jul 2008) | 8 lines Fixes a bug where the interface for a queue member gets reloaded as the state_interface, if a state_interface was set, on reload because the state_interface isn't stored in the ast_db. (closes issue #13043) Reported by: jvandal Patches: app_queue.patch uploaded by jvandal (license 413) ........ 2008-07-10 18:20 +0000 [r129640-129647] Sean Bright * /, channels/chan_sip.c: Merged revisions 129642 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r129642 | seanbright | 2008-07-10 14:19:17 -0400 (Thu, 10 Jul 2008) | 1 line A couple more minor text changes ........ * /, channels/chan_sip.c: Merged revisions 129638 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r129638 | seanbright | 2008-07-10 14:16:21 -0400 (Thu, 10 Jul 2008) | 1 line Remove extraneous \n. Pointed out by eliel on #asterisk-dev. ........ 2008-07-10 16:13 +0000 [r129570] Russell Bryant * sample.call, /: Merged revisions 129569 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r129569 | russell | 2008-07-10 11:12:51 -0500 (Thu, 10 Jul 2008) | 11 lines Merged revisions 129567 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129567 | russell | 2008-07-10 11:03:59 -0500 (Thu, 10 Jul 2008) | 3 lines Note that pbx_spool.so is the module used for call files (inspired by a question in #asterisk) ........ ................ 2008-07-10 14:09 +0000 [r129504-129507] Sean Bright * /, main/editline: Merged revisions 129503 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r129503 | seanbright | 2008-07-10 09:54:29 -0400 (Thu, 10 Jul 2008) | 2 lines Update svn:ignore ........ 2008-07-09 19:41 +0000 [r129438] Mark Michelson * main/rtp.c, /: Merged revisions 129437 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r129437 | mmichelson | 2008-07-09 14:40:30 -0500 (Wed, 09 Jul 2008) | 21 lines Merged revisions 129436 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129436 | mmichelson | 2008-07-09 14:32:20 -0500 (Wed, 09 Jul 2008) | 13 lines Fix a problem where inbound rfc2833 audio would be sent to the core instead of being P2P bridged. When the core regenerated the rfc2833 packet for the outbound leg, the SSRC would be different than the RTP audio on the call leg causing DTMF detection issues on the far end. (closes issue #12955) Reported by: tonyredstone Patches: dynamic_rtp.patch uploaded by tsearle (license 373) Tested by: tonyredstone ........ ................ 2008-07-09 16:01 +0000 [r129400] Matthew Fredrickson * main/pbx.c, /: Merged revisions 129399 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r129399 | mattf | 2008-07-09 10:57:06 -0500 (Wed, 09 Jul 2008) | 1 line Add Proceeding() application (#13025) ........ 2008-07-09 13:46 +0000 [r129345] Sean Bright * main/editline/makelist (removed), main/editline/makelist.in (added), /, main/editline/configure, main/editline/Makefile.in, main/editline/configure.in: Merged revisions 129344 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r129344 | seanbright | 2008-07-09 09:44:43 -0400 (Wed, 09 Jul 2008) | 12 lines Merged revisions 129343 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129343 | seanbright | 2008-07-09 09:41:21 -0400 (Wed, 09 Jul 2008) | 4 lines Look for the system installed awk instead of assuming it's at /usr/bin/awk. Pointed out by jmls via #asterisk-dev. ........ ................ 2008-07-08 22:56 +0000 [r129160-129271] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 129270 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r129270 | mmichelson | 2008-07-08 17:56:12 -0500 (Tue, 08 Jul 2008) | 3 lines Fix compilation error when IMAP storage is enabled ........ 2008-07-08 21:04 +0000 [r129157] Brett Bryant * main/dns.c, main/srv.c, /: Merged revisions 129156 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r129156 | bbryant | 2008-07-08 16:00:01 -0500 (Tue, 08 Jul 2008) | 6 lines Fix a bug in SRV lookups where dnsmgr would discard everything but the first SRV result from DNS before processing weights and priorities and dns_parse_answer wouldn't report that there were no records in DNS unless a failure occured. Also fixed a bug where dnsmgr_refresh would report that a entry was being changed when ast_gethostbyname had failed. ........ 2008-07-08 20:31 +0000 [r129049-129153] Tilghman Lesher * apps/app_dial.c, /, channels/chan_sip.c, include/asterisk/causes.h: Merged revisions 129152 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r129152 | tilghman | 2008-07-08 15:30:29 -0500 (Tue, 08 Jul 2008) | 16 lines Merged revisions 129149 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129149 | tilghman | 2008-07-08 15:27:47 -0500 (Tue, 08 Jul 2008) | 8 lines Cause SIP to return a 480 instead of a 404 when a sip peer exists, but is not registered. (closes issue #12885) Reported by: ibc Patches: 20080701__bug12885__2.diff.txt uploaded by Corydon76 (license 14) Tested by: ibc ........ ................ * /, channels/chan_iax2.c: Merged revisions 129048 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r129048 | tilghman | 2008-07-08 11:49:01 -0500 (Tue, 08 Jul 2008) | 15 lines Merged revisions 129047 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129047 | tilghman | 2008-07-08 11:45:23 -0500 (Tue, 08 Jul 2008) | 7 lines Timestamp decoding for video mini-frames is bogus, because the timestamp only includes 15 bits, unlike voice frames, which contain a 16-bit timestamp. (closes issue #13013) Reported by: jpgrayson Patches: chan_iax2_unwrap_ts.patch uploaded by jpgrayson (license 492) ........ ................ 2008-07-08 16:41 +0000 [r129041-129046] Brett Bryant * main/rtp.c, main/channel.c, channels/chan_dahdi.c, main/manager.c, formats/format_pcm.c, main/logger.c, main/callerid.c, apps/app_parkandannounce.c, apps/app_adsiprog.c, main/pbx.c, main/frame.c, /, channels/chan_sip.c, apps/app_meetme.c, channels/h323/ast_h323.cxx, res/res_limit.c, main/acl.c, channels/iax2-provision.c, pbx/dundi-parser.c, channels/chan_iax2.c: Merged revisions 129045 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r129045 | bbryant | 2008-07-08 11:40:28 -0500 (Tue, 08 Jul 2008) | 7 lines Janitor project to convert sizeof to ARRAY_LEN macro. (closes issue #13002) Reported by: caio1982 Patches: janitor_arraylen5.diff uploaded by caio1982 (license 22) ........ * /, channels/chan_sip.c: Merged revisions 127621 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127621 | bbryant | 2008-07-02 17:16:29 -0500 (Wed, 02 Jul 2008) | 1 line Update transport= in sip so that the option is not broken from a recent commit. ........ * /, channels/chan_sip.c: Merged revisions 127434 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127434 | bbryant | 2008-07-02 12:27:36 -0500 (Wed, 02 Jul 2008) | 1 line Fix to sip_parse_host so that it passes the correct information to sip_registry. ........ 2008-07-08 14:18 +0000 [r129007] Russell Bryant * /, apps/app_fax.c: Merged revisions 129006 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r129006 | russell | 2008-07-08 09:17:37 -0500 (Tue, 08 Jul 2008) | 9 lines Update app_fax for better compatibility with spandsp 0.0.5. Add a call to t38_terminal_release, and make sure that the phase E handler gets called with proper status. (closes issue #13020) Reported by: dimas Patches: v1-appfax.patch uploaded by dimas (license 88) ........ 2008-07-08 10:06 +0000 [r128913-128952] Olle Johansson * /, channels/chan_sip.c: Merged revisions 128951 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r128951 | oej | 2008-07-08 12:02:12 +0200 (Tis, 08 Jul 2008) | 19 lines Merged revisions 128950 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r128950 | oej | 2008-07-08 11:52:21 +0200 (Tis, 08 Jul 2008) | 11 lines Don't hangup the call if we can't resolve the Contact if there's a proxy route set for the call. ---- This comment was added a while ago and today it hit me badly. /* OEJ: Possible issue that may need a check: If we have a proxy route between us and the device, should we care about resolving the contact or should we just send it? */ ........ ................ * /, channels/chan_sip.c: Merged revisions 128927 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r128927 | oej | 2008-07-08 11:26:37 +0200 (Tis, 08 Jul 2008) | 15 lines Merged revisions 128912 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r128912 | oej | 2008-07-08 11:06:08 +0200 (Tis, 08 Jul 2008) | 7 lines Fix issues where repeated messages where ignored, but retransmitted reliably instead of unreliably. Reported by: johan Patches: 12746.txt uploaded by oej (license 306) Tested by: johan (issue #12746) ........ ................ 2008-07-08 00:03 +0000 [r128855-128858] Tilghman Lesher * /: Merged revisions 128857 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r128857 | tilghman | 2008-07-07 19:02:11 -0500 (Mon, 07 Jul 2008) | 15 lines Merged revisions 128856 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r128856 | tilghman | 2008-07-07 19:01:30 -0500 (Mon, 07 Jul 2008) | 7 lines Check for non-NULL before stripping characters. (closes issue #12954) Reported by: bfsworks Patches: 20080701__bug12954.diff.txt uploaded by Corydon76 (license 14) Tested by: deti ........ ................ * apps/app_voicemail.c, /: Merged revisions 128830 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r128830 | tilghman | 2008-07-07 18:25:39 -0500 (Mon, 07 Jul 2008) | 10 lines Merged revisions 128812 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r128812 | tilghman | 2008-07-07 18:21:52 -0500 (Mon, 07 Jul 2008) | 2 lines Stop using deprecated method, as requested by Kevin. ........ ................ 2008-07-07 22:44 +0000 [r128797] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 128796 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r128796 | russell | 2008-07-07 17:42:30 -0500 (Mon, 07 Jul 2008) | 16 lines Merged revisions 128795 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r128795 | russell | 2008-07-07 17:41:48 -0500 (Mon, 07 Jul 2008) | 8 lines Fix handling of when a pvt disappears. Properly return the pvt locked and don't hold the pvt lock while destroying the ast_channel. (closes issue #13014) Reported by: jpgrayson Patches: chan_iax2_ast_iax2_new2.patch uploaded by jpgrayson (license 492) ........ ................ 2008-07-07 20:51 +0000 [r128739] Sean Bright * /, channels/chan_iax2.c: Merged revisions 128738 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r128738 | seanbright | 2008-07-07 16:50:29 -0400 (Mon, 07 Jul 2008) | 17 lines Merged revisions 128737 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r128737 | seanbright | 2008-07-07 16:47:56 -0400 (Mon, 07 Jul 2008) | 9 lines Remove spurious trailing whitespace from log messages and fix a spelling error in a log message. (closes issue #13017) Reported by: jpgrayson Patches: chan_iax2_space_after_newline.patch uploaded by jpgrayson (license 492) chan_iax2_spelling.patch uploaded by jpgrayson (license 492) ........ ................ 2008-07-07 20:31 +0000 [r128601-128735] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 128733 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128733 | mmichelson | 2008-07-07 15:30:46 -0500 (Mon, 07 Jul 2008) | 3 lines Crap ........ * apps/app_voicemail.c, /: Merged revisions 128731 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128731 | mmichelson | 2008-07-07 15:28:33 -0500 (Mon, 07 Jul 2008) | 7 lines If imapfolder=foo were set in voicemail.conf, then when calling VoiceMailMain, app_voicemail would attempt to play a file called vm-foo instead of playing vm-INBOX to play the "new" sound file. This commit fixes that issue. This may fix one of the problems reported in issue #12987 ........ * /, channels/chan_iax2.c: Merged revisions 128640 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r128640 | mmichelson | 2008-07-07 12:09:11 -0500 (Mon, 07 Jul 2008) | 18 lines Merged revisions 128639 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r128639 | mmichelson | 2008-07-07 12:02:28 -0500 (Mon, 07 Jul 2008) | 10 lines By using the iaxdynamicthreadcount to identify a thread, it was possible for thread identifiers to be duplicated. By using a globally-unique monotonically- increasing integer, this is now avoided. (closes issue #13009) Reported by: jpgrayson Patches: chan_iax2_dyn_threadnum.patch uploaded by jpgrayson (license 492) ........ ................ * configs/extensions.conf.sample, /, doc/tex/extensions.tex: Merged revisions 128599 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128599 | mmichelson | 2008-07-07 09:35:27 -0500 (Mon, 07 Jul 2008) | 6 lines Update a few instances of "extensions reload" to "dialplan reload" in the documentation. Patch provided by caio1982 (license 22) ........ 2008-07-06 20:22 +0000 [r128288-128543] Olle Johansson * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 128524 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128524 | oej | 2008-07-06 22:11:37 +0200 (Sön, 06 Jul 2008) | 5 lines - Fixing issues with "sip show settings" - Adding IP address for TCP and/or TLS too if auto-domain is enabled and binding to a different IP address - Fixing documentation in sip.conf.sample ........ * /, channels/chan_sip.c: Merged revisions 128491 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128491 | oej | 2008-07-06 21:14:06 +0200 (Sön, 06 Jul 2008) | 3 lines - Remove unused variable "expiry" - Set global_outboundproxy.force at reload. ........ * doc/realtimetext.txt (added), /: The following patch with references to t140red removed, since it only exists in trunk. Merged revisions 128417 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128417 | oej | 2008-07-06 12:13:45 +0200 (Sön, 06 Jul 2008) | 3 lines Adding documentation on the T.140 support in Asterisk. This is a function that we're the reference implementation on now. :-) ........ * /: Merged revisions 128343 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128343 | oej | 2008-07-06 10:10:27 +0200 (Sön, 06 Jul 2008) | 2 lines Removing the CLI dumpdb command (see asterisk-dev discussion and decision) ........ * /, channels/chan_sip.c: Merged revisions 128290 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128290 | oej | 2008-07-05 23:55:57 +0200 (Lör, 05 Jul 2008) | 5 lines Adding doxygen comments to missing parts, moving some #define ...trying to get my head around the thoughts behind the TCP/TLS stuff and figure out what needs to be done to make it useful... ........ * /, channels/chan_sip.c: Merged revisions 128287 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128287 | oej | 2008-07-05 23:37:57 +0200 (Lör, 05 Jul 2008) | 3 lines Adding TCP and TLS to "sip show settings". TLS needs to have one configuration per configured domain at some point. ........ * /: Blocking changes in trunk. 2008-07-05 21:02 +0000 [r128238-128243] Olle Johansson * /: Keep the "sip-user" structure in 1.6.0, while testing new funky stuff in trunk. * /: Blocking the AGi changes from 1.6.0. Let's test them for a while in trunk before a release. * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 128237 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128237 | oej | 2008-07-05 22:39:54 +0200 (Lör, 05 Jul 2008) | 2 lines Make TCP disabled by default (it's considered experimental) ........ * /, configs/sip.conf.sample: Merged revisions 128236 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128236 | oej | 2008-07-05 22:37:53 +0200 (Lör, 05 Jul 2008) | 2 lines Reformatting the config sample ........ 2008-07-05 15:19 +0000 [r128161] Tilghman Lesher * contrib/scripts/asterisk.ldap-schema, contrib/scripts/asterisk.ldif, /: Merged revisions 128160 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128160 | tilghman | 2008-07-05 10:17:51 -0500 (Sat, 05 Jul 2008) | 7 lines LDAP schema updates (closes issue #12860) Reported by: flyn Patches: asterisk.ldif uploaded by suretec (license 70) asterisk.schema uploaded by suretec (license 70) ........ 2008-07-05 03:40 +0000 [r128124-128127] Matthew Fredrickson * channels/chan_dahdi.c, /: Merged revisions 128125 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128125 | mattf | 2008-07-04 22:39:07 -0500 (Fri, 04 Jul 2008) | 1 line It would help if we actually parsed the ss7_explicitacm option in the config file... ........ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged revisions 128122 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128122 | mattf | 2008-07-04 22:26:42 -0500 (Fri, 04 Jul 2008) | 1 line Add option to wait to be able to explicitly send ACM via the Proceeding() application in the dialplan. Also minor documentation update explaining how to setup multiple signalling links within a linkset ........ 2008-07-04 16:12 +0000 [r128028-128031] Tilghman Lesher * main/pbx.c, /, include/asterisk/pbx.h, pbx/pbx_config.c: Merged revisions 128027 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r128027 | tilghman | 2008-07-04 11:06:34 -0500 (Fri, 04 Jul 2008) | 16 lines Merged revisions 127973 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r127973 | tilghman | 2008-07-03 22:30:30 -0500 (Thu, 03 Jul 2008) | 8 lines Fix the 'dialplan remove extension' logic, so that it a) works with cidmatch, and b) completes contexts correctly when the extension is ambiguous. (closes issue #12980) Reported by: licedey Patches: 20080703__bug12980.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ ................ 2008-07-03 22:23 +0000 [r127905] Kevin P. Fleming * Makefile, /, apps/Makefile, main/editline/np/vis.c: Merged revisions 127903 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r127903 | kpfleming | 2008-07-03 17:23:04 -0500 (Thu, 03 Jul 2008) | 20 lines Merged revisions 127892,127895 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r127892 | kpfleming | 2008-07-03 17:18:38 -0500 (Thu, 03 Jul 2008) | 6 lines a couple of small Solaris-related fixes (closes issue #11885) Reported by: snuffy, asgaroth ........ r127895 | kpfleming | 2008-07-03 17:20:16 -0500 (Thu, 03 Jul 2008) | 3 lines remove this, it has been moved to the main Makefile ........ ................ 2008-07-03 19:12 +0000 [r127830] Steve Murphy * main/cdr.c, main/channel.c, channels/chan_dahdi.c, main/pbx.c, /, channels/chan_sip.c, main/features.c, include/asterisk/cdr.h: Merged revisions 127793 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r127793 | murf | 2008-07-03 11:16:44 -0600 (Thu, 03 Jul 2008) | 38 lines Merged revisions 127663 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r127663 | murf | 2008-07-02 18:16:25 -0600 (Wed, 02 Jul 2008) | 30 lines The CDRfix4/5/6 omnibus cdr fixes. (closes issue #10927) Reported by: murf Tested by: murf, deeperror (closes issue #12907) Reported by: falves11 Tested by: murf, falves11 (closes issue #11849) Reported by: greyvoip As to 11849, I think these changes fix the core problems brought up in that bug, but perhaps not the more global problems created by the limitations of CDR's themselves not being oriented around transfers. Reopen if necc, but bug reports are not the best medium for enhancement discussions. We need to start a second-generation CDR standardization effort to cover transfers. (closes issue #11093) Reported by: rossbeer Tested by: greyvoip, murf ........ ................ 2008-07-03 16:50 +0000 [r127790-127792] Olle Johansson * /, channels/chan_sip.c: Merged revisions 127791 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127791 | oej | 2008-07-03 18:48:23 +0200 (Tor, 03 Jul 2008) | 5 lines Make sure we stop session timers as soon as we start hanging up an active call. May fix issue 12919. ........ * /, channels/chan_sip.c: Merged revisions 127779 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127779 | oej | 2008-07-03 18:25:59 +0200 (Tor, 03 Jul 2008) | 4 lines Revert some logic for session timers. We do send in-dialog requests that should not have session-timer require headers, like MESSAGE and REFER. So in the future, only add them on requests and responses that are related to INVITEs and re-INVITEs. ........ 2008-07-03 16:24 +0000 [r127778] Kevin P. Fleming * /, configure, include/asterisk/autoconfig.h.in, configure.ac, acinclude.m4: Merged revisions 127767 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127767 | kpfleming | 2008-07-03 11:22:02 -0500 (Thu, 03 Jul 2008) | 2 lines some minor fixes found while working on issue #12911 (and block the rev from 1.4 since the equivalent is already here) ........ 2008-07-02 21:10 +0000 [r127567] Mark Michelson * /, doc/janitor-projects.txt: Merged revisions 127566 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127566 | mmichelson | 2008-07-02 16:09:18 -0500 (Wed, 02 Jul 2008) | 4 lines Add a janitor project to use ARRAY_LEN instead of in-line sizeof() and division. ........ 2008-07-02 20:49 +0000 [r127559-127563] Mark Michelson * /, channels/chan_agent.c: Merged revisions 127562 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r127562 | mmichelson | 2008-07-02 15:49:08 -0500 (Wed, 02 Jul 2008) | 11 lines Merged revisions 127560 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r127560 | mmichelson | 2008-07-02 15:47:38 -0500 (Wed, 02 Jul 2008) | 3 lines Fix thread-safety of some of the pbx_builtin_getvar_helper calls ........ ................ 2008-07-02 19:48 +0000 [r127467-127503] Tilghman Lesher * /, main/acl.c: Merged revisions 127466 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127466 | tilghman | 2008-07-02 13:31:11 -0500 (Wed, 02 Jul 2008) | 6 lines Solaris fix (closes issue #12949) Reported by: snuffy Patches: bug_12949.diff uploaded by snuffy (license 35) ........ 2008-07-02 14:30 +0000 [r127396-127399] Sean Bright * cdr/cdr_tds.c, /: Merged revisions 127398 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127398 | seanbright | 2008-07-02 10:30:09 -0400 (Wed, 02 Jul 2008) | 1 line Fix a bug I noticed while doing the previous merge ........ * cdr/cdr_tds.c, /, doc/tex/freetds.tex, configure, include/asterisk/autoconfig.h.in, configure.ac, UPGRADE.txt: Merged revisions 126226,126513 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r126226 | seanbright | 2008-06-28 17:28:16 -0400 (Sat, 28 Jun 2008) | 8 lines Merge in changes from my cdr-tds-conversion branch. This changes the internal implementation from using the volatile libtds, to using the db-lib front end. The unintended side effect of this is that we support (at least) versions 0.62 through 0.82 of the FreeTDS distribution without any #ifdef ugliness. (closes issue #12844) Reported by: jcollie ........ r126513 | seanbright | 2008-06-30 07:57:42 -0400 (Mon, 30 Jun 2008) | 4 lines Cast a few more strings to char *, so that we can compile cleanly against FreeTDS 0.60. Update the docs to reflect that we can now compile and run against all modern releases of FreeTDS (0.60 through 0.82) ........ * /: Unblock some revisions so I can merge the cdr_tds changes from trunk 2008-07-02 12:09 +0000 [r127364] Russell Bryant * doc/CODING-GUIDELINES, /: Merged revisions 127363 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127363 | russell | 2008-07-02 07:08:33 -0500 (Wed, 02 Jul 2008) | 13 lines Add a locking section to the coding guidelines document. This section covers some locking fundamentals, as well as some information on locking as it is used in Asterisk. It describes some of the ways that are used and could be used to achieve deadlock avoidance. It also demonstrates the unfortunate conclusion that with the use of recursive locks, none of the constructs in use today are failsafe from deadlocks. Finally, it makes some recommendations for new code being written. As proper locking strategies is a complex subject, this section still has room for expansion and improvement. This is a result of collaboration between Luigi Rizzo and myself on the asterisk-dev mailing list. ........ 2008-07-02 02:49 +0000 [r127298] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 127297 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127297 | tilghman | 2008-07-01 21:48:43 -0500 (Tue, 01 Jul 2008) | 12 lines Change the global timer B to be dependent on the value of the T1 timer, as recommended in RFC 3261, instead of being hardcoded to 32 seconds. This is important for LANs, as it allows autocongestion to occur much more quickly, if desired by the local PBX administrator. It also corrects a bug: if the T1 timer was increased beyond 500ms, then timer B would have been set at a much lower value than recommended. (closes issue #12544) Reported by: kactus Patches: 20080616__bug12544.diff.txt uploaded by Corydon76 (license 14) Tested by: kactus ........ 2008-07-01 23:39 +0000 [r127246] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 127245 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r127245 | mmichelson | 2008-07-01 18:38:12 -0500 (Tue, 01 Jul 2008) | 13 lines Merged revisions 127244 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r127244 | mmichelson | 2008-07-01 18:36:40 -0500 (Tue, 01 Jul 2008) | 5 lines Add error message to failed open(2) calls inside the copy() function of app_voicemail. This idea came as part of my work in helping to resolve issue #12764. ........ ................ 2008-07-01 21:19 +0000 [r127163] Brett Bryant * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 127154 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127154 | bbryant | 2008-07-01 16:03:52 -0500 (Tue, 01 Jul 2008) | 2 lines Add a configuration option so the global outboundproxy can use tcptls without it being defined by each sip user. ........ 2008-07-01 21:16 +0000 [r127156-127158] Mark Michelson * main/channel.c, /: Merged revisions 127157 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127157 | mmichelson | 2008-07-01 16:16:00 -0500 (Tue, 01 Jul 2008) | 8 lines Place the delay in __ast_answer prior to the channel-specific answer callback. This change differs from commit 127113 in that now the channel is not set to AST_STATE_UP until after the answer callback. (closes issue #12924) Reported by: snyfer ........ * main/channel.c, /: Merging Revision 127113 from trunk 2008-07-01 20:52 +0000 [r127153] Jason Parker * Makefile, /: Merged revisions 127152 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127152 | qwell | 2008-07-01 15:51:43 -0500 (Tue, 01 Jul 2008) | 7 lines Fix a typo that caused this asterisk.conf to not get correctly generated. (closes issue #12966) Reported by: ibc Patches: 12966.patch uploaded by bkruse (license 132) ........ 2008-07-01 20:29 +0000 [r127085-127149] Tilghman Lesher * build_tools/cflags.xml, /, channels/chan_iax2.c: Merged revisions 127143 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r127143 | tilghman | 2008-07-01 15:28:54 -0500 (Tue, 01 Jul 2008) | 10 lines Merged revisions 127133 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r127133 | tilghman | 2008-07-01 15:25:37 -0500 (Tue, 01 Jul 2008) | 2 lines Disable the old, slow search for matching callno in chan_iax2 (but allow it to be reenabled for debugging) ........ ................ * /, channels/chan_iax2.c: Merged revisions 127074 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r127074 | tilghman | 2008-07-01 14:20:25 -0500 (Tue, 01 Jul 2008) | 16 lines Merged revisions 127068 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r127068 | tilghman | 2008-07-01 13:52:53 -0500 (Tue, 01 Jul 2008) | 8 lines Change around how we schedule pings and lagrqs, and fix a reason why the jobs were not getting properly cancelled. (closes issue #12903) Reported by: stevedavies Patches: 20080620__bug12903__2.diff.txt uploaded by Corydon76 (license 14) Tested by: stevedavies ........ ................ 2008-07-01 16:53 +0000 [r127001] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 127000 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r127000 | tilghman | 2008-07-01 11:52:29 -0500 (Tue, 01 Jul 2008) | 10 lines Merged revisions 126999 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r126999 | tilghman | 2008-07-01 11:50:46 -0500 (Tue, 01 Jul 2008) | 2 lines Suppress annoying warning by finding the remaining cases where the callno is not in the hash. ........ ................ 2008-07-01 15:05 +0000 [r126756-126904] Olle Johansson * /, channels/chan_sip.c: Merged revisions 126903 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r126903 | oej | 2008-07-01 17:03:59 +0200 (Tis, 01 Jul 2008) | 15 lines Merged revisions 126902 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r126902 | oej | 2008-07-01 16:59:31 +0200 (Tis, 01 Jul 2008) | 7 lines Use domain part of SIP uri in register= configuration as fromdomain. Reported by: one47 Patches: sip-reg-fromdom2.dpatch uploaded by one47 (license 23) (closes issue #12474) ........ ................ * /, channels/chan_sip.c: Merged revisions 126900 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r126900 | oej | 2008-07-01 16:32:15 +0200 (Tis, 01 Jul 2008) | 16 lines Merged revisions 126899 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r126899 | oej | 2008-07-01 16:27:33 +0200 (Tis, 01 Jul 2008) | 8 lines Handle escaped URI's in call pickups. Patch by oej and IgorG. Reported by: IgorG Patches: bug12299-11062-v2.patch uploaded by IgorG (license 20) Tested by: IgorG, oej (closes issue #12299) ........ ................ * /, configs/sip.conf.sample: Merged revisions 126845 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r126845 | oej | 2008-07-01 14:54:57 +0200 (Tis, 01 Jul 2008) | 14 lines Merged revisions 126844 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r126844 | oej | 2008-07-01 14:53:01 +0200 (Tis, 01 Jul 2008) | 5 lines Clear up documentation on "domain=" setting in sip.conf Reported by: davidw (closes issue #12413) ........ ................ * /, channels/chan_sip.c: Merged revisions 126790 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r126790 | oej | 2008-07-01 13:58:17 +0200 (Tis, 01 Jul 2008) | 14 lines Merged revisions 126789 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r126789 | oej | 2008-07-01 13:51:38 +0200 (Tis, 01 Jul 2008) | 6 lines Report 200 OK to all in-dialog OPTIONs requests (to confirm that the dialog exist). Don't bother checking the request URI. (closes issue #11264) Reported by: ibc ........ ................ * /, channels/chan_sip.c: Merged revisions 126755 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r126755 | oej | 2008-07-01 11:51:22 +0200 (Tis, 01 Jul 2008) | 15 lines Merged revisions 126735 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r126735 | oej | 2008-07-01 09:49:15 +0200 (Tis, 01 Jul 2008) | 7 lines Fix bad XML for hold notification. Reported by: gowen72 Patches: hold.patch uploaded by gowen72 (license 432) (closes issue #12942) ........ ................ 2008-06-30 22:34 +0000 [r126676] Jeff Peeler * configs/zapata.conf.sample (removed), configs/chan_dahdi.conf.sample (added), /: Merged revisions 126675 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r126675 | jpeeler | 2008-06-30 17:34:08 -0500 (Mon, 30 Jun 2008) | 1 line rename zapata.conf.sample to chan_dahdi.conf.sample ........ 2008-06-30 20:32 +0000 [r126638] Matthew Fredrickson * channels/chan_dahdi.c, /: Merged revisions 126637 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r126637 | mattf | 2008-06-30 15:25:46 -0500 (Mon, 30 Jun 2008) | 1 line Add support to see MTP2 down events when the link layer drops in SS7 ........ 2008-06-30 16:09 +0000 [r126575] Russell Bryant * /, include/asterisk/lock.h: Merged revisions 126574 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r126574 | russell | 2008-06-30 11:07:25 -0500 (Mon, 30 Jun 2008) | 18 lines Merged revisions 126573 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r126573 | russell | 2008-06-30 11:05:08 -0500 (Mon, 30 Jun 2008) | 10 lines Fix a typo in the non-DEBUG_THREADS version of the recently added DEADLOCK_AVOIDANCE() macro. This caused the lock to not actually be released, and as a result, not avoid deadlocks at all. This resolves the issues reported in the last while about Asterisk locking up all over the place (and most commonly, in chan_iax2). (closes issue #12927) (closes issue #12940) (closes issue #12925) (potentially closes others ...) ........ ................ 2008-06-30 13:07 +0000 [r126518] Olle Johansson * /, channels/chan_sip.c: Merged revisions 126517 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r126517 | oej | 2008-06-30 15:03:53 +0200 (MÃ¥n, 30 Jun 2008) | 20 lines The following patch with some changes for trunk... Merged revisions 126516 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r126516 | oej | 2008-06-30 14:50:55 +0200 (MÃ¥n, 30 Jun 2008) | 10 lines Send all responses to an INVITE reliably, so that we retransmit if we don't get an ACK and also fail if we don't get the very same precious ACK. Based on patch by tsearle, with my own additions. (closes issue #12951) Reported by: tsearle Patches: busy_retransmit.patch uploaded by tsearle (license 373) ........ ................ 2008-06-29 17:02 +0000 [r126362-126364] Kevin P. Fleming * apps/app_zapbarge.c (removed): finish converting this module * pbx/pbx_gtkconsole.c, /, configure, configure.ac, pbx/pbx_lua.c, pbx/Makefile: Merged revisions 126356 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r126356 | kpfleming | 2008-06-29 09:19:29 -0700 (Sun, 29 Jun 2008) | 9 lines various minor fixes created while i worked on getting *every* Asterisk module to build on laptop in dev mode: remove weird pre-setting of LUA paths; they are not necessary; also use the proper path for including the files in pbx_lua.c make the compiler shut up about some ignored function results in pbx_gtkconsole; this module is badly coded anyway ........ * apps/app_dahdibarge.c (added): don't know how this got missed in the DAHDI conversion of this branch 2008-06-29 13:20 +0000 [r126227-126322] Sean Bright * /, cdr/cdr_pgsql.c: Merged revisions 126274 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r126274 | seanbright | 2008-06-29 08:06:46 -0400 (Sun, 29 Jun 2008) | 6 lines Quote column names when inserting CDRs into postgres to avoid conflicts with reserved words. (closes issue #12947) Reported by: panolex ........ 2008-06-28 15:58 +0000 [r126155-126188] Kevin P. Fleming * Makefile, /: update this branch to use the trunk goodness version of menuselect 2008-06-27 22:43 +0000 [r126058-126112] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 126057 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r126057 | tilghman | 2008-06-27 17:10:34 -0500 (Fri, 27 Jun 2008) | 12 lines Merged revisions 126056 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r126056 | tilghman | 2008-06-27 17:01:09 -0500 (Fri, 27 Jun 2008) | 4 lines When we get a 408 Timeout, don't stop trying to re-register. (closes issue #12863) Reported by: ricvil ........ ................ 2008-06-27 21:00 +0000 [r126023] Mark Michelson * apps/app_queue.c: Port revisions 124661 and 123650 from trunk to 1.6.0 Thanks to Atis Lezdins for pointing this out on the asterisk-dev mailing list 2008-06-27 19:20 +0000 [r125994] Russell Bryant * /, doc/siptls.txt: Merged revisions 125988 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125988 | russell | 2008-06-27 14:19:08 -0500 (Fri, 27 Jun 2008) | 3 lines Fix a typo. Someone on IRC copied this literally and then wondered why it wasn't working. :) ........ 2008-06-27 19:06 +0000 [r125981-125985] Matthew Fredrickson * channels/chan_dahdi.c, /: Merged revisions 125984 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125984 | mattf | 2008-06-27 14:05:40 -0500 (Fri, 27 Jun 2008) | 1 line Revert this part of the fix. We'll fix it in libss7 ........ * channels/chan_dahdi.c, /: Merged revisions 125982 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125982 | mattf | 2008-06-27 14:00:44 -0500 (Fri, 27 Jun 2008) | 1 line Obviously somebody didn't compile with libss7 support when doing the DAHDI conversion. ........ * channels/chan_dahdi.c, /: Merged revisions 125980 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125980 | mattf | 2008-06-27 13:32:17 -0500 (Fri, 27 Jun 2008) | 1 line Add support for new commands to block/unblock all CICs on a linkset ........ 2008-06-27 17:36 +0000 [r125948] Brett Bryant * /, channels/chan_sip.c: Merged revisions 125947 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125947 | bbryant | 2008-06-27 12:35:41 -0500 (Fri, 27 Jun 2008) | 1 line Small error in the function that converts peer transports to a string. ........ 2008-06-27 16:29 +0000 [r125892] Brett Bryant * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 125891 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125891 | bbryant | 2008-06-27 11:28:06 -0500 (Fri, 27 Jun 2008) | 6 lines Change the way that the transport option works for sip users. transport will now take multiple arguments, the first one listed will be the one used for new dialogs, and the rest listed will be acceptable ways for that peer to contact us. This fixes a minor bug where, because SIP TCP/UDP run on the same port, could cause a TCP peer to be saved in the ast_db. There will also be warnings when a transport is changed for an unexpected reason. (issue #12799) ........ 2008-06-27 16:19 +0000 [r125859-125863] Mark Michelson * /, apps/app_queue.c: Merged revisions 125855 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125855 | mmichelson | 2008-06-27 11:16:13 -0500 (Fri, 27 Jun 2008) | 5 lines Ensure the thread-safety of the monexec variable in app_queue. Thanks to Russell for pointing out the problem ........ 2008-06-27 16:01 +0000 [r125854] Tilghman Lesher * channels/chan_local.c, /: Merged revisions 125853 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125853 | tilghman | 2008-06-27 11:00:05 -0500 (Fri, 27 Jun 2008) | 3 lines Revert half of the fix, as this part may have been unnecessary (related to issue #12914) Requested here: http://lists.digium.com/pipermail/asterisk-dev/2008-June/033658.html ........ 2008-06-27 14:57 +0000 [r125800-125852] Mark Michelson * main/asterisk.c, main/channel.c, channels/chan_iax2.c: Make sure to only include dahdi/user.h if we have installed DAHDI. * channels/chan_iax2.c: I accidentally committed a change to chan_iax2.c in addition to a change to app_queue.c. Reverting the change to chan_iax2.c, even though it may turn out that this change is necessary. * utils/Makefile, /: Merged revisions 125799 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125799 | mmichelson | 2008-06-27 09:14:09 -0500 (Fri, 27 Jun 2008) | 3 lines Remove an unneeded target from the Makefile ........ 2008-06-27 14:09 +0000 [r125742-125797] Tilghman Lesher * /, main/utils.c, include/asterisk/lock.h: Merged revisions 125794 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r125794 | tilghman | 2008-06-27 08:54:13 -0500 (Fri, 27 Jun 2008) | 10 lines Merged revisions 125793 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125793 | tilghman | 2008-06-27 08:45:03 -0500 (Fri, 27 Jun 2008) | 2 lines In this debugging function, copy to a buffer instead of using potentially unsafe pointers. ........ ................ * channels/chan_local.c, /: Merged revisions 125741 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r125741 | tilghman | 2008-06-27 07:28:38 -0500 (Fri, 27 Jun 2008) | 15 lines Merged revisions 125740 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125740 | tilghman | 2008-06-27 07:19:39 -0500 (Fri, 27 Jun 2008) | 7 lines Add proper deadlock avoidance. (closes issue #12914) Reported by: ozan Patches: 20080625__bug12914.diff.txt uploaded by Corydon76 (license 14) Tested by: ozan ........ ................ 2008-06-27 07:41 +0000 [r125704] Philippe Sultan * /, include/asterisk/jabber.h, res/res_jabber.c: Merged revisions 125703 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125703 | phsultan | 2008-06-27 09:28:17 +0200 (Fri, 27 Jun 2008) | 1 line Fix a compile time error that occurs if OpenSSL is not installed. Reported by Noel Morais on the users mailing list ........ 2008-06-27 01:09 +0000 [r125648-125684] Mark Michelson * apps/app_queue.c, channels/chan_iax2.c: SENTINEL is not defined in 1.6.0 * /, apps/app_queue.c: Merged revisions 125666 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125666 | mmichelson | 2008-06-26 19:22:03 -0500 (Thu, 26 Jun 2008) | 3 lines Make this compile with dev-mode on ........ * /, apps/app_queue.c: Merged revisions 125649 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125649 | mmichelson | 2008-06-26 19:15:54 -0500 (Thu, 26 Jun 2008) | 15 lines The monitor-join option for queues was deprecated in favor of using MixMonitor to mix audio. However, it was pointed out to me that because of this, the command set for the MONITOR_EXEC variable is ignored as well. This means that people can't do their own custom mixing commands at the end of recordings in order to make, for instance, stereo recordings of calls. With this patch, app_queue will set the "joinfiles" variable for the channel's monitor if MONITOR_EXEC is not zero-length. This means that for normal audio mixing, MixMonitor is still the preferred choice, but we allow custom mixing to be done with the two Monitor streams if desired. (closes issue #12923) Reported by: snyfer ........ 2008-06-26 23:06 +0000 [r125592] Mark Michelson * /, apps/app_queue.c: Merged revisions 125591 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125591 | mmichelson | 2008-06-26 18:06:18 -0500 (Thu, 26 Jun 2008) | 3 lines Fix a really stupid mistake ........ 2008-06-26 23:05 +0000 [r125590] Jason Parker * /, main/utils.c: Merged revisions 125589 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r125589 | qwell | 2008-06-26 18:04:18 -0500 (Thu, 26 Jun 2008) | 9 lines Merged revisions 125587 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125587 | qwell | 2008-06-26 18:03:15 -0500 (Thu, 26 Jun 2008) | 1 line Make sure to unlock the lock_info lock (huh?). Possible deadlock? ........ ................ 2008-06-26 23:04 +0000 [r125588] Mark Michelson * /, apps/app_queue.c: Merged revisions 125586 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r125586 | mmichelson | 2008-06-26 18:01:02 -0500 (Thu, 26 Jun 2008) | 19 lines Merged revisions 125585 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125585 | mmichelson | 2008-06-26 17:52:39 -0500 (Thu, 26 Jun 2008) | 11 lines Add the interface of a queue member to the output of the "queue show" command so that it can easily be associated with a queue member's name. This helps so that the appropriate queue member can be removed or paused since the interface is required, not the member's name. (closes issue #12783) Reported by: davevg Patches: app_queue.diff uploaded by davevg (license 209) with small mod from me ........ ................ 2008-06-26 22:50 +0000 [r125584] Tilghman Lesher * /, contrib/scripts/astcli: Merged revisions 125583 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125583 | tilghman | 2008-06-26 17:49:16 -0500 (Thu, 26 Jun 2008) | 2 lines Don't hang if the command is blank ........ 2008-06-26 22:06 +0000 [r125478-125532] Mark Michelson * /, apps/app_queue.c: Merged revisions 125477 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r125477 | mmichelson | 2008-06-26 15:57:41 -0500 (Thu, 26 Jun 2008) | 19 lines Merged revisions 125476 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125476 | mmichelson | 2008-06-26 15:56:01 -0500 (Thu, 26 Jun 2008) | 11 lines Prior to this patch, the "queue show" command used cached information for realtime queues instead of giving up-to-date info. Now realtime is queried for the latest and greatest in queue info. (closes issue #12858) Reported by: bcnit Patches: queue_show.patch uploaded by putnopvut (license 60) ........ ................ 2008-06-26 17:07 +0000 [r125388] Olle Johansson * /, channels/chan_sip.c: Merged revisions 125385 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r125385 | oej | 2008-06-26 18:54:22 +0200 (Tor, 26 Jun 2008) | 12 lines Merged revisions 125384 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125384 | oej | 2008-06-26 18:32:08 +0200 (Tor, 26 Jun 2008) | 3 lines Add support for peer realm based auth (a few missing lines, the rest is well documented but never worked) ........ ................ 2008-06-26 15:52 +0000 [r125280-125334] Kevin P. Fleming * channels/chan_dahdi.c, /: Merged revisions 125333 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r125333 | kpfleming | 2008-06-26 10:50:07 -0500 (Thu, 26 Jun 2008) | 13 lines Merged revisions 125327 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125327 | kpfleming | 2008-06-26 10:30:33 -0500 (Thu, 26 Jun 2008) | 5 lines ensure that (whenever possible) if we generate a log message because an ioctl() call to DAHDI/Zaptel failed, that we include the reason it failed by including the stringified error number (issue AST-80) ........ ................ * /, res/res_musiconhold.c: Merged revisions 125279 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125279 | kpfleming | 2008-06-26 07:09:24 -0500 (Thu, 26 Jun 2008) | 2 lines fix compile failure found by buildbot (go, buildbot!) ........ 2008-06-26 11:08 +0000 [r125192-125278] Tilghman Lesher * main/rtp.c, /: Merged revisions 125277 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r125277 | tilghman | 2008-06-26 06:02:11 -0500 (Thu, 26 Jun 2008) | 15 lines Merged revisions 125276 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125276 | tilghman | 2008-06-26 06:01:21 -0500 (Thu, 26 Jun 2008) | 7 lines Check for rtcp structure before trying to delete schedule. (closes issue #12872) Reported by: destiny6628 Patches: 20080621__bug12872.diff.txt uploaded by Corydon76 (license 14) Tested by: destiny6628 ........ ................ * configs/agents.conf.sample, /: Merged revisions 125223 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r125223 | tilghman | 2008-06-25 20:25:16 -0500 (Wed, 25 Jun 2008) | 12 lines Merged revisions 125218 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125218 | tilghman | 2008-06-25 20:24:26 -0500 (Wed, 25 Jun 2008) | 4 lines Document ackcall=always. (closes issue #12852) Reported by: davidw ........ ................ * configs/http.conf.sample, /: Merged revisions 125191 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125191 | tilghman | 2008-06-25 20:11:43 -0500 (Wed, 25 Jun 2008) | 6 lines Update sample configuration to match what are now the defaults for the prefix. (closes issue #12838, related to issue #12198) Reported by: pabelanger Patches: http.conf.diff2 uploaded by pabelanger (license 224) ........ 2008-06-25 23:20 +0000 [r125146] Kevin P. Fleming * main/channel.c, channels/chan_dahdi.c, apps/app_flash.c, configure, codecs/codec_dahdi.c, apps/app_rpt.c, main/asterisk.c, /, apps/app_meetme.c, main/Makefile, apps/app_dahdiscan.c, apps/app_dahdiras.c, configure.ac, include/asterisk/dahdi.h (removed), res/res_musiconhold.c, channels/chan_iax2.c: Merged revisions 125138 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r125138 | kpfleming | 2008-06-25 18:05:28 -0500 (Wed, 25 Jun 2008) | 18 lines Merged revisions 125132 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125132 | kpfleming | 2008-06-25 17:21:30 -0500 (Wed, 25 Jun 2008) | 10 lines allow tonezone to live in a different place than DAHDI/Zaptel, since dahdi-tools and dahdi-linux are now separate packages and can be installed in different places don't include tonezone.h in dahdi_compat.h, because only a couple of modules need it get app_rpt building again after the DAHDI changes (closes issue #12911) Reported by: tzafrir ........ ................ 2008-06-25 01:13 +0000 [r124964-124967] Tilghman Lesher * channels/chan_dahdi.c, /, include/asterisk/lock.h: Merged revisions 124966 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r124966 | tilghman | 2008-06-24 20:08:37 -0500 (Tue, 24 Jun 2008) | 15 lines Merged revisions 124965 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r124965 | tilghman | 2008-06-24 19:46:24 -0500 (Tue, 24 Jun 2008) | 7 lines Pvt deadlock causes some channels to get stuck in Reserved status. (closes issue #12621) Reported by: fabianoheringer Patches: 20080612__bug12621.diff.txt uploaded by Corydon76 (license 14) Tested by: fabianoheringer ........ ................ * apps/app_voicemail.c, /: Merged revisions 124912 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r124912 | tilghman | 2008-06-24 16:18:52 -0500 (Tue, 24 Jun 2008) | 16 lines Merged revisions 124910 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r124910 | tilghman | 2008-06-24 16:08:52 -0500 (Tue, 24 Jun 2008) | 8 lines Occasionally control characters find their way into CallerID. These need to be stripped prior to placing CallerID in the headers of an email. (closes issue #12759) Reported by: RobH Patches: 20080602__bug12759__2.diff.txt uploaded by Corydon76 (license 14) Tested by: RobH ........ ................ 2008-06-24 17:52 +0000 [r124871-124873] Philippe Sultan * /, res/res_jabber.c: Merged revisions 124872 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r124872 | phsultan | 2008-06-24 19:50:22 +0200 (Tue, 24 Jun 2008) | 6 lines Subscribe to buddy's presence only if we really need to. That is, if the corresponding roster item has a subscription value set to "none" or "from". Make the code more readable. ........ * /, res/res_jabber.c: Merged revisions 124870 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r124870 | phsultan | 2008-06-24 19:28:39 +0200 (Tue, 24 Jun 2008) | 1 line Code simplification ........ 2008-06-23 15:44 +0000 [r124708] Dwayne M. Hubbard * /: blocked revision 124707, taskprocessors are not in 1.6.0 2008-06-22 03:18 +0000 [r124542] Steve Murphy * apps/app_forkcdr.c, /: Merged revisions 124541 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r124541 | murf | 2008-06-21 20:58:06 -0600 (Sat, 21 Jun 2008) | 17 lines Merged revisions 124540 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r124540 | murf | 2008-06-21 20:54:52 -0600 (Sat, 21 Jun 2008) | 9 lines (closes issue #12910) Reported by: chris-mac Sorry, my testing did not contain the simple case of forkCDR(v), I am much embarrassed to admit. If I had, I would have more solidly initialized the opts element for varset. ........ ................ 2008-06-21 12:54 +0000 [r124397-124506] Tilghman Lesher * /, res/res_config_ldap.c: Merged revisions 124505 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r124505 | tilghman | 2008-06-21 07:53:48 -0500 (Sat, 21 Jun 2008) | 4 lines Reduce warning to debug, otherwise we flood the log when we (legitimately) can't find a record. (Closes issue #12908) ........ * apps/app_rpt.c, /: Merged revisions 124451 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r124451 | tilghman | 2008-06-20 18:13:21 -0500 (Fri, 20 Jun 2008) | 14 lines Merged revisions 124450 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r124450 | tilghman | 2008-06-20 18:12:33 -0500 (Fri, 20 Jun 2008) | 6 lines usleep with a value over 1,000,000 is nonportable. Changing to use sleep() instead. (closes issue #12814) Reported by: pputman Patches: app_rtp_sleep.patch uploaded by pputman (license 81) ........ ................ * /, main/app.c: Merged revisions 124396 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r124396 | tilghman | 2008-06-20 17:04:37 -0500 (Fri, 20 Jun 2008) | 11 lines Merged revisions 124395 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r124395 | tilghman | 2008-06-20 17:02:55 -0500 (Fri, 20 Jun 2008) | 3 lines If the last character in a string to be parsed is the delimiter, then we should count that final empty string as an additional argument. ........ ................ 2008-06-20 21:48 +0000 [r124394] Jeff Gehlbach * doc/asterisk-mib.txt, /, doc/digium-mib.txt: Merged revisions 124392-124393 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r124392 | jeffg | 2008-06-20 17:36:01 -0400 (Fri, 20 Jun 2008) | 9 lines Merged revisions 124372 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r124372 | jeffg | 2008-06-20 17:14:40 -0400 (Fri, 20 Jun 2008) | 1 line Fix issues in digium-mib.txt and asterisk-mib.txt to placate smilint - bug 12905 ........ ................ r124393 | jeffg | 2008-06-20 17:43:18 -0400 (Fri, 20 Jun 2008) | 12 lines (Missed committing . on previous commit.....) Merged revisions 124372 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r124372 | jeffg | 2008-06-20 17:14:40 -0400 (Fri, 20 Jun 2008) | 1 line Fix issues in digium-mib.txt and asterisk-mib.txt to placate smilint - bug 12905 ........ ................ ................ 2008-06-20 20:18 +0000 [r124317] Tilghman Lesher * channels/chan_local.c, /: Merged revisions 124316 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r124316 | tilghman | 2008-06-20 15:17:04 -0500 (Fri, 20 Jun 2008) | 16 lines Merged revisions 124315 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r124315 | tilghman | 2008-06-20 15:16:02 -0500 (Fri, 20 Jun 2008) | 8 lines When using a Local channel, started by a call file, with a destination of an AGI script, the AGI script does not always get notified of a hangup if the underlying channel hangs up early. (closes issue #11833) Reported by: IgorG Patches: local_hangup-v1.diff uploaded by IgorG (license 20) ........ ................ 2008-06-20 16:31 +0000 [r124244-124279] Mark Michelson * main/ast_expr2.fl, include/asterisk/doxyref.h, /, main/ast_expr2f.c: Merged revisions 124278 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r124278 | mmichelson | 2008-06-20 11:30:18 -0500 (Fri, 20 Jun 2008) | 6 lines Change references to doc/channelvariables.txt to doc/tex/channelvariables.tex. This issue came up on the asterisk-dev mailing list. ........ * /, channels/chan_sip.c: Merged revisions 124243 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r124243 | mmichelson | 2008-06-20 10:20:11 -0500 (Fri, 20 Jun 2008) | 9 lines Add a missing "ChannelType" header to one of the "PeerStatus" manager events in chan_sip (closes issue #12904) Reported by: eliel Patches: chan_sip.c.patch uploaded by eliel (license 64) ........ 2008-06-19 23:02 +0000 [r124184] Tilghman Lesher * channels/chan_dahdi.c, /: Merged revisions 124183 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r124183 | tilghman | 2008-06-19 17:59:41 -0500 (Thu, 19 Jun 2008) | 15 lines Merged revisions 124182 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r124182 | tilghman | 2008-06-19 17:53:22 -0500 (Thu, 19 Jun 2008) | 7 lines It's possible for a hangup to be received, even just after the initial cid spill. (closes issue #12453) Reported by: Alex728 Patches: 20080604__bug12453.diff.txt uploaded by Corydon76 (license 14) ........ ................ 2008-06-19 20:32 +0000 [r124124] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 124121 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r124121 | mmichelson | 2008-06-19 15:30:23 -0500 (Thu, 19 Jun 2008) | 16 lines Merged revisions 124112 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r124112 | mmichelson | 2008-06-19 15:28:41 -0500 (Thu, 19 Jun 2008) | 8 lines Fix IMAP forwarding so that messages are sent to the proper mailbox. (closes issue #12897) Reported by: jaroth Patches: destination_forward.patch uploaded by jaroth (license 50) ........ ................ 2008-06-19 19:49 +0000 [r124065] Brett Bryant * /, main/utils.c: Merged revisions 124064 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r124064 | bbryant | 2008-06-19 14:48:26 -0500 (Thu, 19 Jun 2008) | 2 lines Add errors that report any locks held by threads when they are being closed. ........ 2008-06-19 18:57 +0000 [r124026] Brett Bryant * /, channels/chan_sip.c: Merged revisions 124024 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r124024 | bbryant | 2008-06-19 13:57:04 -0500 (Thu, 19 Jun 2008) | 2 lines Fix bug in sip registration that sets the default port to 5060 for tls. ........ 2008-06-19 17:58 +0000 [r123871-123989] Tilghman Lesher * /, res/res_config_ldap.c: Merged revisions 123952 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r123952 | tilghman | 2008-06-19 12:22:27 -0500 (Thu, 19 Jun 2008) | 6 lines Don't change pointers that need to be later passed back for deallocation. (closes issue #12572) Reported by: flyn Patches: 20080613__bug12572.diff.txt uploaded by Corydon76 (license 14) ........ * main/channel.c, /: Merged revisions 123931 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123931 | tilghman | 2008-06-19 12:02:54 -0500 (Thu, 19 Jun 2008) | 13 lines Merged revisions 123930 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123930 | tilghman | 2008-06-19 11:58:19 -0500 (Thu, 19 Jun 2008) | 5 lines Change informative messages to use the _multiple variant when multiple formats are possible. (Closes issue #12848) Reported by klaus3000 ........ ................ * /, build_tools/strip_nonapi: Merged revisions 123913 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123913 | tilghman | 2008-06-19 11:26:50 -0500 (Thu, 19 Jun 2008) | 13 lines Merged revisions 123909 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123909 | tilghman | 2008-06-19 11:26:03 -0500 (Thu, 19 Jun 2008) | 5 lines Only process 40 arguments (20 files) at once with xargs, because some older shells may force xargs to separate on an odd boundary. (Closes issue #12883) Reported by Nik Soggia ........ ................ * /, configs/sip.conf.sample: Merged revisions 123887 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123887 | tilghman | 2008-06-19 11:21:32 -0500 (Thu, 19 Jun 2008) | 12 lines Merged revisions 123883 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123883 | tilghman | 2008-06-19 11:20:41 -0500 (Thu, 19 Jun 2008) | 4 lines Correct description of notifyringing option. (Closes issue #12890) Reported by gminet ........ ................ * main/asterisk.c, /: Merged revisions 123870 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123870 | tilghman | 2008-06-19 11:08:29 -0500 (Thu, 19 Jun 2008) | 14 lines Merged revisions 123869 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123869 | tilghman | 2008-06-19 11:07:23 -0500 (Thu, 19 Jun 2008) | 6 lines The RDTSC instruction was introduced on the Pentium line of microprocessors, and is not compatible with certain 586 clones, like Cyrix. Hence, asking for i386 compatibility was always incorrect. See http://en.wikipedia.org/wiki/RDTSC (Closes issue #12886) Reported by tecnoxarxa ........ ................ 2008-06-18 22:18 +0000 [r123718-123772] Tilghman Lesher * /, main/say.c, doc/lang (added), doc/lang/hebrew.ods: Merged revisions 123770 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123770 | tilghman | 2008-06-18 17:17:17 -0500 (Wed, 18 Jun 2008) | 16 lines Merged revisions 123769 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123769 | tilghman | 2008-06-18 17:08:30 -0500 (Wed, 18 Jun 2008) | 8 lines Add support for saying numbers in Hebrew. (closes issue #11662) Reported by: greenfieldtech Patches: say.c.patch-12042008 uploaded by greenfieldtech (license 369) Hebrew-Sounds.ods uploaded by greenfieldtech (with signficant changes to the spreadsheet by me) ........ ................ * pbx/pbx_spool.c, /: Merged revisions 123715 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123715 | tilghman | 2008-06-18 15:23:58 -0500 (Wed, 18 Jun 2008) | 15 lines Merged revisions 123710 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123710 | tilghman | 2008-06-18 15:22:42 -0500 (Wed, 18 Jun 2008) | 7 lines Set the variables top-down, so that if a script sets a variable more than once, the last one will take precedence. (closes issue #12673) Reported by: phber Patches: 20080519__bug12673.diff.txt uploaded by Corydon76 (license 14) ........ ................ 2008-06-18 20:08 +0000 [r123693] Brett Bryant * main/tcptls.c, /: Merged revisions 123692 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r123692 | bbryant | 2008-06-18 15:07:56 -0500 (Wed, 18 Jun 2008) | 2 lines Fix a crash in tcp and tls connections related to reference counts. ........ 2008-06-18 15:09 +0000 [r123651-123653] Mark Michelson * /, apps/app_queue.c: Merged revisions 123652 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r123652 | mmichelson | 2008-06-18 10:08:56 -0500 (Wed, 18 Jun 2008) | 7 lines A portion of the code which handled the 'c' queue option had been removed. No telling when it happened. Anyway, it's back in now and works properly. (Based on issue reported on mailing list) ........ 2008-06-18 12:34 +0000 [r123646-123647] Russell Bryant * apps/app_fax.c: don't use trunk only API for frame data (closes issue #12881) * apps/app_fax.c (added): re-add app_fax ... it got accidentally removed (closes issue #12881) 2008-06-17 21:57 +0000 [r123547] Brett Bryant * main/tcptls.c, main/manager.c, /, channels/chan_sip.c, main/http.c, include/asterisk/tcptls.h: Merged revisions 123546 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r123546 | bbryant | 2008-06-17 16:46:57 -0500 (Tue, 17 Jun 2008) | 5 lines Updates all usages of ast_tcptls_session_instance to be managed by reference counts so that they only get destroyed when all threads are done using them, and memory does not get free'd causing strange issues with SIP. This code was originally written by russellb in the team/group/issue_11972/ branch. ........ 2008-06-17 21:34 +0000 [r123487-123542] Mark Michelson * /, channels/chan_sip.c: Merged revisions 123486 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123486 | mmichelson | 2008-06-17 15:28:47 -0500 (Tue, 17 Jun 2008) | 12 lines Merged revisions 123485 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123485 | mmichelson | 2008-06-17 15:26:38 -0500 (Tue, 17 Jun 2008) | 4 lines Make chan_sip build under dev mode with compilers >= GCC 4.2 Thanks to jpeeler for alerting me of this ........ ................ 2008-06-17 20:23 +0000 [r123473] Steve Murphy * /: block 123448 from trunk; it doesn't apply here. 2008-06-17 19:01 +0000 [r123394] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 123392 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123392 | tilghman | 2008-06-17 13:57:45 -0500 (Tue, 17 Jun 2008) | 11 lines Merged revisions 123391 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123391 | tilghman | 2008-06-17 13:56:53 -0500 (Tue, 17 Jun 2008) | 3 lines Fix 3 more places where failure to lock the structure could cause the wrong lock to be unlocked. (Closes issue #12795) ........ ................ 2008-06-17 18:28 +0000 [r123382-123387] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 123238 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r123238 | jpeeler | 2008-06-16 18:05:18 -0500 (Mon, 16 Jun 2008) | 1 line Fix some (more) variables that were forgotten to be renamed, related to 117658 ........ 2008-06-17 18:10 +0000 [r123335] Mark Michelson * /, channels/chan_sip.c: Merged revisions 123334 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123334 | mmichelson | 2008-06-17 13:09:54 -0500 (Tue, 17 Jun 2008) | 19 lines Merged revisions 123333 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123333 | mmichelson | 2008-06-17 13:09:16 -0500 (Tue, 17 Jun 2008) | 11 lines Cisco BTS sends SIP responses with a tab between the Cseq number and SIP request method in the Cseq: header. Asterisk did not handle this properly, but with this patch, all is well. (closes issue #12834) Reported by: tobias_e Patches: 12834.patch uploaded by putnopvut (license 60) Tested by: tobias_e ........ ................ 2008-06-17 18:08 +0000 [r123332] Jeff Peeler * doc/tex/configuration.tex, configs/zapata.conf.sample, Makefile, doc/janitor-projects.txt, configs/vpb.conf.sample, doc/sms.txt, contrib/scripts/loadtest.tcl, codecs/codec_dahdi.c (added), configs/smdi.conf.sample, pbx/pbx_config.c, apps/app_chanspy.c, main/asterisk.c, configs/users.conf.sample, doc/ss7.txt, apps/app_meetme.c, configs/rpt.conf.sample, doc/backtrace.txt, doc/tex/queues-with-callback-members.tex, include/asterisk/dahdi.h (added), configs/extensions.ael.sample, res/res_musiconhold.c, configs/meetme.conf.sample, codecs/codec_zap.c (removed), contrib/init.d/rc.mandrake.zaptel, cdr/cdr_csv.c, main/channel.c, doc/tex/manager.tex, doc/tex/sla.tex, include/asterisk/dsp.h, doc/tex/localchannel.tex, apps/app_rpt.c, channels/chan_mgcp.c, contrib/scripts/autosupport, doc/manager_1_1.txt, channels/chan_zap.c (removed), doc/asterisk.8, doc/tex/ael.tex, doc/tex/channelvariables.tex, apps/app_getcpeid.c, doc/tex/enum.tex, apps/app_queue.c, configs/sla.conf.sample, doc/tex/security.tex, include/asterisk/zapata.h (removed), doc/tex/privacy.tex, build_tools/menuselect-deps.in, apps/app_flash.c, main/file.c, doc/osp.txt, contrib/utils/zones2indications.c, utils/extconf.c, makeopts.in, configs/extensions.conf.sample, doc/asterisk.sgml, README, contrib/init.d/rc.mandrake.asterisk, /, include/asterisk/autoconfig.h.in, apps/app_dahdiscan.c (added), apps/app_chanisavail.c, channels/chan_iax2.c, configs/muted.conf.sample, main/loader.c, channels/chan_dahdi.c (added), include/asterisk/doxyref.h, configure, doc/tex/backtrace.tex, apps/app_zapscan.c (removed), doc/tex/app-sms.tex, apps/app_zapras.c (removed), configs/extensions.lua.sample, include/asterisk/options.h, contrib/init.d/rc.suse.asterisk, apps/app_dial.c, apps/app_page.c, doc/tex/hardware.tex, apps/app_fax.c (removed), apps/app_dahdiras.c (added), configure.ac, configs/queues.conf.sample, include/asterisk/channel.h: Goodbye Zaptel, hello DAHDI. Removes Zaptel driver support with DAHDI. Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow. 2008-06-17 15:58 +0000 [r123276] Mark Michelson * /, apps/app_queue.c: Merged revisions 123275 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123275 | mmichelson | 2008-06-17 10:57:43 -0500 (Tue, 17 Jun 2008) | 20 lines Merged revisions 123274 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123274 | mmichelson | 2008-06-17 10:56:55 -0500 (Tue, 17 Jun 2008) | 12 lines davidw pointed out that the holdtime calculation used by app_queue does not use "boxcar" filtering as the comments say. The term "boxcar" means that the number of samples used to calculate stays constant, with new samples replacing the oldest ones. The queue holdtime calculation uses all holdtime samples collected since the queue was loaded, so the comment has been changed to be accurate. (closes issue #12781) Reported by: davidw ........ ................ 2008-06-17 15:52 +0000 [r123273] Russell Bryant * main/astobj2.c, /: Merged revisions 123272 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123272 | russell | 2008-06-17 10:52:13 -0500 (Tue, 17 Jun 2008) | 12 lines Merged revisions 123271 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123271 | russell | 2008-06-17 10:48:31 -0500 (Tue, 17 Jun 2008) | 4 lines Fix a memory leak in astobj2 that was pointed out by seanbright. When a container got destroyed, the underlying bucket list entry for each object that was in the container at that time did not get free'd. ........ ................ 2008-06-16 21:20 +0000 [r123178] Jeff Peeler * channels/chan_zap.c: Fix some variables that were forgotten to be renamed, related to 117658. Couldn't merge from trunk since the chan_dahdi transition has not occurred here yet 2008-06-16 21:19 +0000 [r123173] Steve Murphy * apps/app_stack.c, apps/app_dial.c, main/pbx.c, /, main/features.c, include/asterisk/pbx.h, apps/app_queue.c: Merged revisions 123165 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r123165 | murf | 2008-06-16 14:43:46 -0600 (Mon, 16 Jun 2008) | 19 lines (closes issue #12689) Reported by: ys Many thanks to ys for doing the research on this problem. I didn't think it would be best to unlock the contexts and then relock them after the remove_extension2() call, so I added an extra arg to remove_extension2() and set it appropriately in each call. There were not that many. I considered forcing the code to lock the contexts before the call to remove_extension2(), but that would require a slightly greater degree of changes, especially since the find_context_locked is local to pbx.c I did a simple sanity test to make sure the code doesn't mess things up in general. ........ 2008-06-16 20:03 +0000 [r123112-123116] Tilghman Lesher * channels/chan_mgcp.c, /, channels/chan_sip.c, channels/chan_skinny.c, channels/chan_h323.c, channels/chan_iax2.c: Merged revisions 123114 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123114 | tilghman | 2008-06-16 14:57:05 -0500 (Mon, 16 Jun 2008) | 10 lines Merged revisions 123113 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123113 | tilghman | 2008-06-16 14:50:12 -0500 (Mon, 16 Jun 2008) | 2 lines Port "hasvoicemail" change from SIP to other channel drivers ........ ................ * /, channels/chan_sip.c: Merged revisions 123111 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123111 | tilghman | 2008-06-16 14:23:51 -0500 (Mon, 16 Jun 2008) | 16 lines Merged revisions 123110 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123110 | tilghman | 2008-06-16 14:21:58 -0500 (Mon, 16 Jun 2008) | 8 lines People expect that if "hasvoicemail" is set in users.conf, even if "mailbox" isn't set, that SIP will detect a mailbox. (closes issue #12855) Reported by: PLL Patches: 20080614__bug12855__2.diff.txt uploaded by Corydon76 (license 14) Tested by: PLL ........ ................ 2008-06-16 17:29 +0000 [r123075] Chris Tooley * apps/app_externalivr.c: Fixes and closes bug number 12804 2008-06-16 12:32 +0000 [r122871-122921] Joshua Colp * /, channels/chan_sip.c: Merged revisions 122920 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r122920 | file | 2008-06-16 09:32:02 -0300 (Mon, 16 Jun 2008) | 14 lines Merged revisions 122919 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122919 | file | 2008-06-16 09:31:09 -0300 (Mon, 16 Jun 2008) | 6 lines Only compare the first 15 characters so that even if the charset is specified we still accept it as SDP. (closes issue #12803) Reported by: lanzaandrea Patches: chan_sip.c.diff uploaded by lanzaandrea (license 496) ........ ................ * /, channels/chan_sip.c: Merged revisions 122870 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r122870 | file | 2008-06-16 09:09:54 -0300 (Mon, 16 Jun 2008) | 14 lines Merged revisions 122869 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122869 | file | 2008-06-16 09:08:28 -0300 (Mon, 16 Jun 2008) | 6 lines Don't send a BYE on a dialog that is already gone during a REFER. (closes issue #12865) Reported by: flefoll Patches: chan_sip.c.br14.121495.patch-ALREADYGONE uploaded by flefoll (license 244) ........ ................ 2008-06-13 21:47 +0000 [r122715] Mark Michelson * main/autoservice.c, /: Merged revisions 122714 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r122714 | mmichelson | 2008-06-13 16:45:21 -0500 (Fri, 13 Jun 2008) | 17 lines Merged revisions 122713 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122713 | mmichelson | 2008-06-13 16:44:53 -0500 (Fri, 13 Jun 2008) | 9 lines Short circuit the loop in autoservice_run if there are no channels to poll. If we continued, then the result would be calling poll() with a NULL pollfd array. While this is fine with POSIX's poll(2) system call, those who use Asterisk's internal poll mechanism (Darwin systems) would have a failed assertion occur when poll is called. (related to issue #10342) ........ ................ 2008-06-13 14:15 +0000 [r122558] Tilghman Lesher * main/dial.c, /: Merged revisions 122557 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r122557 | tilghman | 2008-06-13 09:15:07 -0500 (Fri, 13 Jun 2008) | 7 lines Convert one more delimiter to use comma. (closes issue #12850) Reported by: bcnit Patches: 20080613__bug12850.diff.txt uploaded by Corydon76 (license 14) Tested by: bcnit ........ 2008-06-13 00:18 +0000 [r122467] Jeff Peeler * apps/app_parkandannounce.c, /, main/features.c: Merged revisions 122433 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r122433 | jpeeler | 2008-06-12 18:08:37 -0500 (Thu, 12 Jun 2008) | 4 lines (closes issue 0012193) Reported by: davidw Patch by: Corydon76, modified by me to work properly with ParkAndAnnounce app ........ 2008-06-12 18:54 +0000 [r122313] Mark Michelson * /, apps/app_queue.c: Merged revisions 122312 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r122312 | mmichelson | 2008-06-12 13:53:17 -0500 (Thu, 12 Jun 2008) | 17 lines Merged revisions 122311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122311 | mmichelson | 2008-06-12 13:50:58 -0500 (Thu, 12 Jun 2008) | 9 lines Properly play a holdtime message if the announce-holdtime option is set to "once." (closes issue #12842) Reported by: ramonpeek Patches: patch001.diff uploaded by ramonpeek (license 266) ........ ................ 2008-06-12 18:24 +0000 [r122242-122266] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 122262 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r122262 | russell | 2008-06-12 13:23:54 -0500 (Thu, 12 Jun 2008) | 11 lines Merged revisions 122259 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122259 | russell | 2008-06-12 13:22:44 -0500 (Thu, 12 Jun 2008) | 3 lines Fix some race conditions that cause ast_assert() to report that chan_iax2 tried to remove an entry that wasn't in the scheduler ........ ................ 2008-06-12 15:27 +0000 [r122132-122180] Tilghman Lesher * /, apps/app_meetme.c: Merged revisions 122174 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r122174 | tilghman | 2008-06-12 10:26:07 -0500 (Thu, 12 Jun 2008) | 16 lines Merged revisions 122137 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122137 | tilghman | 2008-06-12 10:18:39 -0500 (Thu, 12 Jun 2008) | 8 lines Flipflop the sections for two options, since the section for 'X' (exit context) may otherwise absorb keypresses meant for 's' (admin/user menu). (closes issue #12836) Reported by: blitzrage Patches: 20080611__bug12836.diff.txt uploaded by Corydon76 (license 14) Tested by: blitzrage ........ ................ * main/channel.c, /: Merged revisions 122131 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r122131 | tilghman | 2008-06-12 10:14:37 -0500 (Thu, 12 Jun 2008) | 12 lines Merged revisions 122130 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122130 | tilghman | 2008-06-12 10:11:30 -0500 (Thu, 12 Jun 2008) | 4 lines Occasionally, the alertpipe loses its nonblocking status, so detect and correct that situation before it causes a deadlock. (Reported and tested by ctooley via #asterisk-dev) ........ ................ 2008-06-12 15:01 +0000 [r122126-122129] Steve Murphy * main/cdr.c, apps/app_forkcdr.c, /, CHANGES: Merged revisions 122128 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r122128 | murf | 2008-06-12 08:56:26 -0600 (Thu, 12 Jun 2008) | 9 lines Merged revisions 122127 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122127 | murf | 2008-06-12 08:51:44 -0600 (Thu, 12 Jun 2008) | 1 line Arkadia tried to warn me, but the code added to ast_cdr_busy, _failed, and _noanswer was redundant. Didn't spot it until I was resolving conflicts in trunk. Ugh. Redundant code removed. It wasn't harmful. Just dumb. ........ ................ * main/cdr.c, apps/app_forkcdr.c, /, funcs/func_cdr.c, include/asterisk/cdr.h, CHANGES: Merged revisions 122091 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r122091 | murf | 2008-06-12 08:28:01 -0600 (Thu, 12 Jun 2008) | 45 lines Merged revisions 122046 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122046 | murf | 2008-06-12 07:47:34 -0600 (Thu, 12 Jun 2008) | 37 lines (closes issue #10668) Reported by: arkadia Tested by: murf, arkadia Options added to forkCDR() app and the CDR() func to remove some roadblocks for CDR applications. The "show application ForkCDR" output was upgraded to more fully explain the inner workings of forkCDR. The A option was added to forkCDR to force the CDR system to NOT change the disposition on the original CDR, after the fork. This involves ast_cdr_answer, _busy, _failed, and so on. The T option was added to forkCDR to force obedience of the cdr LOCKED flag in the ast_cdr_end, all the disposition changing funcs (ast_cdr_answer, etc), and in the ast_cdr_setvar func. The CHANGES file was updated to explain ALL the new options added to satisfy this bug report (and some requests made verbally and via email, irc, etc, over the past months/year) The 's' option was added to the CDR() func, to force it to skip LOCKED cdr's in the chain. Again, the new options should be totally transparent to existing apps! Current behavior of CDR, forkCDR, and the rest of the CDR system should not change one little bit. Until you add the new options, at least! ........ ................ 2008-06-11 18:57 +0000 [r121915] Matthew Fredrickson * channels/chan_zap.c, /: Merged revisions 121914 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r121914 | mattf | 2008-06-11 13:53:10 -0500 (Wed, 11 Jun 2008) | 1 line Fix pseudo channel allocation errors on startup when using SS7 ........ 2008-06-11 18:20 +0000 [r121872] Tilghman Lesher * main/sched.c, main/channel.c, /, channels/chan_agent.c, main/abstract_jb.c: Merged revisions 121867 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r121867 | tilghman | 2008-06-11 13:19:24 -0500 (Wed, 11 Jun 2008) | 11 lines Merged revisions 121861 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r121861 | tilghman | 2008-06-11 13:18:16 -0500 (Wed, 11 Jun 2008) | 3 lines Make calls to ast_assert() actually test something, so that the error message printed is not nonsensical (reported by mvanbaak via #asterisk-bugs). ........ ................ 2008-06-11 17:59 +0000 [r121858] Matthew Fredrickson * channels/chan_zap.c, /: Merged revisions 121857 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r121857 | mattf | 2008-06-11 12:50:17 -0500 (Wed, 11 Jun 2008) | 1 line Make sure we hangup any calls we have and NULL out the ss7call value when we get a reset circuit message. Fixes crash bug ........ 2008-06-11 17:45 +0000 [r121856] Tilghman Lesher * contrib/scripts/realtime_pgsql.sql, /, UPGRADE.txt, include/asterisk/cdr.h: Merged revisions 121855 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r121855 | tilghman | 2008-06-11 12:44:39 -0500 (Wed, 11 Jun 2008) | 3 lines Expand CDR uniqueid field to 150 chars, to account for maximum systemname. (Closes issue #12831) ........ 2008-06-11 16:13 +0000 [r121806] Jeff Peeler * /, doc/backtrace.txt: Merged revisions 121805 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r121805 | jpeeler | 2008-06-11 11:11:40 -0500 (Wed, 11 Jun 2008) | 9 lines Merged revisions 121804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r121804 | jpeeler | 2008-06-11 11:11:09 -0500 (Wed, 11 Jun 2008) | 1 line add instructions for logging gdb output via set logging on ........ ................ 2008-06-10 18:36 +0000 [r121598] Sean Bright * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 121597 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r121597 | seanbright | 2008-06-10 14:35:37 -0400 (Tue, 10 Jun 2008) | 14 lines Merged revisions 121596 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r121596 | seanbright | 2008-06-10 14:34:45 -0400 (Tue, 10 Jun 2008) | 6 lines Fixes a problem with some buggy versions of GNU awk (3.1.3) not liking carriage returns in scripts. (closes issue #12749) Reported by: alinux Tested by: Laureano (on #asterisk-dev), juggie ........ ................ 2008-06-10 12:55 +0000 [r121445] Joshua Colp * main/channel.c, /: Merged revisions 121444 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r121444 | file | 2008-06-10 09:54:39 -0300 (Tue, 10 Jun 2008) | 12 lines Merged revisions 121442 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r121442 | file | 2008-06-10 09:52:06 -0300 (Tue, 10 Jun 2008) | 4 lines Update BRIDGEPEER variable before we do a generic bridge in case we just broke out of a native bridge and fell through to generic. (closes issue #12815) Reported by: ramonpeek ........ ................ 2008-06-10 00:53 +0000 [r121404-121408] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 121407 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r121407 | russell | 2008-06-09 19:52:46 -0500 (Mon, 09 Jun 2008) | 2 lines Bump up the debug level of a couple of messages ........ 2008-06-09 16:37 +0000 [r121283] Russell Bryant * main/channel.c, /: Merged revisions 121282 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r121282 | russell | 2008-06-09 11:37:08 -0500 (Mon, 09 Jun 2008) | 18 lines Merged revisions 121280 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r121280 | russell | 2008-06-09 11:35:40 -0500 (Mon, 09 Jun 2008) | 10 lines Do not attempt to do emulation if an END digit is received and the length is less than the defined minimum digit length, and the other end only wants END digits (SIP INFO, for example). (closes issue #12778) Reported by: tsearle Patches: 12778.rev1.txt uploaded by russell (license 2) Tested by: tsearle ........ ................ 2008-06-09 16:36 +0000 [r121281] Tilghman Lesher * main/pbx.c, /: Merged revisions 121279 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r121279 | tilghman | 2008-06-09 11:35:06 -0500 (Mon, 09 Jun 2008) | 6 lines Implement FINDLABEL matching for the new extension matching engine. (closes issue #12800) Reported by: chris-mac Patches: 20080608__bug12800.diff.txt uploaded by Corydon76 (license 14) ........ 2008-06-09 15:10 +0000 [r121231] Mark Michelson * /, channels/chan_agent.c: Merged revisions 121230 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r121230 | mmichelson | 2008-06-09 10:08:58 -0500 (Mon, 09 Jun 2008) | 27 lines Merged revisions 121229 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (Note that this is being merged to trunk/1.6.0 because it may affect non-callback agents with ackcall set) ........ r121229 | mmichelson | 2008-06-09 10:02:37 -0500 (Mon, 09 Jun 2008) | 16 lines A unique situation of timeouts brought forth a failure situation for autologoff in chan_agent. If using AgentCallbackLogin-style agents, then if the timeout specified by the Dial() to reach the agent's phone was shorter than the timeout specified in queues.conf, then autologoff would only work if the caller hung up while the agent's phone was ringing. This patch allows autologoff to work in this situation when the call in queue transfers to the next available agent (as it would have if the timeout in queues.conf were less than the timeout in the Dial()). (closes issue #12754) Reported by: Rodrigo Patches: 12754.patch uploaded by putnopvut (license 60) Tested by: Rodrigo ........ ................ 2008-06-08 01:43 +0000 [r121138-121164] Jeff Peeler * /, channels/chan_console.c: Merged revisions 121163 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r121163 | jpeeler | 2008-06-07 20:41:59 -0500 (Sat, 07 Jun 2008) | 4 lines This was accidentally reverted. Fixes a bug where if a stream monitor thread was not created (caused from failure of opening or starting the stream) pthread_cancel was called with an invalid thread ID. ........ * apps/app_parkandannounce.c, /: Merged revisions 121131 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r121131 | jpeeler | 2008-06-07 20:16:25 -0500 (Sat, 07 Jun 2008) | 2 lines Fixes segfault when using ParkAndAnnounce. Also, loop made more efficient as announce template only needs to be checked until the number of colon separated arguments run out, not the entire pointer storage array. Was done in a similiar fashion in 1.4, but here we're using less variables. ........ 2008-06-07 14:19 +0000 [r121080] Russell Bryant * channels/chan_local.c, /, channels/chan_agent.c: Merged revisions 121079 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r121079 | russell | 2008-06-07 09:18:44 -0500 (Sat, 07 Jun 2008) | 15 lines Merged revisions 121078 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r121078 | russell | 2008-06-07 09:10:56 -0500 (Sat, 07 Jun 2008) | 7 lines Don't run LIST_HEAD_DESTROY on a STATIC list (closes issue #12807) Reported by: ys Patches: chan_agent_local.diff uploaded by ys (license 281) ........ ................ 2008-06-06 20:25 +0000 [r121011-121047] Tilghman Lesher * main/pbx.c, /: Merged revisions 121010 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r121010 | tilghman | 2008-06-06 14:55:08 -0500 (Fri, 06 Jun 2008) | 6 lines Make extension match characters case-insensitive. (closes issue #12777) Reported by: jsmith Patches: lower_case_patterns-trunk-v1.patch uploaded by jsmith (license 15) ........ 2008-06-06 18:31 +0000 [r120907-120961] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 120960 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r120960 | jpeeler | 2008-06-06 13:30:17 -0500 (Fri, 06 Jun 2008) | 9 lines Merged revisions 120959 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120959 | jpeeler | 2008-06-06 13:29:14 -0500 (Fri, 06 Jun 2008) | 1 line add another LOW_MEMORY define I forgot ........ ................ * /, channels/chan_sip.c: Merged revisions 120909 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r120909 | jpeeler | 2008-06-06 13:06:06 -0500 (Fri, 06 Jun 2008) | 9 lines Merged revisions 120908 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120908 | jpeeler | 2008-06-06 13:05:15 -0500 (Fri, 06 Jun 2008) | 1 line only define thread storage variable if necessary for LOW_MEMORY ........ ................ * channels/chan_sip.c: Merged revisions 120906 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r120906 | jpeeler | 2008-06-06 12:50:05 -0500 (Fri, 06 Jun 2008) | 16 lines Merged revisions 120863,120885 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120863 | jpeeler | 2008-06-06 10:33:15 -0500 (Fri, 06 Jun 2008) | 3 lines This fixes a crash when LOW_MEMORY is turned on. Two allocations of the ast_rtp struct that were previously allocated on the stack have been modified to use thread local storage instead. ........ r120885 | jpeeler | 2008-06-06 11:39:20 -0500 (Fri, 06 Jun 2008) | 2 lines Correction to commmit 120863, make sure proper destructor function is called as well define two thread storage local variables. ........ ................ 2008-06-06 17:35 +0000 [r120864-120905] Tilghman Lesher * /, apps/app_exec.c: Merged revisions 120904 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r120904 | tilghman | 2008-06-06 12:34:21 -0500 (Fri, 06 Jun 2008) | 3 lines For the purpose of making the changed syntax to ExecIf easier to transition, allow the deprecated syntax (fixed for jmls on -dev). ........ 2008-06-05 21:39 +0000 [r120829] Steve Murphy * main/pbx.c, /: Merged revisions 120828 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r120828 | murf | 2008-06-05 15:34:42 -0600 (Thu, 05 Jun 2008) | 1 line a small fix for a crash that occurs when compiling AEL with global vars ........ 2008-06-05 17:17 +0000 [r120677] Philippe Sultan * /, res/res_jabber.c: Merged revisions 120676 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r120676 | phsultan | 2008-06-05 19:02:39 +0200 (Thu, 05 Jun 2008) | 10 lines Merged revisions 120675 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120675 | phsultan | 2008-06-05 18:56:15 +0200 (Thu, 05 Jun 2008) | 2 lines Ignore appended resource when comparing JIDs. ........ ................ 2008-06-05 16:42 +0000 [r120643-120674] Brett Bryant 2008-06-05 16:01 +0000 [r120566-120603] Tilghman Lesher * apps/app_stack.c, main/loader.c, /, res/res_agi.c: Merged revisions 120602 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r120602 | tilghman | 2008-06-05 10:58:11 -0500 (Thu, 05 Jun 2008) | 4 lines Conditionally load the AGI command gosub, depending on whether or not res_agi has been loaded, fix a return value in the loader, and ensure that the help workhorse header does not print on load. ........ * /, UPGRADE.txt: Merged revisions 120567 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r120567 | tilghman | 2008-06-05 09:35:47 -0500 (Thu, 05 Jun 2008) | 2 lines Add info on the [compat] section of asterisk.conf. ........ * apps/app_fax.c: Fix frame API for 1.6.0 (Closes issue #12793) 2008-06-04 22:08 +0000 [r120515] Mark Michelson * /, apps/app_queue.c: Merged revisions 120514 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r120514 | mmichelson | 2008-06-04 17:07:37 -0500 (Wed, 04 Jun 2008) | 14 lines Merged revisions 120513 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120513 | mmichelson | 2008-06-04 17:05:33 -0500 (Wed, 04 Jun 2008) | 6 lines Make sure that the string we set will survive the unref of the queue member. Thanks to Russell, who pointed this out. ........ ................ 2008-06-04 20:35 +0000 [r120478] Tilghman Lesher * main/pbx.c, /: Merged revisions 120477 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r120477 | tilghman | 2008-06-04 15:34:52 -0500 (Wed, 04 Jun 2008) | 2 lines MSet doesn't necessarily need chan to be set ........ 2008-06-04 15:38 +0000 [r120338] Joshua Colp * /, pbx/pbx_config.c: Merged revisions 120337 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r120337 | file | 2008-06-04 12:38:00 -0300 (Wed, 04 Jun 2008) | 2 lines We like tabs. ........ 2008-06-04 14:13 +0000 [r120287] Mark Michelson * /, apps/app_queue.c: Merged revisions 120286 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r120286 | mmichelson | 2008-06-04 09:12:45 -0500 (Wed, 04 Jun 2008) | 15 lines Merged revisions 120285 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120285 | mmichelson | 2008-06-04 09:11:12 -0500 (Wed, 04 Jun 2008) | 7 lines Tab completion when removing a member should give the member's interface, not the name, since the interface is what is expected for the command. (closes issue #12783) Reported by: davevg ........ ................ 2008-06-04 13:34 +0000 [r120284] Joshua Colp * /, pbx/pbx_config.c: Merged revisions 120283 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r120283 | file | 2008-06-04 10:33:59 -0300 (Wed, 04 Jun 2008) | 14 lines Merged revisions 120282 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120282 | file | 2008-06-04 10:31:09 -0300 (Wed, 04 Jun 2008) | 6 lines Fix a log message and add a message for when the dialplan is done reloading. (closes issue #12716) Reported by: chappell Patches: dialplan_reload_2.diff uploaded by chappell (license 8) ........ ................ 2008-06-03 23:18 +0000 [r120228-120234] Tilghman Lesher * pbx/pbx_loopback.c, /: Merged revisions 120227 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r120227 | tilghman | 2008-06-03 17:42:03 -0500 (Tue, 03 Jun 2008) | 16 lines Merged revisions 120226 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120226 | tilghman | 2008-06-03 17:41:04 -0500 (Tue, 03 Jun 2008) | 8 lines Due to incorrect use of the AST_LIST_INSERT_HEAD() macro the loopback switch cannot perform any translation on the extension number before searching for it in the target context. (closes issue #12473) Reported by: chappell Patches: pbx_loopback.c.diff uploaded by chappell (license 8) ........ ................ 2008-06-03 22:18 +0000 [r120178] Jeff Peeler * main/config.c: Merged revisions 120174 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r120174 | jpeeler | 2008-06-03 17:17:07 -0500 (Tue, 03 Jun 2008) | 14 lines Merged revisions 120173 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120173 | jpeeler | 2008-06-03 17:15:33 -0500 (Tue, 03 Jun 2008) | 6 lines (closes issue #11594) Reported by: yem Tested by: yem This change decreases the buffer size allocated on the stack substantially in config_text_file_load when LOW_MEMORY is turned on. This change combined with the fix from revision 117462 (making mkintf not copy the zt_chan_conf structure) was enough to prevent the crash. ........ ................ 2008-06-03 22:08 +0000 [r120172] Tilghman Lesher * include/asterisk/options.h, main/asterisk.c, Makefile, main/pbx.c, /, res/res_agi.c, pbx/pbx_realtime.c, configs/pbx_realtime.conf (removed): Merged revisions 120171 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r120171 | tilghman | 2008-06-03 17:05:16 -0500 (Tue, 03 Jun 2008) | 5 lines Move compatibility options into asterisk.conf, default them to on for upgrades, and off for new installations. This includes the translation from pipes to commas for pbx_realtime and the EXEC command for AGI, as well as the change to the Set application not to support multiple variables at once. ........ 2008-06-03 21:35 +0000 [r120170] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 120169 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r120169 | russell | 2008-06-03 16:35:11 -0500 (Tue, 03 Jun 2008) | 12 lines Merged revisions 120168 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120168 | russell | 2008-06-03 16:34:55 -0500 (Tue, 03 Jun 2008) | 4 lines Fix another place where peer->callno could change at a very bad time, and also fix a place where a peer was used after the reference was released. (inspired by rev 120001) ........ ................ 2008-06-03 16:24 +0000 [r120034] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 120012 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r120012 | tilghman | 2008-06-03 11:19:35 -0500 (Tue, 03 Jun 2008) | 17 lines Merged revisions 120001 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120001 | tilghman | 2008-06-03 11:10:53 -0500 (Tue, 03 Jun 2008) | 9 lines Save the callno when we're poking, because our peer structure could change during destruction (and thus we unlock the wrong callno, causing a cascade failure). (closes issue #12717) Reported by: gewfie Patches: 20080525__bug12717.diff.txt uploaded by Corydon76 (license 14) Tested by: gewfie ........ ................ 2008-06-03 15:57 +0000 [r119931-120000] Steve Murphy * pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18, pbx/ael/ael-test/ref.ael-vtest21, pbx/ael/ael-test/ref.ael-test19, pbx/ael/ael-test/ref.ael-vtest13, pbx/ael/ael-test/ref.ael-vtest17, pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test5, pbx/ael/ael-test/ref.ael-test15: Merged revisions 119998 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119998 | murf | 2008-06-03 09:49:34 -0600 (Tue, 03 Jun 2008) | 16 lines Merged revisions 119966 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119966 | murf | 2008-06-03 09:26:56 -0600 (Tue, 03 Jun 2008) | 8 lines Updated the regressions on AEL. Hadn't updated this for the changes I made to preserve ${EXTEN} in switches, which affected several tests because it adds extra priorities, and at least one needed to be updated because of the removal of the empty extension warning message. ........ ................ * res/ael/pval.c, /: Merged revisions 119930 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119930 | murf | 2008-06-03 09:07:20 -0600 (Tue, 03 Jun 2008) | 24 lines Merged revisions 119929 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119929 | murf | 2008-06-03 08:49:46 -0600 (Tue, 03 Jun 2008) | 16 lines as per http://lists.digium.com/pipermail/asterisk-users/2008-June/212934.html, which is a message from Philipp Kempgen, requesting that the WARNING that an extension is empty be reduced to a NOTICE or less, as empty extensions are syntactically possible, and no big deal. With which I agree, and have removed that WARNING message entirely. I think it is not necessary to see this message. It didn't state that a NoOp() was inserted automatically on your behalf, and really, as users, who cares? Why freak out dialplan writers with unnecessary warnings? The details of the machinations a compiler goes thru to produce working assembly code is of little interest to most programmers-- we will follow the unix principal of doing our work silently. ........ ................ 2008-06-03 14:48 +0000 [r119928] Joshua Colp * /, channels/chan_sip.c: Merged revisions 119927 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119927 | file | 2008-06-03 11:47:54 -0300 (Tue, 03 Jun 2008) | 10 lines Merged revisions 119926 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119926 | file | 2008-06-03 11:46:24 -0300 (Tue, 03 Jun 2008) | 2 lines Treat ECONNREFUSED as an error that will stop further retransmissions. (issue #AST-58, patch from Switchvox) ........ ................ 2008-06-03 13:30 +0000 [r119745-119893] Russell Bryant * /, main/logger.c: Merged revisions 119892 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r119892 | russell | 2008-06-03 08:29:16 -0500 (Tue, 03 Jun 2008) | 9 lines Do a deep copy of file and function strings to avoid a potential crash when modules are unloaded. (closes issue #12780) Reported by: ys Patches: logger.diff uploaded by ys (license 281) -- modified by me for coding guidelines ........ * /, channels/chan_iax2.c: Merged revisions 119839 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119839 | russell | 2008-06-02 15:08:24 -0500 (Mon, 02 Jun 2008) | 15 lines Merged revisions 119838 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119838 | russell | 2008-06-02 15:08:04 -0500 (Mon, 02 Jun 2008) | 7 lines Revert a change made for issue #12479. This change caused a regression such that a dial string such as (IAX2/foo) did not automatically fall back to dialing the 's' extension anymore. (closes issue #12770) Reported by: dagmoller ........ ................ * /, apps/app_fax.c (added): Merged revisions 119801 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r119801 | russell | 2008-06-02 11:14:15 -0500 (Mon, 02 Jun 2008) | 4 lines Add app_fax from asterisk-addons, with some additional changes to resolve compiler warnings, as well as update to the APIs in spandsp 0.0.5. Spandsp 0.0.5 is being distributed under the LGPL, so we can move this module into the main tree. ........ * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 119799 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r119799 | russell | 2008-06-02 10:57:43 -0500 (Mon, 02 Jun 2008) | 4 lines After determining that the version of spandsp installed is an acceptable version, do a build and link test to ensure that the library is usable, and that libtiff is also available ........ * /, build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Merged revisions 119795 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r119795 | russell | 2008-06-02 10:43:40 -0500 (Mon, 02 Jun 2008) | 2 lines Add a configure script check for spandsp ........ * main/manager.c, /: Merged revisions 119744 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119744 | russell | 2008-06-02 09:41:55 -0500 (Mon, 02 Jun 2008) | 13 lines Merged revisions 119742 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119742 | russell | 2008-06-02 09:39:45 -0500 (Mon, 02 Jun 2008) | 5 lines Improve CLI command blacklist checking for the command manager action. Previously, it did not handle case or whitespace properly. This made it possible for blacklisted commands to get executed anyway. (closes issue #12765) ........ ................ 2008-06-02 14:40 +0000 [r119743] Philippe Sultan * channels/chan_jingle.c, /, channels/chan_gtalk.c, res/res_jabber.c: Merged revisions 119741 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r119741 | phsultan | 2008-06-02 16:35:24 +0200 (Mon, 02 Jun 2008) | 13 lines Do not link the guest account with any configured XMPP client (in jabber.conf). The actual connection is made when a call comes in Asterisk. Apply this fix to Jingle too. Fix the ast_aji_get_client function that was not able to retrieve an XMPP client from its JID. (closes issue #12085) Reported by: junky Tested by: phsultan ........ 2008-06-02 12:32 +0000 [r119532-119690] Russell Bryant * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, /: Merged revisions 119586,119637 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119586 | crichter | 2008-06-02 03:46:23 -0500 (Mon, 02 Jun 2008) | 9 lines Merged revisions 119585 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119585 | crichter | 2008-06-02 10:35:28 +0200 (Mo, 02 Jun 2008) | 1 line Added counter for unhandled_bmsg Print, this prevents the logs to be flooded to fast and save CPU in this error scenario. Added 'last_used' element to bc structure, when a bchannel changes from used to free this exact time will be marked in last_used. When a new channel is requested the find_free_chan function will check if the new empty channel was used within the last second, if yes it will search for the next channel, if no it will return this channel. This simple mechanism has prooven to prevent race conditions where the NT and TE tried to allocate the exact same channel at the same time (RELEASE cause: 44). ........ ................ r119637 | crichter | 2008-06-02 04:35:04 -0500 (Mon, 02 Jun 2008) | 9 lines Merged revisions 119636 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119636 | crichter | 2008-06-02 11:29:21 +0200 (Mo, 02 Jun 2008) | 1 line fixed compile issue when dev-mode is enabled ........ ................ * /, channels/chan_iax2.c: Merged revisions 119688 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119688 | russell | 2008-06-02 07:30:42 -0500 (Mon, 02 Jun 2008) | 11 lines Merged revisions 119687 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119687 | russell | 2008-06-02 07:30:17 -0500 (Mon, 02 Jun 2008) | 3 lines Even of the first PING or LAGRQ doesn't get sent because it comes up too soon, make sure to reschedule so it gets sent later. ........ ................ * /, channels/chan_iax2.c: Merged revisions 119534 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119534 | russell | 2008-06-01 20:08:16 -0500 (Sun, 01 Jun 2008) | 10 lines Merged revisions 119533 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119533 | russell | 2008-06-01 20:06:09 -0500 (Sun, 01 Jun 2008) | 2 lines Change a debug message to an actual debug message ........ ................ * apps/app_dial.c, /: Merged revisions 119531 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119531 | russell | 2008-06-01 20:04:01 -0500 (Sun, 01 Jun 2008) | 10 lines Merged revisions 119530 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119530 | russell | 2008-06-01 20:03:22 -0500 (Sun, 01 Jun 2008) | 2 lines Fix another typo in documentation ........ ................ 2008-06-01 21:59 +0000 [r119529] Michiel van Baak * apps/app_dial.c, /: Merged revisions 119479 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119479 | mvanbaak | 2008-06-01 23:06:27 +0200 (Sun, 01 Jun 2008) | 10 lines Merged revisions 119478 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119478 | mvanbaak | 2008-06-01 22:47:55 +0200 (Sun, 01 Jun 2008) | 2 lines small typo fix 'retires' => 'retries' ........ ................ 2008-05-30 21:24 +0000 [r119420] Tilghman Lesher * /, apps/app_queue.c: Merged revisions 119419 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119419 | tilghman | 2008-05-30 16:23:14 -0500 (Fri, 30 May 2008) | 14 lines Merged revisions 119404 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119404 | tilghman | 2008-05-30 16:17:45 -0500 (Fri, 30 May 2008) | 6 lines When joinempty=strict, it only failed on join if there were busy members. If all members were logged out OR paused, then it (incorrectly) let callers join the queue. (closes issue #12451) Reported by: davidw ........ ................ 2008-05-30 19:48 +0000 [r119356] Joshua Colp * main/autoservice.c, /: Merged revisions 119355 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119355 | file | 2008-05-30 16:47:30 -0300 (Fri, 30 May 2008) | 10 lines Merged revisions 119354 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119354 | file | 2008-05-30 16:46:37 -0300 (Fri, 30 May 2008) | 2 lines Fix a bug I found while testing for another issue. ........ ................ 2008-05-30 17:13 +0000 [r119304] Tilghman Lesher * apps/app_stack.c: Oops, broke 1.6 (thanks MattF) 2008-05-30 16:57 +0000 [r119303] Michiel van Baak * contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk, contrib/init.d/rc.debian.asterisk, contrib/init.d/rc.mandrake.asterisk, /, contrib/init.d/rc.redhat.asterisk, contrib/init.d/rc.gentoo.asterisk, contrib/init.d/rc.slackware.asterisk: Merged revisions 119302 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119302 | mvanbaak | 2008-05-30 18:47:24 +0200 (Fri, 30 May 2008) | 22 lines Merged revisions 119301 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119301 | mvanbaak | 2008-05-30 18:44:39 +0200 (Fri, 30 May 2008) | 14 lines dont use a bashism way to check the $VERSION variable. The rc/init.d scripts, and safe_asterisk work on normal sh now again. Tested on: OpenBSD 4.2 (me) Debian etch (me) Ubuntu Hardy (me and loloski) FC9 (loloski) (closes issue #12687) Reported by: loloski Patches: 20080529-12687-safe_asterisk-fixversion.diff.txt uploaded by mvanbaak (license 7) Tested by: loloski, mvanbaak ........ ................ 2008-05-30 16:40 +0000 [r119297-119300] Tilghman Lesher * apps/app_stack.c, /: Merged revisions 119299 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r119299 | tilghman | 2008-05-30 11:40:13 -0500 (Fri, 30 May 2008) | 2 lines Suppress warning about pbx structure already existing ........ * apps/app_stack.c, apps/app_dial.c, include/asterisk/agi.h, /, CHANGES: Merged revisions 119296 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r119296 | tilghman | 2008-05-30 11:10:46 -0500 (Fri, 30 May 2008) | 8 lines Add native AGI command GOSUB, as invoking Gosub with EXEC does not work properly. (closes issue #12760) Reported by: Corydon76 Patches: 20080530__bug12760.diff.txt uploaded by Corydon76 (license 14) Tested by: tim_ringenbach, Corydon76 ........ 2008-05-30 13:01 +0000 [r119158-119240] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 119239 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119239 | russell | 2008-05-30 07:59:11 -0500 (Fri, 30 May 2008) | 23 lines Merged revisions 119238 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r119238 | russell | 2008-05-30 07:55:36 -0500 (Fri, 30 May 2008) | 15 lines Merged revisions 119237 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r119237 | russell | 2008-05-30 07:49:39 -0500 (Fri, 30 May 2008) | 7 lines - Instead of only enforcing destination call number checking on an ACK, check all full frames except for PING and LAGRQ, which may be sent by older versions too quickly to contain the destination call number. (As suggested by Tim Panton on the asterisk-dev list) - Merge changes from team/russell/iax2-frame-race, which prevents PING and LAGRQ from being sent before the destination call number is known. ........ ................ ................ * main/autoservice.c, /: Merged revisions 119157 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119157 | russell | 2008-05-29 17:28:50 -0500 (Thu, 29 May 2008) | 18 lines Merged revisions 119156 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119156 | russell | 2008-05-29 17:24:29 -0500 (Thu, 29 May 2008) | 10 lines Fix a race condition in channel autoservice. There was still a small window of opportunity for a DTMF frame, or some other deferred frame type, to come in and get dropped. (closes issue #12656) (closes issue #12656) Reported by: dimas Patches: v3-12656.patch uploaded by dimas (license 88) -- with some modifications by me ........ ................ 2008-05-29 20:26 +0000 [r119073] Tilghman Lesher * channels/chan_zap.c, /: Merged revisions 119072 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119072 | tilghman | 2008-05-29 15:25:33 -0500 (Thu, 29 May 2008) | 15 lines Merged revisions 119071 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119071 | tilghman | 2008-05-29 15:24:11 -0500 (Thu, 29 May 2008) | 7 lines Call waiting tone occurs too often, because it's getting serviced by both subchannels. (closes issue #11354) Reported by: cahen Patches: 20080512__bug11354.diff.txt uploaded by Corydon76 (license 14) ........ ................ 2008-05-29 19:06 +0000 [r118960-119014] Russell Bryant * apps/app_milliwatt.c, /: Merged revisions 119013 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119013 | russell | 2008-05-29 14:05:33 -0500 (Thu, 29 May 2008) | 12 lines Merged revisions 119012 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119012 | russell | 2008-05-29 14:04:52 -0500 (Thu, 29 May 2008) | 4 lines - Fix a typo in the argument to Playtones - use ast_safe_sleep() instead of calling the wait application (thanks to tilghman for pointing these out!) ........ ................ * /, channels/chan_iax2.c: Merged revisions 119010 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119010 | russell | 2008-05-29 13:54:11 -0500 (Thu, 29 May 2008) | 24 lines Merged revisions 119009 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r119009 | russell | 2008-05-29 13:49:12 -0500 (Thu, 29 May 2008) | 16 lines Merged revisions 119008 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r119008 | russell | 2008-05-29 13:45:21 -0500 (Thu, 29 May 2008) | 7 lines Merge changes from team/russell/iax2-another-fix-to-the-fix As described in the following post to the asterisk-dev mailing list, only enforce destination call numbers when processing an ACK. http://lists.digium.com/pipermail/asterisk-dev/2008-May/033217.html (closes issue #12631) ........ ................ ................ * apps/app_milliwatt.c, /: Merged revisions 118962 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118962 | russell | 2008-05-29 12:52:00 -0500 (Thu, 29 May 2008) | 11 lines Merged revisions 118961 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118961 | russell | 2008-05-29 12:51:29 -0500 (Thu, 29 May 2008) | 3 lines - Mark app_milliwatt dependent on res_indications (thanks to jsmith) - fix a typo in a log message (thanks to qwell) ........ ................ * apps/app_milliwatt.c, /: Merged revisions 118959 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118959 | russell | 2008-05-29 12:46:04 -0500 (Thu, 29 May 2008) | 11 lines Merged revisions 118956 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118956 | russell | 2008-05-29 12:38:38 -0500 (Thu, 29 May 2008) | 3 lines Change milliwatt to use the proper tone by default (1004 Hz) instead of 1000 Hz. An option is there to use 1000 Hz for anyone that might want it. ........ ................ 2008-05-29 17:42 +0000 [r118958] Tilghman Lesher * channels/chan_mgcp.c, channels/chan_zap.c, /, channels/chan_agent.c, channels/chan_alsa.c, main/utils.c, include/asterisk/lock.h, channels/chan_iax2.c: Merged revisions 118955,118957 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118955 | tilghman | 2008-05-29 12:35:19 -0500 (Thu, 29 May 2008) | 11 lines Merged revisions 118953 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118953 | tilghman | 2008-05-29 12:20:16 -0500 (Thu, 29 May 2008) | 3 lines Add some debugging code that ensures that when we do deadlock avoidance, we don't lose the information about how a lock was originally acquired. ........ ................ r118957 | tilghman | 2008-05-29 12:39:50 -0500 (Thu, 29 May 2008) | 10 lines Merged revisions 118954 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118954 | tilghman | 2008-05-29 12:33:01 -0500 (Thu, 29 May 2008) | 2 lines Define also when not DEBUG_THREADS ........ ................ 2008-05-29 04:11 +0000 [r118909] Steve Murphy * main/cdr.c, apps/app_forkcdr.c, /: Merged revisions 118880 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118880 | murf | 2008-05-28 19:29:09 -0600 (Wed, 28 May 2008) | 54 lines Merged revisions 118858 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118858 | murf | 2008-05-28 18:25:28 -0600 (Wed, 28 May 2008) | 46 lines (closes issue #10668) (closes issue #11721) (closes issue #12726) Reported by: arkadia Tested by: murf These changes: 1. revert the changes made via bug 10668; I should have known that such changes, even tho they made sense at the time, seemed like an omission, etc, were actually integral to the CDR system via forkCDR. It makes sense to me now that forkCDR didn't natively end any CDR's, but rather depended on natively closing them all at hangup time via traversing and closing them all, whether locked or not. I still don't completely understand the benefits of setvar and answer operating on locked cdrs, but I've seen enough to revert those changes also, and stop messing up users who depended on that behavior. bug 12726 found reverting the changes fixed his changes, and after a long review and working on forkCDR, I can see why. 2. Apply the suggested enhancements proposed in 10668, but in a completely compatible way. ForkCDR will behave exactly as before, but now has new options that will allow some actions to be taken that will slightly modify the outcome and side-effects of forkCDR. Based on conversations I've had with various people, these small tweaks will allow some users to get the behavior they need. For instance, users executing forkCDR in an AGI script will find the answer time set, and DISPOSITION set, a situation not covered when the routines were first written. 3. A small problem in the cdr serializer would output answer and end times even when they were not set. This is now fixed. ........ ................ 2008-05-28 18:07 +0000 [r118781] Michiel van Baak * /, channels/chan_skinny.c: Merged revisions 118750 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r118750 | mvanbaak | 2008-05-28 19:58:21 +0200 (Wed, 28 May 2008) | 2 lines remove unused astobj.h header file from chan_skinny.c ........ 2008-05-28 14:31 +0000 [r118648] Joshua Colp * /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Merged revisions 118647 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118647 | file | 2008-05-28 11:29:01 -0300 (Wed, 28 May 2008) | 12 lines Merged revisions 118646 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118646 | file | 2008-05-28 11:23:34 -0300 (Wed, 28 May 2008) | 4 lines Add an option to use the source IP address of RTP as the destination IP address of UDPTL when a specific option is enabled. If the remote side is properly configured (ports forwarded) then UDPTL will flow. (closes issue #10417) Reported by: cstadlmann ........ ................ 2008-05-28 14:13 +0000 [r118615-118645] Philippe Sultan * channels/chan_jingle.c, /, include/asterisk/jingle.h: Merged revisions 118644 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r118644 | phsultan | 2008-05-28 16:10:48 +0200 (Wed, 28 May 2008) | 10 lines Changed to temporary namespaces to match with latest XEPs. As soon as Jingle is completely standardized, we can set those namespaces to their final values. Added two attributes to the jingle_pvt struct to store the content name attributes. Reported by Robert McQueen on Telepathy's framework mailing list : http://lists.freedesktop.org/archives/telepathy/2008-May/001971.html Keeping working on our Jingle stack! ........ * channels/chan_jingle.c, /: Merged revisions 118614 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r118614 | phsultan | 2008-05-28 10:39:10 +0200 (Wed, 28 May 2008) | 1 line Code simplification ........ 2008-05-27 19:35 +0000 [r118561] Joshua Colp * /, channels/chan_sip.c: Merged revisions 118560 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118560 | file | 2008-05-27 16:34:14 -0300 (Tue, 27 May 2008) | 12 lines Merged revisions 118558 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118558 | file | 2008-05-27 16:32:38 -0300 (Tue, 27 May 2008) | 4 lines Fix an issue where codec preferences were not set on dialogs that were not authenticated via a user or peer and allow framing to work without rtpmap in the SDP. (closes issue #12501) Reported by: slimey ........ ................ 2008-05-27 19:28 +0000 [r118557] Russell Bryant * /, include/asterisk/compat.h: Merged revisions 118556 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r118556 | russell | 2008-05-27 14:27:48 -0500 (Tue, 27 May 2008) | 6 lines Add printf format attribute for vasprintf(). (closes issue #12729) Reported by: snuffy Patches: bug_12729.diff uploaded by snuffy (license 35) ........ 2008-05-27 19:22 +0000 [r118555] Tilghman Lesher * main/cli.c, /: Merged revisions 118554 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118554 | tilghman | 2008-05-27 14:21:03 -0500 (Tue, 27 May 2008) | 14 lines Merged revisions 118551 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118551 | tilghman | 2008-05-27 14:15:27 -0500 (Tue, 27 May 2008) | 6 lines When showing an error message for a command, don't shorten the command output, as it tends to confuse the user (it's fine for suggesting other commands, however). Reported by: seanbright (on #asterisk-dev) Fixed by: me ........ ................ 2008-05-27 19:09 +0000 [r118518] Mark Michelson * apps/app_chanspy.c, /: Merged revisions 118514 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118514 | mmichelson | 2008-05-27 14:08:24 -0500 (Tue, 27 May 2008) | 19 lines Merged revisions 118509 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118509 | mmichelson | 2008-05-27 14:07:26 -0500 (Tue, 27 May 2008) | 11 lines Russell noted to me that in the case that separate threads use their own addressing system, the fix I made for issue 12376 does not guarantee uniqueness to the datastores' uids. Though I know of no system that works this way, I am going to change this right now to prevent trying to track down some future bug that may occur and cause untold hours of debugging time to track down. The change involves using a global counter which increases with each new chanspy_ds which is created. This guarantees uniqueness. ........ ................ 2008-05-27 18:59 +0000 [r118471] Tilghman Lesher * main/asterisk.c, /: Merged revisions 118466 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118466 | tilghman | 2008-05-27 13:59:06 -0500 (Tue, 27 May 2008) | 16 lines Merged revisions 118465 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118465 | tilghman | 2008-05-27 13:58:09 -0500 (Tue, 27 May 2008) | 8 lines NULL character should terminate only commands back to the core, not log messages to the console. (closes issue #12731) Reported by: seanbright Patches: 20080527__bug12731.diff.txt uploaded by Corydon76 (license 14) Tested by: seanbright ........ ................ 2008-05-27 17:25 +0000 [r118418] Michiel van Baak * apps/app_voicemail.c: small update to the g() option of app_voicemail to note that gain changes only work on zap channels right now. issue #12578 shows it's not clear right now. 2008-05-27 16:48 +0000 [r118378-118382] Mark Michelson * apps/app_chanspy.c, /: Merged revisions 118371 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118371 | mmichelson | 2008-05-27 11:43:36 -0500 (Tue, 27 May 2008) | 22 lines Merged revisions 118365 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118365 | mmichelson | 2008-05-27 11:38:38 -0500 (Tue, 27 May 2008) | 14 lines Add a unique id to the datastore allocated in app_chanspy since it is possible that multiple spies may be listening to the same channel. (closes issue #12376) Reported by: DougUDI Patches: 12376_chanspy_uid.diff uploaded by putnopvut (license 60) Tested by: destiny6628 (closes issue #12243) Reported by: atis ........ ................ * /: Hmm, I apparently forgot to commit the block of revision 118175. Now I'm doing it. 2008-05-27 15:47 +0000 [r118360] Tilghman Lesher * /, configs/queues.conf.sample: Merged revisions 118359 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118359 | tilghman | 2008-05-27 10:46:58 -0500 (Tue, 27 May 2008) | 11 lines Merged revisions 118358 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118358 | tilghman | 2008-05-27 10:45:37 -0500 (Tue, 27 May 2008) | 3 lines Add a note that pbx_config.so is needed for Local channels. (Closes issue #12671) ........ ................ 2008-05-27 14:51 +0000 [r118331] Russell Bryant * /, include/asterisk/compat.h: Merged revisions 118328 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r118328 | russell | 2008-05-27 09:51:13 -0500 (Tue, 27 May 2008) | 2 lines Add printf attribute to asprintf ........ 2008-05-27 13:30 +0000 [r118301-118303] Tilghman Lesher * /, res/res_config_ldap.c: Merged revisions 118302 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r118302 | tilghman | 2008-05-27 08:30:10 -0500 (Tue, 27 May 2008) | 6 lines When binding anonymously, credentials are still needed. (closes issue #12601) Reported by: suretec Patches: res_config_ldap.c.patch uploaded by suretec (license 70) ........ * /, pbx/pbx_realtime.c: Merged revisions 118300 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r118300 | tilghman | 2008-05-27 08:13:17 -0500 (Tue, 27 May 2008) | 4 lines In compat14 mode, don't translate pipes inside expressions, as they aren't argument delimiters, but rather 'or' symbols. (Closes issue #12723) ........ 2008-05-25 16:20 +0000 [r118253] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 118252 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118252 | tilghman | 2008-05-25 11:17:05 -0500 (Sun, 25 May 2008) | 20 lines Merged revisions 118251 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118251 | tilghman | 2008-05-25 11:02:04 -0500 (Sun, 25 May 2008) | 12 lines Realtime flag affects construction in multiple ways, so consulting whether rtcachefriends was set was done too soon (needed to be done inside build_peer, not just as a flag to build_peer). Also, fullcontact needed to be reconstructed, because realtime separates the embedded ';' into multiple fields. (closes issue #12722) Reported by: barthpbx Patches: 20080525__bug12722.diff.txt uploaded by Corydon76 (license 14) Tested by: barthpbx (Much of the discussion happened on #asterisk-dev for diagnosing this issue) ........ ................ 2008-05-24 01:15 +0000 [r118177-118179] Jeff Peeler * doc/api-1.6.0-changes.odt (added), /: Merged revisions 118178 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r118178 | jpeeler | 2008-05-23 20:14:41 -0500 (Fri, 23 May 2008) | 1 line add document describing API changes from 1.4.0 to 1.6.0 ........ 2008-05-23 21:37 +0000 [r118168] Brett Bryant * main/manager.c, /, main/http.c, include/asterisk/manager.h: Merged revisions 118161 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r118161 | bbryant | 2008-05-23 16:19:42 -0500 (Fri, 23 May 2008) | 3 lines Add new functionality to http server that requires manager authentication for any path that includes a directory named 'private'. This patch also requires manager authentication for any POST's being sent to the server as well to help secure uploads. ........ 2008-05-23 21:31 +0000 [r118165] Jeff Peeler * channels/chan_zap.c: Merged revisions 118164 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118164 | jpeeler | 2008-05-23 16:26:39 -0500 (Fri, 23 May 2008) | 9 lines Merged revisions 118163 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118163 | jpeeler | 2008-05-23 16:21:35 -0500 (Fri, 23 May 2008) | 1 line Fix a few things I missed to ensure zt_chan_conf structure is not modified in mkintf ........ ................ 2008-05-23 18:15 +0000 [r118130] Tilghman Lesher * res/res_odbc.c, /: Merged revisions 118129 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r118129 | tilghman | 2008-05-23 13:09:14 -0500 (Fri, 23 May 2008) | 3 lines Protect the object from changing while the 'odbc show' CLI command is running (Closes issue #12704) ........ 2008-05-23 13:00 +0000 [r118054] Tilghman Lesher * doc/cli.txt (added), /: Merged revisions 118053 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118053 | tilghman | 2008-05-23 08:00:10 -0500 (Fri, 23 May 2008) | 11 lines Merged revisions 118052 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118052 | tilghman | 2008-05-23 07:59:16 -0500 (Fri, 23 May 2008) | 3 lines Add information on using the Asterisk console, including tab command line completion. (Closes issue #12681) ........ ................ 2008-05-23 12:37 +0000 [r118050] Russell Bryant * include/asterisk/utils.h, /, main/utils.c: Merged revisions 118049 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118049 | russell | 2008-05-23 07:37:31 -0500 (Fri, 23 May 2008) | 17 lines Merged revisions 118048 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118048 | russell | 2008-05-23 07:30:53 -0500 (Fri, 23 May 2008) | 9 lines Don't declare a function that takes variable arguments as inline, because it's not valid, and on some compilers, will emit a warning. http://gcc.gnu.org/onlinedocs/gcc/Inline.html#Inline (closes issue #12289) Reported by: francesco_r Patches by Tilghman, final patch by me ........ ................ 2008-05-23 11:02 +0000 [r118021] Philippe Sultan * /, channels/chan_gtalk.c, res/res_jabber.c: Merged revisions 118020 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r118020 | phsultan | 2008-05-23 12:33:21 +0200 (Fri, 23 May 2008) | 15 lines - remove whitespaces between tags in received XML packets before giving them to the parser ; - report Gtalk error messages from a buddy to the console. This patch makes Asterisk "Google Jingle" (chan_gtalk) implementation work with Empathy. Note that this is only true for audio streams, not video. Thank you to PH for his great help! (closes issue #12647) Reported by: PH Patches: trunk-12647-1.diff uploaded by phsultan (license 73) Tested by: phsultan, PH ........ 2008-05-22 21:43 +0000 [r117984-117987] Tilghman Lesher * /, pbx/pbx_realtime.c, configs/pbx_realtime.conf (added): Merged revisions 117986 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r117986 | tilghman | 2008-05-22 16:42:50 -0500 (Thu, 22 May 2008) | 2 lines Add a compatibility option for upgrading realtime extensions ........ 2008-05-22 18:55 +0000 [r117901] Tilghman Lesher * main/asterisk.c, /: Merged revisions 117900 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r117900 | tilghman | 2008-05-22 13:54:41 -0500 (Thu, 22 May 2008) | 10 lines Merged revisions 117899 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117899 | tilghman | 2008-05-22 13:53:53 -0500 (Thu, 22 May 2008) | 2 lines Also remove preamble from asynchronous events (reported by jsmith on #asterisk-dev) ........ ................ 2008-05-22 15:51 +0000 [r117793] Sean Bright * /, configs/jabber.conf.sample: Merged revisions 117792 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r117792 | seanbright | 2008-05-22 11:49:17 -0400 (Thu, 22 May 2008) | 1 line Minor text fix. roster -> resource. ........ 2008-05-22 13:41 +0000 [r117757] Russell Bryant * main/asterisk.c, /, build_tools/make_buildopts_h: Merged revisions 117756 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r117756 | russell | 2008-05-22 08:40:52 -0500 (Thu, 22 May 2008) | 5 lines Store build-time options as a string in AST_BUILDOPTS in buildopts.h. Also, display this information in the "core show settings" CLI command. This is useful if you want to verify that you're running a build with DONT_OPTIMIZE, DEBUG_THREADS, etc. ........ 2008-05-21 22:01 +0000 [r117659-117660] Jeff Peeler * channels/chan_zap.c: Merged revisions 117658 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r117658 | jpeeler | 2008-05-21 16:31:17 -0500 (Wed, 21 May 2008) | 10 lines Merged revisions 117582 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117582 | jpeeler | 2008-05-21 15:11:14 -0500 (Wed, 21 May 2008) | 2 lines Ensure that passed in zt_chan_conf structure is not modified in mkintf. ........ ................ * channels/chan_zap.c, /: Merged revisions 117628 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r117628 | jpeeler | 2008-05-21 15:44:04 -0500 (Wed, 21 May 2008) | 12 lines Merged revisions 117462 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117462 | jpeeler | 2008-05-21 11:58:40 -0500 (Wed, 21 May 2008) | 3 lines Pass a pointer for the conf parameter to the function mkintf rather than the whole zt_chan_conf structure. Another commit is following to make sure the zt_chan_conf structure is not modified. ........ ................ 2008-05-21 19:45 +0000 [r117576] Joshua Colp * /, channels/chan_sip.c: Merged revisions 117575 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r117575 | file | 2008-05-21 16:39:42 -0300 (Wed, 21 May 2008) | 10 lines Merged revisions 117574 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117574 | file | 2008-05-21 16:38:28 -0300 (Wed, 21 May 2008) | 2 lines Apply the autoframing setting to dialogs that do not get matched against a user or peer. ........ ................ 2008-05-21 18:44 +0000 [r117522] Tilghman Lesher * main/asterisk.c, /: Merged revisions 117520 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r117520 | tilghman | 2008-05-21 13:43:26 -0500 (Wed, 21 May 2008) | 11 lines Merged revisions 117519 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117519 | tilghman | 2008-05-21 13:40:14 -0500 (Wed, 21 May 2008) | 3 lines Strip the preamble from the output also when -rx is not being used (Related to issue #12702) ........ ................ 2008-05-21 18:29 +0000 [r117486-117516] Russell Bryant * main/asterisk.c, /: Merged revisions 117515 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r117515 | russell | 2008-05-21 13:29:05 -0500 (Wed, 21 May 2008) | 12 lines Merged revisions 117514 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117514 | russell | 2008-05-21 13:28:46 -0500 (Wed, 21 May 2008) | 4 lines Don't filter the magic character in the network verboser. It gets filtered once it reaches the client. (related to issue #12702, pointed out by tilghman) ........ ................ * main/asterisk.c, pbx/pbx_gtkconsole.c, /: Merged revisions 117508 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r117508 | russell | 2008-05-21 13:20:11 -0500 (Wed, 21 May 2008) | 15 lines Merged revisions 117507 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117507 | russell | 2008-05-21 13:19:34 -0500 (Wed, 21 May 2008) | 7 lines 1) Don't print the verbose marker in front of every message from ast_verbose() being sent to remote consoles. 2) Fix pbx_gtkconsole to filter out the verbose marker. (related to issue #12702) ........ ................ * main/asterisk.c, /: Merged revisions 117481 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r117481 | russell | 2008-05-21 13:12:19 -0500 (Wed, 21 May 2008) | 14 lines Merged revisions 117479 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117479 | russell | 2008-05-21 13:11:51 -0500 (Wed, 21 May 2008) | 6 lines Don't display the verbose marker for calls to ast_verbose() that do not include a VERBOSE_PREFIX in front of the message. (closes issue #12702) Reported by: johnlange Patched by me ........ ................ 2008-05-21 02:21 +0000 [r117368] Mark Michelson * main/config.c, /: Merged revisions 117367 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r117367 | mmichelson | 2008-05-20 21:20:31 -0500 (Tue, 20 May 2008) | 19 lines Be sure that we cache included files for each source file which loads a configuration file. As it was, only the first did so. This led to a problem if the included file was changed (but not the configuration file which includes it) and the second source file attempted to reload the configuration. It would not see that the included file had changed. In this particular example, res_phoneprov and chan_sip both loaded sip.conf, which included a file call sip.peers.conf. Since res_phoneprov was the first to load sip.conf, only it cached the fact that sip.conf included sip.peers.conf. If sip.peers.conf were changed and sip.conf were not and a sip reload were issued (meaning that chan_sip attempts to reload sip.conf only if it and its included files have changed) the changes made to sip.peers.conf would not be seen and therefore no action would be taken. (closes issue #12693) Reported by: marsosa ........ 2008-05-21 01:20 +0000 [r117365] Steve Murphy * /, utils/ael_main.c: Merged revisions 117335 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r117335 | murf | 2008-05-20 19:00:28 -0600 (Tue, 20 May 2008) | 10 lines These changes were made via the comments atis_work made at 4:30am (Mountain Time zone- US) in #asterisk-dev on 20 May 2008. He noted that a backslash was being inserted before commas in app call arguments in the extensions.conf.aeldump file that you get from aelparse with the -w arg. This was being generated from code left over from 1.4, where commas were substituted with '|', and any remaining commas needed to be escaped. Many thanks to atis for his comment; please let us know if these changes break anything! ........ 2008-05-19 16:58 +0000 [r117134-117137] Joshua Colp * res/res_smdi.c, /: Merged revisions 117136 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r117136 | file | 2008-05-19 13:53:33 -0300 (Mon, 19 May 2008) | 14 lines Merged revisions 117135 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117135 | file | 2008-05-19 13:50:52 -0300 (Mon, 19 May 2008) | 6 lines Use the right pthread lock and condition when waiting. (closes issue #12664) Reported by: tomo1657 Patches: res_smdi.c.patch uploaded by tomo1657 (license 484) ........ ................ 2008-05-19 16:07 +0000 [r117089] Tilghman Lesher * include/asterisk/utils.h, /: Merged revisions 117088 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r117088 | tilghman | 2008-05-19 11:07:09 -0500 (Mon, 19 May 2008) | 10 lines Merged revisions 117086 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117086 | tilghman | 2008-05-19 11:05:05 -0500 (Mon, 19 May 2008) | 2 lines The addition of usleep(2) within ast_assert requires the inclusion of the unistd.h header ........ ................ 2008-05-19 16:05 +0000 [r117083-117087] Joshua Colp * /, main/logger.c: Merged revisions 117085 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r117085 | file | 2008-05-19 13:03:33 -0300 (Mon, 19 May 2008) | 4 lines The logger closes the files it is logging to when reloading so we have to read in the logger configuration even if it has not changed so that the logs get opened again. (closes issue #12665) Reported by: DennisD ........ * /, channels/h323/ast_h323.cxx: Merged revisions 117082 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r117082 | file | 2008-05-19 12:24:44 -0300 (Mon, 19 May 2008) | 14 lines Merged revisions 117081 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117081 | file | 2008-05-19 12:22:10 -0300 (Mon, 19 May 2008) | 6 lines Make chan_h323 work with pwlib 1.12.0 (closes issue #12682) Reported by: bamby Patches: pwlib_nopipe.diff uploaded by bamby (license 430) ........ ................ 2008-05-19 03:44 +0000 [r116980] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 116979 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r116979 | russell | 2008-05-18 22:44:28 -0500 (Sun, 18 May 2008) | 12 lines Merged revisions 116978 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116978 | russell | 2008-05-18 22:44:04 -0500 (Sun, 18 May 2008) | 4 lines Avoid access of uninitialized memory. This caused a bunch of crashes for me while doing load testing of development branch where I'm working on some performance improvements. ........ ................ 2008-05-18 21:18 +0000 [r116949] Tilghman Lesher * /, utils/astcanary.c: Merged revisions 116948 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r116948 | tilghman | 2008-05-18 16:15:58 -0500 (Sun, 18 May 2008) | 4 lines Add a set of text to the file astcanary uses to communicate back the main Asterisk process, which explains the purpose for the file being there. This should assist people who find the file and wonder why it exists. ........ 2008-05-18 19:59 +0000 [r116922] Russell Bryant * /, channels/chan_sip.c: Merged revisions 116919 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r116919 | russell | 2008-05-18 14:58:10 -0500 (Sun, 18 May 2008) | 3 lines Remove duplicate colon on Reason header (closes issue #12678) ........ 2008-05-17 19:40 +0000 [r116849-116885] Joshua Colp * /, channels/chan_skinny.c: Merged revisions 116800 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r116800 | file | 2008-05-16 17:30:24 -0300 (Fri, 16 May 2008) | 12 lines Merged revisions 116799 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116799 | file | 2008-05-16 17:28:11 -0300 (Fri, 16 May 2008) | 4 lines Check to make sure an RTP structure exists before calling ast_rtp_new_source on it. (closes issue #12669) Reported by: sbisker ........ ................ 2008-05-16 20:03 +0000 [r116798] Matthew Fredrickson * channels/chan_zap.c, /: Merged revisions 116797 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r116797 | mattf | 2008-05-16 15:00:04 -0500 (Fri, 16 May 2008) | 1 line Try to see if we can make our ringback situation a little better ........ 2008-05-15 22:07 +0000 [r116636-116695] Tilghman Lesher * include/asterisk/utils.h, /, include/asterisk/strings.h: Merged revisions 116694 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r116694 | tilghman | 2008-05-15 17:05:47 -0500 (Thu, 15 May 2008) | 4 lines Add an extra check in ast_strlen_zero, and make ast_assert() not print the file, line, and function name twice. (Closes issue #12650) ........ * cdr/cdr_csv.c, /: Merged revisions 116631 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r116631 | tilghman | 2008-05-15 12:58:22 -0500 (Thu, 15 May 2008) | 3 lines Don't unload config on reload, when config has not changed. (Closes issue #12652) ........ 2008-05-14 21:41 +0000 [r116470] Russell Bryant * main/rtp.c, main/sched.c, main/channel.c, main/udptl.c, include/asterisk/utils.h, /, channels/chan_agent.c, main/abstract_jb.c, include/asterisk/channel.h: Merged revisions 116469 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r116469 | russell | 2008-05-14 16:40:43 -0500 (Wed, 14 May 2008) | 12 lines Merged revisions 116463 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116463 | russell | 2008-05-14 16:32:00 -0500 (Wed, 14 May 2008) | 4 lines Add ast_assert(), which can be used to handle fatal errors. It is only compiled in if dev-mode is enabled, and only aborts if DO_CRASH is defined. (inspired by issue #12650) ........ ................ 2008-05-14 21:39 +0000 [r116468] Tilghman Lesher * /, res/res_agi.c: Merged revisions 116467 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r116467 | tilghman | 2008-05-14 16:39:06 -0500 (Wed, 14 May 2008) | 15 lines Merged revisions 116466 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116466 | tilghman | 2008-05-14 16:38:09 -0500 (Wed, 14 May 2008) | 7 lines Avoid zombies when the channel exits before the AGI. (closes issue #12648) Reported by: gkloepfer Patches: 20080514__bug12648.diff.txt uploaded by Corydon76 (license 14) Tested by: gkloepfer ........ ................ 2008-05-14 20:43 +0000 [r116408-116411] Jason Parker * /, configs/voicemail.conf.sample: Merged revisions 116410 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r116410 | qwell | 2008-05-14 15:43:26 -0500 (Wed, 14 May 2008) | 9 lines Merged revisions 116409 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116409 | qwell | 2008-05-14 15:43:08 -0500 (Wed, 14 May 2008) | 1 line Document exitcontext in app_voicemail sample config ........ ................ * apps/app_voicemail.c, /: Merged revisions 116407 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r116407 | qwell | 2008-05-14 15:36:55 -0500 (Wed, 14 May 2008) | 9 lines Voicemail "* exit" should not require an exitcontext to be specified. The behavior in 1.4 was that it would use the current context if an exitcontext existed. (closes issue #12605) Reported by: kenjreno Patches: 12605-starexit.diff uploaded by qwell (license 4) Tested by: file ........ 2008-05-14 18:54 +0000 [r116351-116354] Joshua Colp * /, main/Makefile: Merged revisions 116353 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r116353 | file | 2008-05-14 15:54:16 -0300 (Wed, 14 May 2008) | 12 lines Merged revisions 116352 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116352 | file | 2008-05-14 15:53:39 -0300 (Wed, 14 May 2008) | 4 lines Add linux-gnueabi in. (closes issue #12529) Reported by: tzafrir ........ ................ * /, res/res_config_ldap.c: Merged revisions 116350 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r116350 | file | 2008-05-14 15:25:54 -0300 (Wed, 14 May 2008) | 4 lines Make the ldap version setting work without having both version and protocol set. (closes issue #12613) Reported by: suretec ........ 2008-05-14 17:01 +0000 [r116319] Tilghman Lesher * /, apps/app_externalivr.c: Merged revisions 116298 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r116298 | tilghman | 2008-05-14 11:53:23 -0500 (Wed, 14 May 2008) | 15 lines Merged revisions 116296 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116296 | tilghman | 2008-05-14 11:46:48 -0500 (Wed, 14 May 2008) | 2 lines Detect another way for a connection to have gone away. (closes issue #12618) Reported by: ctooley Patches: 1.4-externalivr-test_fd.diff uploaded by ctooley (license 136) trunk-externalivr-test_fd.diff uploaded by ctooley (license 136) ........ ................ 2008-05-14 Russell Bryant * Asterisk 1.6.0-beta9 released. 2008-05-14 13:13 +0000 [r116236] Olle Johansson * /, channels/chan_sip.c: Merged revisions 116234 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r116234 | oej | 2008-05-14 15:05:15 +0200 (Ons, 14 Maj 2008) | 11 lines Merged revisions 116230 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116230 | oej | 2008-05-14 14:51:06 +0200 (Ons, 14 Maj 2008) | 3 lines Accept text messages even with Content-Type: text/plain;charset=Södermanländska ........ ................ 2008-05-14 00:20 +0000 [r116096-116139] Mark Michelson * main/channel.c, /, include/asterisk/lock.h: Merged revisions 116089 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r116089 | mmichelson | 2008-05-13 18:54:01 -0500 (Tue, 13 May 2008) | 20 lines Merged revisions 116088 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116088 | mmichelson | 2008-05-13 18:47:49 -0500 (Tue, 13 May 2008) | 12 lines A change to the way channel locks are handled when DEBUG_CHANNEL_LOCKS is defined. After debugging a deadlock, it was noticed that when DEBUG_CHANNEL_LOCKS is enabled in menuselect, the actual origin of channel locks is obscured by the fact that all channel locks appear to happen in the function ast_channel_lock(). This code change redefines ast_channel_lock to be a macro which maps to __ast_channel_lock(), which then relays the proper file name, line number, and function name information to the core lock functions so that this information will be displayed in the case that there is some sort of locking error or core show locks is issued. ........ ................ 2008-05-13 21:19 +0000 [r116020-116040] Russell Bryant * channels/chan_local.c, /: Merged revisions 116039 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r116039 | russell | 2008-05-13 16:18:55 -0500 (Tue, 13 May 2008) | 32 lines Merged revisions 116038 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116038 | russell | 2008-05-13 16:17:23 -0500 (Tue, 13 May 2008) | 24 lines Fix a deadlock involving channel autoservice and chan_local that was debugged and fixed by mmichelson and me. We observed a system that had a bunch of threads stuck in ast_autoservice_stop(). The reason these threads were waiting around is because this function waits to ensure that the channel list in the autoservice thread gets rebuilt before the stop() function returns. However, the autoservice thread was also locked, so the autoservice channel list was never getting rebuilt. The autoservice thread was stuck waiting for the channel lock on a local channel. However, the local channel was locked by a thread that was stuck in the autoservice stop function. It turned out that the issue came down to the local_queue_frame() function in chan_local. This function assumed that one of the channels passed in as an argument was locked when called. However, that was not always the case. There were multiple cases in which this channel was not locked when the function was called. We fixed up chan_local to indicate to this function whether this channel was locked or not. The previous assumption had caused local_queue_frame() to improperly return with the channel locked, where it would then never get unlocked. (closes issue #12584) (related to issue #12603) ........ ................ * main/autoservice.c, /: Merged revisions 116001 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r116001 | russell | 2008-05-13 16:07:59 -0500 (Tue, 13 May 2008) | 13 lines Merged revisions 115990 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115990 | russell | 2008-05-13 16:05:57 -0500 (Tue, 13 May 2008) | 5 lines Fix an issue that I noticed in autoservice while mmichelson and I were debugging a different problem. I noticed that it was theoretically possible for two threads to attempt to start the autoservice thread at the same time. This change makes the process of starting the autoservice thread, thread-safe. ........ ................ 2008-05-13 20:30 +0000 [r115946] Joshua Colp * /, channels/chan_alsa.c: Merged revisions 115945 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115945 | file | 2008-05-13 17:29:27 -0300 (Tue, 13 May 2008) | 12 lines Merged revisions 115944 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115944 | file | 2008-05-13 17:28:23 -0300 (Tue, 13 May 2008) | 4 lines Use the right flag to open the audio in non-blocking. (closes issue #12616) Reported by: nicklewisdigiumuser ........ ................ 2008-05-13 20:19 +0000 [r115940-115942] Matthew Fredrickson * channels/chan_zap.c, /: Merged revisions 115941 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115941 | mattf | 2008-05-13 15:18:04 -0500 (Tue, 13 May 2008) | 1 line Need to clear calling_party_cat variable after we retrieve it ........ * channels/chan_zap.c, /: Merged revisions 115939 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115939 | mattf | 2008-05-13 15:11:20 -0500 (Tue, 13 May 2008) | 1 line Add support for receiving calling party category ........ 2008-05-13 18:38 +0000 [r115887] Tilghman Lesher * main/asterisk.c, /: Merged revisions 115886 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115886 | tilghman | 2008-05-13 13:38:11 -0500 (Tue, 13 May 2008) | 11 lines Merged revisions 115884 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115884 | tilghman | 2008-05-13 13:36:13 -0500 (Tue, 13 May 2008) | 3 lines If the socket dies (read returns 0=EOF), return immediately. (Closes issue #12637) ........ ................ 2008-05-13 17:48 +0000 [r115848-115851] Russell Bryant * res/res_smdi.c, /: Merged revisions 115847 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115847 | russell | 2008-05-13 12:14:22 -0500 (Tue, 13 May 2008) | 2 lines Initialize the start time in smdi_msg_wait. Somehow this code got lost in trunk. ........ 2008-05-12 17:57 +0000 [r115738] Mark Michelson * main/utils.c: Merged revisions 115737 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115737 | mmichelson | 2008-05-12 12:55:08 -0500 (Mon, 12 May 2008) | 15 lines Merged revisions 115735 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115735 | mmichelson | 2008-05-12 12:51:14 -0500 (Mon, 12 May 2008) | 7 lines If a thread holds no locks, do not print any information on the thread when issuing a core show locks command. This will help to de-clutter output somewhat. Russell said it would be fine to place this improvement in the 1.4 branch, so that's why it's going here too. ........ ................ 2008-05-12 16:36 +0000 [r115706] Jason Parker * /, apps/app_queue.c: Merged revisions 115705 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115705 | qwell | 2008-05-12 11:35:50 -0500 (Mon, 12 May 2008) | 1 line Correctly document state interface for AddQueueMember. Discovered while looking at issue #12626. ........ 2008-05-12 15:18 +0000 [r115672] Brett Bryant * /, channels/chan_iax2.c: Merged revisions 115669 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115669 | bbryant | 2008-05-12 10:17:32 -0500 (Mon, 12 May 2008) | 3 lines A small change to fix iax2 native bridging. ........ 2008-05-11 03:27 +0000 [r115599-115601] Matthew Fredrickson * channels/chan_zap.c, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 115600 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115600 | mattf | 2008-05-10 22:23:05 -0500 (Sat, 10 May 2008) | 1 line Add Zap MTP2 support to chan_zap ........ * channels/chan_zap.c, /: Merged revisions 115598 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115598 | mattf | 2008-05-10 21:19:21 -0500 (Sat, 10 May 2008) | 1 line Open up audio channel when we get ACM on SS7 event ........ 2008-05-10 14:22 +0000 [r115597] Tilghman Lesher * /, cdr/cdr_pgsql.c: Merged revisions 115596 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115596 | tilghman | 2008-05-10 09:19:41 -0500 (Sat, 10 May 2008) | 2 lines Ensure that "calldate" is acceptable for a column name. ........ 2008-05-09 16:38 +0000 [r115581] Joshua Colp * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 115580 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115580 | file | 2008-05-09 13:36:58 -0300 (Fri, 09 May 2008) | 10 lines Merged revisions 115579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115579 | file | 2008-05-09 13:34:08 -0300 (Fri, 09 May 2008) | 2 lines Improve res_ninit and res_ndestroy autoconf logic on the Darwin platform. ........ ................ 2008-05-08 19:21 +0000 [r115553-115570] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 115569 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115569 | russell | 2008-05-08 14:20:35 -0500 (Thu, 08 May 2008) | 10 lines Merged revisions 115568 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115568 | russell | 2008-05-08 14:19:50 -0500 (Thu, 08 May 2008) | 2 lines Remove debug output. ........ ................ * /, channels/chan_iax2.c: Merged revisions 115566 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115566 | russell | 2008-05-08 14:17:04 -0500 (Thu, 08 May 2008) | 41 lines Merged revisions 115565 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r115565 | russell | 2008-05-08 14:15:25 -0500 (Thu, 08 May 2008) | 33 lines Merged revisions 115564 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r115564 | russell | 2008-05-08 14:14:04 -0500 (Thu, 08 May 2008) | 25 lines Fix a race condition that bbryant just found while doing some IAX2 testing. He was running Asterisk trunk running IAX2 calls through a few Asterisk boxes, however, the audio was extremely choppy. We looked at a packet trace and saw a storm of INVAL and VNAK frames being sent from one box to another. It turned out that what had happened was that one box tried to send a CONTROL frame before the 3 way handshake had completed. So, that frame did not include the destination call number, because it didn't have it yet. Part of our recent work for security issues included an additional check to ensure that frames that are supposed to include the destination call number have the correct one. This caused the frame to be rejected with an INVAL. The frame would get retransmitted for forever, rejected every time ... This race condition exists in all versions that got the security changes, in theory. However, it is really only likely that this would cause a problem in Asterisk trunk. There was a control frame being sent (SRCUPDATE) at the _very_ beginning of the call, which does not exist in 1.2 or 1.4. However, I am fixing all versions that could potentially be affected by the introduced race condition. These changes are what bbryant and I came up with to fix the issue. Instead of simply dropping control frames that get sent before the handshake is complete, the code attempts to wait a little while, since in most cases, the handshake will complete very quickly. If it doesn't complete after yielding for a little while, then the frame gets dropped. ........ ................ ................ * /, channels/chan_sip.c: Merged revisions 115562 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115562 | russell | 2008-05-08 11:14:08 -0500 (Thu, 08 May 2008) | 11 lines Merged revisions 115561 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115561 | russell | 2008-05-08 11:11:33 -0500 (Thu, 08 May 2008) | 3 lines Don't give up on attempting an outbound registration if we receive a 408 Timeout. (closes issue #12323) ........ ................ * /, contrib/scripts/postgres_cdr.sql (removed): Merged revisions 115558 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115558 | russell | 2008-05-08 10:38:27 -0500 (Thu, 08 May 2008) | 11 lines Merged revisions 115557 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115557 | russell | 2008-05-08 10:37:49 -0500 (Thu, 08 May 2008) | 3 lines remove postgres_cdr.sql, as the CDR schema is in realtime_pgsql.sql, as well (closes issue #9676) ........ ................ * contrib/init.d/rc.debian.asterisk, /: Merged revisions 115555 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115555 | russell | 2008-05-08 10:32:48 -0500 (Thu, 08 May 2008) | 11 lines Merged revisions 115554 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115554 | russell | 2008-05-08 10:32:08 -0500 (Thu, 08 May 2008) | 3 lines Don't exit the script if Asterisk is not running. (closes issue #12611) ........ ................ * main/pbx.c, /: Merged revisions 115552 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115552 | russell | 2008-05-08 10:26:49 -0500 (Thu, 08 May 2008) | 12 lines Merged revisions 115551 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115551 | russell | 2008-05-08 10:24:54 -0500 (Thu, 08 May 2008) | 4 lines Don't use a channel before checking for channel allocation failure. (closes issue #12609) Reported by: edantie ........ ................ 2008-05-08 15:08 +0000 [r115549] Matthew Fredrickson * channels/chan_zap.c, /: Merged revisions 115548 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115548 | mattf | 2008-05-08 10:04:45 -0500 (Thu, 08 May 2008) | 1 line Remove unused code as well as demote an error message to a debug message ........ 2008-05-08 14:41 +0000 [r115538-115547] Russell Bryant * contrib/init.d/rc.debian.asterisk, /: Merged revisions 115546 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115546 | russell | 2008-05-08 09:41:12 -0500 (Thu, 08 May 2008) | 12 lines Merged revisions 115545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115545 | russell | 2008-05-08 09:40:53 -0500 (Thu, 08 May 2008) | 4 lines Use the same method for executing Asterisk as the rest of the script. (closes issue #12611) Reported by: b_plessis ........ ................ 2008-05-07 18:35 +0000 [r115514-115524] Russell Bryant * /, res/res_config_ldap.c: Merged revisions 115523 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115523 | russell | 2008-05-07 13:33:50 -0500 (Wed, 07 May 2008) | 6 lines Only save a password if a username exists. (closes issue #12600) Reported By: suretec Patch by me ........ * /, res/res_config_ldap.c: Merged revisions 115521 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115521 | russell | 2008-05-07 13:30:12 -0500 (Wed, 07 May 2008) | 7 lines Use the default that the log output claims will be used for the basedn (closes issue #12599) Reported by: suretec Patches: 12599.patch uploaded by juggie (license 24) ........ * /, channels/chan_h323.c: Merged revisions 115519 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115519 | russell | 2008-05-07 13:24:51 -0500 (Wed, 07 May 2008) | 2 lines Let chan_h323 build in dev mode ........ * /, include/asterisk/dlinkedlists.h (removed), channels/chan_iax2.c: Merged revisions 115513 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115513 | russell | 2008-05-07 12:28:19 -0500 (Wed, 07 May 2008) | 19 lines Merged revisions 115512 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r115512 | russell | 2008-05-07 11:24:09 -0500 (Wed, 07 May 2008) | 11 lines Merged revisions 115511 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r115511 | russell | 2008-05-07 11:22:49 -0500 (Wed, 07 May 2008) | 3 lines Remove remnants of dlinkedlists. I didn't actually use them in the final version of my IAX2 improvements. ........ ................ ................ 2008-05-07 13:49 +0000 [r115510] Tilghman Lesher * contrib/scripts/asterisk.ldap-schema, contrib/scripts/asterisk.ldif, /: Merged revisions 115509 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115509 | tilghman | 2008-05-07 08:49:15 -0500 (Wed, 07 May 2008) | 2 lines Update typos in description fields (closes issue #12598) Reported by: suretec Patches: asterisk_schema_changes.patch uploaded by suretec (license 70) ........ 2008-05-06 19:56 +0000 [r115420-115424] Jason Parker * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 115423 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115423 | qwell | 2008-05-06 14:55:45 -0500 (Tue, 06 May 2008) | 23 lines Merged revisions 115422 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r115422 | qwell | 2008-05-06 14:55:29 -0500 (Tue, 06 May 2008) | 15 lines Merged revisions 115421 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r115421 | qwell | 2008-05-06 14:54:57 -0500 (Tue, 06 May 2008) | 7 lines read requires an argument on some non-bash shells (closes issue #12593) Reported by: bkruse Patches: getilbc.sh_12593_v1.diff uploaded by bkruse (license 132) ........ ................ ................ * /, res/res_musiconhold.c: Merged revisions 115419 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115419 | qwell | 2008-05-06 14:38:44 -0500 (Tue, 06 May 2008) | 15 lines Merged revisions 115418 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115418 | qwell | 2008-05-06 14:34:58 -0500 (Tue, 06 May 2008) | 7 lines Switch to using ast_random() rather than just rand(). This does not fix the bug reported, but I believe it is correct. (from issue #12446) Patches: bug_12446.diff uploaded by snuffy (license 35) ........ ................ 2008-05-06 19:33 +0000 [r115417] Tilghman Lesher * main/asterisk.c, /: Merged revisions 115416 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115416 | tilghman | 2008-05-06 14:32:29 -0500 (Tue, 06 May 2008) | 10 lines Merged revisions 115415 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115415 | tilghman | 2008-05-06 14:31:39 -0500 (Tue, 06 May 2008) | 2 lines Don't print the terminating NUL. (Closes issue #12589) ........ ................ 2008-05-06 13:57 +0000 [r115343] Joshua Colp * /, configure, configure.ac: Merged revisions 115342 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115342 | file | 2008-05-06 10:55:44 -0300 (Tue, 06 May 2008) | 10 lines Merged revisions 115341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115341 | file | 2008-05-06 10:54:15 -0300 (Tue, 06 May 2008) | 2 lines Add in missing argument. ........ ................ 2008-05-05 23:01 +0000 [r115335] Tilghman Lesher * main/asterisk.c, /, main/logger.c: Merged revisions 115334 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115334 | tilghman | 2008-05-05 18:00:31 -0500 (Mon, 05 May 2008) | 15 lines Merged revisions 115333 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115333 | tilghman | 2008-05-05 17:50:31 -0500 (Mon, 05 May 2008) | 7 lines Separate verbose output from CLI output, by using a preamble. (closes issue #12402) Reported by: Corydon76 Patches: 20080410__no_verbose_in_rx_output.diff.txt uploaded by Corydon76 (license 14) 20080501__no_verbose_in_rx_output__1.4.diff.txt uploaded by Corydon76 (license 14) ........ ................ 2008-05-05 22:17 +0000 [r115331] Joshua Colp * /, build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, codecs/codec_speex.c, configure.ac: Merged revisions 115328 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115328 | file | 2008-05-05 19:13:57 -0300 (Mon, 05 May 2008) | 10 lines Merged revisions 115327 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115327 | file | 2008-05-05 19:10:05 -0300 (Mon, 05 May 2008) | 2 lines Make sure that either the main speex library contains preprocess functions or that speexdsp does. If both fail then speex stuff can not be built. ........ ................ 2008-05-05 22:14 +0000 [r115330] Mark Michelson * main/config.c, /: Merged revisions 115329 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115329 | mmichelson | 2008-05-05 17:14:06 -0500 (Mon, 05 May 2008) | 15 lines #execing the same file multiple times led to warning messages saying that the same file was being #included twice. This was due to the fact that #exec created a temporary file which was then #included. The name of the temporary file was the name of the #exec'd file, with the Unix timestamp and thread ID concatenated. The issue was that if multiple #exec statements of the same file were reached in the same second, then the result was that the temporary files would have duplicate names. To resolve this, the temporary file now has microsecond resolution for the timestamp portion. (closes issue #12574) Reported by: jmls Patches: 12574.patch uploaded by putnopvut (license 60) Tested by: jmls, putnopvut ........ 2008-05-05 21:44 +0000 [r115322] Mark Michelson * /, apps/app_queue.c: Merged revisions 115321 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115321 | mmichelson | 2008-05-05 16:43:21 -0500 (Mon, 05 May 2008) | 21 lines Merged revisions 115320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115320 | mmichelson | 2008-05-05 16:41:34 -0500 (Mon, 05 May 2008) | 13 lines Don't consider a caller "handled" until the caller is bridged with a queue member. There was too much of an opportunity for the member to hang up (either during a delay, announcement, or overly long agi) between the time that he answered the phone and the time when he actually was bridged with the caller. The consequence of this was that if the member hung up in that interval, then proper abandonment details would not be noted in the queue log if the caller were to hang up at any point after the member hangup. (closes issue #12561) Reported by: ablackthorn ........ ................ 2008-05-05 20:28 +0000 [r115316] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 115315 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115315 | russell | 2008-05-05 15:28:17 -0500 (Mon, 05 May 2008) | 2 lines Remove my rant, since I have now replaced the rant with code. ........ 2008-05-05 19:58 +0000 [r115310] Tilghman Lesher * include/asterisk/res_odbc.h, /: Merged revisions 115309 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115309 | tilghman | 2008-05-05 14:57:28 -0500 (Mon, 05 May 2008) | 10 lines Merged revisions 115308 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115308 | tilghman | 2008-05-05 14:55:55 -0500 (Mon, 05 May 2008) | 2 lines Err, the documentation on the return value of ast_odbc_backslash_is_escape is exactly backwards. ........ ................ 2008-05-05 19:50 +0000 [r115306] Russell Bryant * /, channels/chan_sip.c: Merged revisions 115305 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115305 | russell | 2008-05-05 14:50:24 -0500 (Mon, 05 May 2008) | 13 lines Merged revisions 115304 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115304 | russell | 2008-05-05 14:49:25 -0500 (Mon, 05 May 2008) | 5 lines Avoid putting opaque="" in Digest authentication. This patch came from switchvox. It fixes authentication with Primus in Canada, and has been in use for a very long time without causing problems with any other providers. (closes issue AST-36) ........ ................ 2008-05-05 19:43 +0000 [r115303] Tilghman Lesher * /, UPGRADE.txt: Merged revisions 115302 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115302 | tilghman | 2008-05-05 14:42:36 -0500 (Mon, 05 May 2008) | 2 lines Note change for ExecIf syntax (caught by jmls on IRC) ........ 2008-05-05 10:55 +0000 [r115289] Kevin P. Fleming * /, UPGRADE.txt: Merged revisions 115288 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115288 | kpfleming | 2008-05-05 05:55:09 -0500 (Mon, 05 May 2008) | 2 lines clarify wording ........ 2008-05-05 03:26 +0000 [r115287] Tilghman Lesher * contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk, contrib/init.d/rc.debian.asterisk, contrib/init.d/rc.mandrake.asterisk, /, contrib/init.d/rc.redhat.asterisk, contrib/init.d/rc.gentoo.asterisk, contrib/init.d/rc.slackware.asterisk: Merged revisions 115286 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115286 | tilghman | 2008-05-04 22:25:35 -0500 (Sun, 04 May 2008) | 15 lines Merged revisions 115285 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115285 | tilghman | 2008-05-04 22:22:25 -0500 (Sun, 04 May 2008) | 7 lines When starting Asterisk, bug out if Asterisk is already running. (closes issue #12525) Reported by: explidous Patches: 20080428__bug12525.diff.txt uploaded by Corydon76 (license 14) Tested by: mvanbaak ........ ................ 2008-05-04 02:12 +0000 [r115278-115284] Joshua Colp * /, configure, acinclude.m4: Merged revisions 115283 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115283 | file | 2008-05-03 23:11:01 -0300 (Sat, 03 May 2008) | 10 lines Merged revisions 115282 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115282 | file | 2008-05-03 23:09:44 -0300 (Sat, 03 May 2008) | 2 lines Expand the test function for GCC attributes so that more complex attributes are properly recognized. ........ ................ * /, include/asterisk/compiler.h: Merged revisions 115280 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115280 | file | 2008-05-03 22:52:00 -0300 (Sat, 03 May 2008) | 10 lines Merged revisions 115279 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115279 | file | 2008-05-03 22:50:59 -0300 (Sat, 03 May 2008) | 2 lines For my next trick I will make these work with what our autoconf header file gives us. ........ ................ * /, configure, acinclude.m4: Merged revisions 115277 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115277 | file | 2008-05-03 22:45:21 -0300 (Sat, 03 May 2008) | 10 lines Merged revisions 115276 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115276 | file | 2008-05-03 22:43:26 -0300 (Sat, 03 May 2008) | 2 lines Treat warnings as errors when checking if a GCC attribute exists. We have to do this as GCC will just ignore the attribute and pop up a warning, it won't actually fail to compile. ........ ................ 2008-05-03 04:25 +0000 [r115269-115275] Dwayne M. Hubbard * /: block voicemail mwi notification subscriptions taskprocessor * /: block pbx taskprocessor * /: block app_queue taskprocessor * /: blocked taskprocessors 2008-05-02 14:55 +0000 [r115198-115200] Mark Michelson * /, include/asterisk/sched.h: Merged revisions 115197 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115197 | mmichelson | 2008-05-02 09:28:55 -0500 (Fri, 02 May 2008) | 14 lines Merged revisions 115196 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115196 | mmichelson | 2008-05-02 09:28:19 -0500 (Fri, 02 May 2008) | 6 lines Clarify a comment that was, well, just wrong. It turns out that ignoring the way that macros expand. Instead, I have clarified in the comment why the macro will work even if the scheduler id for the task to be deleted changes during the execution of the macro. ........ ................ 2008-05-02 02:57 +0000 [r115107-115160] Tilghman Lesher * include/asterisk/res_odbc.h, /: Merged revisions 115104 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115104 | tilghman | 2008-05-01 18:21:13 -0500 (Thu, 01 May 2008) | 10 lines Merged revisions 115102 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115102 | tilghman | 2008-05-01 18:20:25 -0500 (Thu, 01 May 2008) | 2 lines Change the comment of deprecated to an actual compiler deprecation ........ ................ 2008-05-01 19:01 +0000 [r115020] Tilghman Lesher * /, main/utils.c: Merged revisions 115018 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115018 | tilghman | 2008-05-01 14:00:18 -0500 (Thu, 01 May 2008) | 14 lines Merged revisions 115017 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115017 | tilghman | 2008-05-01 13:59:08 -0500 (Thu, 01 May 2008) | 6 lines '#' is another reserved character for URIs that also needs to be escaped. (closes issue #10543) Reported by: blitzrage Patches: 20080418__bug10543.diff.txt uploaded by Corydon76 (license 14) ........ ................ 2008-05-01 17:28 +0000 [r114932] Russell Bryant * /, UPGRADE.txt: Merged revisions 114931 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114931 | russell | 2008-05-01 12:28:25 -0500 (Thu, 01 May 2008) | 4 lines Clarify the deprecation notice about Macro() to note that it will not be removed for the sake of backwards compatibility, since it is a non-trivial task to convert existing large dialplans that depend on Macro() to use GoSub(), instead. ........ 2008-05-01 16:52 +0000 [r114923] Jason Parker * channels/chan_zap.c, /: Merged revisions 114922 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114922 | qwell | 2008-05-01 11:49:24 -0500 (Thu, 01 May 2008) | 10 lines Allow dringXrange to properly default to 10, as was done in 1.4. dringXrange is a new feature that was added, and it attempted to default, but only when the option was specified. (closes issue #12536) Reported by: bjm Patches: 12536-dringXrange.diff uploaded by qwell (license 4) Tested by: bjm ........ 2008-04-30 20:20 +0000 [r114909] Russell Bryant * include/asterisk/dlinkedlists.h (added): Add the dlinkedlists implementation from trunk 2008-04-30 20:17 +0000 [r114907-114908] Mark Michelson * channels/chan_sip.c: Make 1.6.0 compile 2008-04-30 17:06 +0000 [r114900] Olle Johansson * /, channels/chan_sip.c: Merged revisions 114899 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114899 | oej | 2008-04-30 18:55:49 +0200 (Ons, 30 Apr 2008) | 15 lines Merged revisions 114890 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114890 | oej | 2008-04-30 18:23:17 +0200 (Ons, 30 Apr 2008) | 7 lines Don't crash on bad SIP replys. Fix created in Huntsville together with Mark M (putnopvut) (closes issue #12363) Reported by: jvandal Tested by: putnopvut, oej ........ ................ 2008-04-30 16:41 +0000 [r114893] Russell Bryant * /, channels/chan_console.c, channels/chan_iax2.c: Merged revisions 114892 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114892 | russell | 2008-04-30 11:34:24 -0500 (Wed, 30 Apr 2008) | 36 lines Merged revisions 114891 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114891 | russell | 2008-04-30 11:30:01 -0500 (Wed, 30 Apr 2008) | 28 lines Merge changes from team/russell/iax2_find_callno and iax2_find_callno_1.4 These changes address a critical performance issue introduced in the latest release. The fix for the latest security issue included a change that made Asterisk randomly choose call numbers to make them more difficult to guess by attackers. However, due to some inefficient (this is by far, an understatement) code, when Asterisk chose high call numbers, chan_iax2 became unusable after just a small number of calls. On a small embedded platform, it would not be able to handle a single call. On my Intel Core 2 Duo @ 2.33 GHz, I couldn't run more than about 16 IAX2 channels. Ouch. These changes address some performance issues of the find_callno() function that have bothered me for a very long time. On every incoming media frame, it iterated through every possible call number trying to find a matching active call. This involved a mutex lock and unlock for each call number checked. So, if the random call number chosen was 20000, then every media frame would cause 20000 locks and unlocks. Previously, this problem was not as obvious since Asterisk always chose the lowest call number it could. A second container for IAX2 pvt structs has been added. It is an astobj2 hash table. When we know the remote side's call number, the pvt goes into the hash table with a hash value of the remote side's call number. Then, lookups for incoming media frames are a very fast hash lookup instead of an absolutely insane array traversal. In a quick test, I was able to get more than 3600% more IAX2 channels on my machine with these changes. ........ ................ 2008-04-30 16:15 +0000 [r114889] Jeff Peeler * /, channels/chan_console.c: Merged revisions 114888 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114888 | jpeeler | 2008-04-30 11:14:43 -0500 (Wed, 30 Apr 2008) | 3 lines Fixes a bug where if a stream monitor thread was not created (caused from failure of opening or starting the stream) pthread_cancel was called with an invalid thread ID. ........ 2008-04-30 14:55 +0000 [r114877-114886] Kevin P. Fleming * /, channels/iax2.h, channels/chan_iax2.c: Merged revisions 114884 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114884 | kpfleming | 2008-04-30 09:49:51 -0500 (Wed, 30 Apr 2008) | 10 lines Merged revisions 114880 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114880 | kpfleming | 2008-04-30 09:46:57 -0500 (Wed, 30 Apr 2008) | 2 lines use the ARRAY_LEN macro for indexing through the iaxs/iaxsl arrays so that the size of the arrays can be adjusted in one place, and change the size of the arrays from 32768 calls to 2048 calls when LOW_MEMORY is defined ........ ................ * /, Makefile.rules: Merged revisions 114876 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114876 | kpfleming | 2008-04-30 07:15:43 -0500 (Wed, 30 Apr 2008) | 10 lines Merged revisions 114875 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114875 | kpfleming | 2008-04-30 07:14:07 -0500 (Wed, 30 Apr 2008) | 2 lines pay attention to *all* header files for dependency tracking, not just the local ones (inspired by r578 of asterisk-addons by tilghman) ........ ................ 2008-04-29 22:55 +0000 [r114867] Jeff Peeler * /, channels/iax2-provision.c: Merged revisions 114866 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114866 | jpeeler | 2008-04-29 17:54:14 -0500 (Tue, 29 Apr 2008) | 2 lines Fixes a problem where all the templates were marked as dead no matter what. The templates should only be marked as dead if a configuration file has been successfully loaded and has changes. Bug found while making API documentation for 1.6.0. ........ 2008-04-29 21:09 +0000 [r114850-114858] Mark Michelson * /, apps/app_queue.c: Merged revisions 114849 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114849 | mmichelson | 2008-04-29 14:42:04 -0500 (Tue, 29 Apr 2008) | 22 lines Merged revisions 114848 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114848 | mmichelson | 2008-04-29 14:40:06 -0500 (Tue, 29 Apr 2008) | 14 lines Use the MACRO_CONTEXT and MACRO_EXTEN channel variables instead of the channel's macrocontext and macroexten fields. This is needed because if macros are daisy-chained, the incorrect context and extension are placed on the new channel. I also added locking to the channel prior to accessing these variables as noted in trunk's janitor project file. (closes issue #12549) Reported by: darren1713 Patches: app_queue.c.macroextenpatch uploaded by darren1713 (license 116) (with modifications from me) Tested by: putnopvut ........ ................ 2008-04-29 19:04 +0000 [r114846] Kevin P. Fleming * /: bug is not present in this branch 2008-04-29 17:11 +0000 [r114831] Jason Parker * res/res_config_pgsql.c, /: Merged revisions 114830 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114830 | qwell | 2008-04-29 12:10:55 -0500 (Tue, 29 Apr 2008) | 9 lines Merged revisions 114829 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114829 | qwell | 2008-04-29 12:08:55 -0500 (Tue, 29 Apr 2008) | 1 line Change warning message to debug, since there are cases where 0 results is perfectly fine. ........ ................ 2008-04-29 12:55 +0000 [r114825] Kevin P. Fleming * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 114824 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114824 | kpfleming | 2008-04-29 07:54:31 -0500 (Tue, 29 Apr 2008) | 18 lines Merged revisions 114823 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r114823 | kpfleming | 2008-04-29 07:53:12 -0500 (Tue, 29 Apr 2008) | 10 lines Merged revisions 114822 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r114822 | kpfleming | 2008-04-29 07:52:32 -0500 (Tue, 29 Apr 2008) | 2 lines stop script from appending source code if run multiple times ........ ................ ................ 2008-04-28 17:04 +0000 [r114777] Matthew Fredrickson * channels/chan_zap.c, /: Merged revisions 114776 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114776 | mattf | 2008-04-28 12:00:38 -0500 (Mon, 28 Apr 2008) | 1 line Fix deadlock issue in chan_zap with libss7 due to channel variables being set with the channel pvt lock being held. #12512 ........ 2008-04-28 13:44 +0000 [r114714] Joshua Colp * /, configure, configure.ac: Merged revisions 114713 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114713 | file | 2008-04-28 10:42:13 -0300 (Mon, 28 Apr 2008) | 2 lines Update autoconf logic with latest API change for libss7. ........ 2008-04-28 04:54 +0000 [r114707-114710] Tilghman Lesher * channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged revisions 114709 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114709 | tilghman | 2008-04-27 23:53:20 -0500 (Sun, 27 Apr 2008) | 13 lines Merged revisions 114708 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114708 | tilghman | 2008-04-27 23:47:39 -0500 (Sun, 27 Apr 2008) | 5 lines When modules are embedded, they take on a different name, without the ".so" extension. Specifically check for this name, when we're checking if a module is loaded. (Closes issue #12534) ........ ................ 2008-04-27 15:20 +0000 [r114701] Michiel van Baak * /, channels/chan_skinny.c: Merged revisions 114700 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk Merged to 1.6 because it fixes a crash. ........ r114700 | mvanbaak | 2008-04-27 17:17:18 +0200 (Sun, 27 Apr 2008) | 8 lines Make MWI in chan_skinny event based modeled after chan_zap and chan_mgcp. (closes issue #12214) Reported by: DEA Patches: chan_skinny-vm-events-v3.txt uploaded by DEA (license 3) Tested by: DEA and me ........ 2008-04-27 01:30 +0000 [r114697] Sean Bright * /, configure, configure.ac: Merged revisions 114696 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114696 | seanbright | 2008-04-26 21:28:32 -0400 (Sat, 26 Apr 2008) | 13 lines Merged revisions 114695 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114695 | seanbright | 2008-04-26 21:26:15 -0400 (Sat, 26 Apr 2008) | 5 lines When we don't explicitly pass a path to the --with-tds configure option, we may end up finding tds.h in /usr/local/include instead of /usr/include. If this happens, the grep that looks for the version (from tdsver.h) will fail and we'll have some problems during the build. ........ ................ 2008-04-26 15:09 +0000 [r114684-114693] Tilghman Lesher * /, contrib/scripts/vmail.cgi: Merged revisions 114690 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114690 | tilghman | 2008-04-26 08:17:19 -0500 (Sat, 26 Apr 2008) | 14 lines Merged revisions 114689 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114689 | tilghman | 2008-04-26 08:15:21 -0500 (Sat, 26 Apr 2008) | 6 lines Clicking forward without selecting a message leaves an errant .lock file. (closes issue #12528) Reported by: pukepail Patches: patch.diff uploaded by pukepail (license 431) ........ ................ 2008-04-25 22:05 +0000 [r114671-114677] Russell Bryant * /, pbx/pbx_lua.c: Merged revisions 114676 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114676 | russell | 2008-04-25 17:04:46 -0500 (Fri, 25 Apr 2008) | 7 lines Lock the channel around datastore access (closes issue #12527) Reported by: mnicholson Patches: pbx_lua4.diff uploaded by mnicholson (license 96) ........ * /, channels/chan_iax2.c: Merged revisions 114674 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114674 | russell | 2008-04-25 17:00:35 -0500 (Fri, 25 Apr 2008) | 11 lines Merged revisions 114673 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114673 | russell | 2008-04-25 16:54:40 -0500 (Fri, 25 Apr 2008) | 3 lines Use consistent logic for checking to see if a call number has been chosen yet. Also, remove some redundant logic I recently added in a fix. ........ ................ 2008-04-25 19:34 +0000 [r114664] Mark Michelson * apps/app_chanspy.c, /: Merged revisions 114663 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114663 | mmichelson | 2008-04-25 14:33:27 -0500 (Fri, 25 Apr 2008) | 12 lines Merged revisions 114662 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114662 | mmichelson | 2008-04-25 14:32:02 -0500 (Fri, 25 Apr 2008) | 4 lines Move the unlock of the spyee channel to outside the start_spying() function so that the channel is not unlocked twice when using whisper mode. ........ ................ 2008-04-25 16:26 +0000 [r114652] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 114651 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114651 | mmichelson | 2008-04-25 11:25:17 -0500 (Fri, 25 Apr 2008) | 4 lines Fix a memory leak and protect against potential dereferences of a NULL pointer. ........ 2008-04-24 22:14 +0000 [r114636] Joshua Colp * /, channels/chan_sip.c: Merged revisions 114635 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114635 | file | 2008-04-24 19:11:46 -0300 (Thu, 24 Apr 2008) | 4 lines Hey look, it builds. (closes issue #12519) Reported by: falves11 ........ 2008-04-24 21:36 +0000 [r114626-114634] Mark Michelson * /, channels/chan_sip.c: Merged revisions 114633 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114633 | mmichelson | 2008-04-24 16:35:39 -0500 (Thu, 24 Apr 2008) | 19 lines Merged revisions 114632 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114632 | mmichelson | 2008-04-24 16:35:08 -0500 (Thu, 24 Apr 2008) | 11 lines Re-invite RTP during a masquerade so that, for instance, an AMI redirect of two channels which are natively bridged will preserve audio on both channels. This prevents a problem with Asterisk not re-inviting due to one of the channels having being a zombie. (closes issue #12513) Reported by: mneuhauser Patches: asterisk-1.4-114602_restore-RTP-on-fixup.patch uploaded by mneuhauser (license 425) ........ ................ * /, apps/app_queue.c: Merged revisions 114629 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114629 | mmichelson | 2008-04-24 15:43:52 -0500 (Thu, 24 Apr 2008) | 16 lines Merged revisions 114628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114628 | mmichelson | 2008-04-24 15:43:03 -0500 (Thu, 24 Apr 2008) | 8 lines Output of channel variables when eventwhencalled=vars was set was being truncated two characters. This patch corrects the problem. (closes issue #12493) Reported by: davidw ........ ................ * channels/chan_local.c, /: Merged revisions 114625 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114625 | mmichelson | 2008-04-24 15:06:06 -0500 (Thu, 24 Apr 2008) | 18 lines Merged revisions 114624 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114624 | mmichelson | 2008-04-24 15:04:24 -0500 (Thu, 24 Apr 2008) | 10 lines Resolve a deadlock in chan_local by releasing the channel lock temporarily. (closes issue #11712) Reported by: callguy Patches: 11712.patch uploaded by putnopvut (license 60) Tested by: acunningham ........ ................ 2008-04-24 19:55 +0000 [r114619-114623] Tilghman Lesher * channels/chan_local.c, /: Merged revisions 114622 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114622 | tilghman | 2008-04-24 14:54:57 -0500 (Thu, 24 Apr 2008) | 12 lines Merged revisions 114621 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114621 | tilghman | 2008-04-24 14:53:36 -0500 (Thu, 24 Apr 2008) | 4 lines Ensure that when we set the accountcode, it actually shows up in the CDR. (Fix for AMI Originate) (Closes issue #12007) ........ ................ * /, apps/app_meetme.c: Merged revisions 114617 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114617 | tilghman | 2008-04-24 14:24:31 -0500 (Thu, 24 Apr 2008) | 6 lines Fix DST calculation, and fix bug in calculation of whether conf has started yet or not (Closes issue #12292) Reported by: DEA Patches: app_meetme-rt-dst-sched-fix.txt uploaded by DEA (license 3) ........ 2008-04-24 16:48 +0000 [r114613] Jason Parker * channels/chan_misdn.c, /: Merged revisions 114612 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114612 | qwell | 2008-04-24 11:47:01 -0500 (Thu, 24 Apr 2008) | 17 lines Merged revisions 51989 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #12496) Reported by: daniele Patches: misdn-moh-1.6.0-beta7.1.patch uploaded by daniele (license 471) Tested by: daniele Technically, I didn't use the patch above except to find out what revision to merge - but it's the same thing as this revision. ........ r51989 | crichter | 2007-01-24 06:57:22 -0600 (Wed, 24 Jan 2007) | 1 line added fix from #8899 ........ ................ 2008-04-24 15:57 +0000 [r114610] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 114609 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114609 | russell | 2008-04-24 10:56:55 -0500 (Thu, 24 Apr 2008) | 12 lines Merged revisions 114608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114608 | russell | 2008-04-24 10:55:21 -0500 (Thu, 24 Apr 2008) | 4 lines Fix a silly mistake in a change I made yesterday that caused chan_iax2 to blow up very quickly. (issue #12515) ........ ................ 2008-04-24 15:00 +0000 [r114607] Olle Johansson * channels/chan_sip.c: Merged revisions 114606 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114606 | oej | 2008-04-24 16:59:05 +0200 (Tor, 24 Apr 2008) | 11 lines Merged revisions 114603 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114603 | oej | 2008-04-24 16:55:18 +0200 (Tor, 24 Apr 2008) | 3 lines Only have one max-forwards header in outbound REFERs. Discovered in the Asterisk SIP Masterclass in Orlando. Thanks Joe! ........ ................ 2008-04-24 14:56 +0000 [r114599-114605] Russell Bryant * /, channels/chan_sip.c: Merged revisions 114604 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114604 | russell | 2008-04-24 09:55:21 -0500 (Thu, 24 Apr 2008) | 3 lines Change a verbose message to debug. (closes issue #12514) ........ * /, main/http.c: Merged revisions 114601 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114601 | russell | 2008-04-23 17:53:20 -0500 (Wed, 23 Apr 2008) | 14 lines Merged revisions 114600 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114600 | russell | 2008-04-23 17:18:12 -0500 (Wed, 23 Apr 2008) | 6 lines Improve some broken cookie parsing code. Previously, manager login over HTTP would only work if the mansession_id cookie was first. Now, the code builds a list of all of the cookies in the Cookie header. This fixes a problem observed by users of the Asterisk GUI. (closes AST-20) ........ ................ * apps/app_chanspy.c, /: Merged revisions 114598 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114598 | russell | 2008-04-23 15:53:05 -0500 (Wed, 23 Apr 2008) | 18 lines Merged revisions 114597 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114597 | russell | 2008-04-23 15:49:18 -0500 (Wed, 23 Apr 2008) | 10 lines Fix an issue that caused getting the correct next channel to not always work. Also, remove setting the amount of time to wait for a digit from 5 seconds back down to 1/10 of a second. I believe this was so the beep didn't get played over and over really fast, but a while back I put in another fix for that issue. (closes issue #12498) Reported by: jsmith Patches: app_chanspy_channel_walk.trunk.patch uploaded by jsmith (license 15) ........ ................ 2008-04-23 18:34 +0000 [r114596] Jason Parker * /, res/res_musiconhold.c: Merged revisions 114595 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114595 | qwell | 2008-04-23 13:33:28 -0500 (Wed, 23 Apr 2008) | 16 lines Merged revisions 114594 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114594 | qwell | 2008-04-23 13:28:44 -0500 (Wed, 23 Apr 2008) | 8 lines Fix reload/unload for res_musiconhold module. (closes issue #11575) Reported by: sunder Patches: M11575_14_rev3.diff uploaded by junky (license 177) bug11575_trunk.diff.txt uploaded by jamesgolovich (license 176) ........ ................ 2008-04-23 18:01 +0000 [r114589-114593] Russell Bryant * main/manager.c, /, include/asterisk/manager.h: Merged revisions 114592 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114592 | russell | 2008-04-23 13:01:00 -0500 (Wed, 23 Apr 2008) | 13 lines Merged revisions 114591 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114591 | russell | 2008-04-23 12:55:31 -0500 (Wed, 23 Apr 2008) | 5 lines Store the manager session ID explicitly as 4 byte ID instead of a ulong. The mansession_id cookie is coded to be limited to 8 characters of hex, and this could break logins from 64-bit machines in some cases. (inspired by AST-20) ........ ................ * /, channels/chan_iax2.c: Merged revisions 114588 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114588 | russell | 2008-04-23 12:18:29 -0500 (Wed, 23 Apr 2008) | 10 lines Merged revisions 114587 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114587 | russell | 2008-04-23 12:16:32 -0500 (Wed, 23 Apr 2008) | 2 lines Fix find_callno_locked() to actually return the callno locked in some more cases. ........ ................ 2008-04-23 16:57 +0000 [r114586] Olle Johansson * channels/chan_sip.c: Merged revisions 114585 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114585 | oej | 2008-04-23 18:53:34 +0200 (Ons, 23 Apr 2008) | 10 lines Merged revisions 114584 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114584 | oej | 2008-04-23 18:51:41 +0200 (Ons, 23 Apr 2008) | 2 lines Add 502 support for both directions, not only one... (see r114571) ........ ................ 2008-04-23 14:56 +0000 [r114581] Joshua Colp * main/pbx.c, /: Merged revisions 114580 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114580 | file | 2008-04-23 11:55:03 -0300 (Wed, 23 Apr 2008) | 12 lines Merged revisions 114579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114579 | file | 2008-04-23 11:54:11 -0300 (Wed, 23 Apr 2008) | 4 lines Instead of stopping dialplan execution when SayNumber attempts to say a large number that it can not print out a message informing the user and continue on. (closes issue #12502) Reported by: bcnit ........ ................ 2008-04-23 01:00 +0000 [r114576-114578] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 114575 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114575 | mmichelson | 2008-04-22 19:40:30 -0500 (Tue, 22 Apr 2008) | 10 lines Round 1 of IMAP_STORAGE-related app_voicemail changes This makes IMAP_STORAGE include the proper headers if you have specified the "system" option for --with-imap when running the configure script and your IMAP-related headers exist in /usr/include/c-client. This change is due to a hasty merge of a 1.4 change I made. ........ 2008-04-22 23:59 +0000 [r114573] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 114572 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114572 | tilghman | 2008-04-22 18:58:19 -0500 (Tue, 22 Apr 2008) | 10 lines Merged revisions 114571 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114571 | tilghman | 2008-04-22 18:51:44 -0500 (Tue, 22 Apr 2008) | 2 lines Treat a 502 just like a 503, when it comes to processing a response code ........ ................ 2008-04-22 Russell Bryant * Asterisk 1.6.0-beta8 released. 2008-04-22 22:18 +0000 [r114560] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 114559 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114559 | russell | 2008-04-22 17:17:31 -0500 (Tue, 22 Apr 2008) | 13 lines Merged revisions 114558 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114558 | russell | 2008-04-22 17:15:36 -0500 (Tue, 22 Apr 2008) | 5 lines When we receive a full frame that is supposed to contain our call number, ensure that it has the correct one. (closes issue #10078) (AST-2008-006) ........ ................ 2008-04-22 22:04 +0000 [r114556] Steve Murphy * main/pbx.c, /: Merged revisions 114553 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114553 | murf | 2008-04-22 15:57:57 -0600 (Tue, 22 Apr 2008) | 14 lines (closes issue #12469) Reported by: triccyx I had a bit a problem reproducing this in my setup (trying not to disturb my other stuff) but finally, I got it. The problem appears to be that the extension is being added in replace mode, which kinda assumes that the pattern trie has been formed, when in fact, in this case, it was not. The checks being done are not nec. when the tree is not yet formed, as changes like this will be summarized when the trie is formed in the future. I tested the fix, and the crash no longer happens. Feel free to open the bug again if this fix doesn't cure the problem. ........ 2008-04-22 21:16 +0000 [r114544-114552] Russell Bryant * main/channel.c, /: Merged revisions 114548 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114548 | russell | 2008-04-22 15:25:56 -0500 (Tue, 22 Apr 2008) | 2 lines re-add a fix that got lost with a recent change ........ 2008-04-22 18:14 +0000 [r114541] Jason Parker * main/pbx.c, /, include/asterisk/pbx.h, apps/app_queue.c: Merged revisions 114540 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114540 | qwell | 2008-04-22 13:14:09 -0500 (Tue, 22 Apr 2008) | 8 lines Allow setqueuevar=yes (et al) to work, after changes to pbx_builtin_setvar() (closes issue #12490) Reported by: bcnit Patches: 12490-queuevars-3.diff uploaded by qwell (license 4) Tested by: qwell ........ 2008-04-22 18:06 +0000 [r114534-114539] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 114538 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114538 | russell | 2008-04-22 13:04:39 -0500 (Tue, 22 Apr 2008) | 17 lines Merged revisions 114537 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114537 | russell | 2008-04-22 13:03:33 -0500 (Tue, 22 Apr 2008) | 9 lines If the dial string passed to the call channel callback does not indicate an extension, then consider the extension on the channel before falling back to the default. (closes issue #12479) Reported by: darren1713 Patches: exten_dial_fix_chan_iax2.c.patch uploaded by darren1713 (license 116) ........ ................ 2008-04-22 15:46 +0000 [r114524-114528] Russell Bryant * main/manager.c, /: Merged revisions 114527 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114527 | russell | 2008-04-22 10:46:01 -0500 (Tue, 22 Apr 2008) | 8 lines Correct action_ping() and action_events() with regards to Manager 1.1 documentation. Also, fix a bug in xml_translate(). (closes issue #11649) Reported by: ys Patches: trunk_manager.c.diff uploaded by ys (license 281) ........ 2008-04-21 20:23 +0000 [r114422] Matthew Fredrickson * channels/chan_zap.c, /: Merged revisions 114389 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114389 | mattf | 2008-04-21 13:44:35 -0500 (Mon, 21 Apr 2008) | 1 line Add support for generic name transmission (#12484) on SS7 in chan_zap ........ 2008-04-21 15:38 +0000 [r114328] Jeff Peeler * /, apps/app_authenticate.c: Merged revisions 114327 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114327 | jpeeler | 2008-04-21 10:34:37 -0500 (Mon, 21 Apr 2008) | 2 lines This removes an invalid warning message for an incorrectly entered pin, but more importantly removes an inapplicable check. If the first argument passed to app_authenticate does not contain a '/', the argument should be treated as the sole fixed "password" to match against and that is all. (Previous behavior was attempting to open a file based on the pin.) ........ 2008-04-21 14:42 +0000 [r114321-114324] Joshua Colp * /, channels/chan_sip.c: Merged revisions 114323 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114323 | file | 2008-04-21 11:40:33 -0300 (Mon, 21 Apr 2008) | 12 lines Merged revisions 114322 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114322 | file | 2008-04-21 11:39:32 -0300 (Mon, 21 Apr 2008) | 4 lines Only drop audio if we receive it without a progress indication. We allow other frames through such as DTMF because they may be needed to complete the call. (closes issue #12440) Reported by: aragon ........ ................ * /, res/res_config_ldap.c: Merged revisions 114320 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114320 | file | 2008-04-21 11:34:06 -0300 (Mon, 21 Apr 2008) | 6 lines Only print out the error message if ldap_modify_ext_s actually returns an error, and not success. (closes issue #12438) Reported by: gservat Patches: res_config_ldap.c-patch-code uploaded by gservat (license 466) ........ 2008-04-19 17:00 +0000 [r114304] Matthew Fredrickson * channels/chan_zap.c: SS7:Added - Generic Name / Access Transport / Redirecting Number handling. #12425 2008-04-18 21:51 +0000 [r114277-114286] Russell Bryant * main/manager.c, /: Merged revisions 114285 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114285 | russell | 2008-04-18 16:51:05 -0500 (Fri, 18 Apr 2008) | 10 lines Merged revisions 114284 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114284 | russell | 2008-04-18 16:48:06 -0500 (Fri, 18 Apr 2008) | 2 lines Don't destroy a manager session if poll() returns an error of EAGAIN. ........ ................ * Makefile, /: Merged revisions 114279 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114279 | russell | 2008-04-18 15:01:47 -0500 (Fri, 18 Apr 2008) | 10 lines Merged revisions 114278 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114278 | russell | 2008-04-18 15:01:09 -0500 (Fri, 18 Apr 2008) | 2 lines ensure directories are created before we try to install stuff into them ........ ................ * Makefile, /: Merged revisions 114276 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114276 | russell | 2008-04-18 14:59:17 -0500 (Fri, 18 Apr 2008) | 10 lines Merged revisions 114275 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114275 | russell | 2008-04-18 14:58:55 -0500 (Fri, 18 Apr 2008) | 2 lines SUBDIRS_INSTALL is already listed as a subtarget for bininstall ........ ................ 2008-04-18 19:36 +0000 [r114262-114272] Joshua Colp * channels/chan_unistim.c, /: Merged revisions 114271 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114271 | file | 2008-04-18 16:35:33 -0300 (Fri, 18 Apr 2008) | 4 lines Make sure ADSI is marked as unavailable on Unistim channels so voicemail does not try to do some ADSI jazz. (closes issue #12460) Reported by: PerryB ........ 2008-04-18 18:04 +0000 [r114260] Mark Michelson * channels/chan_zap.c, /, main/callerid.c: Merged revisions 114259 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114259 | mmichelson | 2008-04-18 13:03:06 -0500 (Fri, 18 Apr 2008) | 14 lines Merged revisions 114257 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114257 | mmichelson | 2008-04-18 12:44:29 -0500 (Fri, 18 Apr 2008) | 6 lines Clearing up error messages so they make a bit more sense. Also removing a redundant error message. Issue AST-15 ........ ................ 2008-04-18 16:12 +0000 [r114255] Joshua Colp * /, res/res_config_ldap.c: Merged revisions 114254 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114254 | file | 2008-04-18 13:11:27 -0300 (Fri, 18 Apr 2008) | 4 lines If the parsing of the config file fails make sure we unlock ldap_lock. (closes issue #12477) Reported by: IgorG ........ 2008-04-18 13:40 +0000 [r114247] Sean Bright * channels/chan_sip.c: Merged revisions 114246 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114246 | seanbright | 2008-04-18 09:38:07 -0400 (Fri, 18 Apr 2008) | 9 lines Merged revisions 114245 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114245 | seanbright | 2008-04-18 09:33:32 -0400 (Fri, 18 Apr 2008) | 1 line Only complete the SIP channel name once for 'sip show channel ' ........ ................ 2008-04-18 06:54 +0000 [r114244] Tilghman Lesher * apps/app_setcallerid.c, /: Merged revisions 114243 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114243 | tilghman | 2008-04-18 01:53:47 -0500 (Fri, 18 Apr 2008) | 11 lines Merged revisions 114242 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114242 | tilghman | 2008-04-18 01:49:16 -0500 (Fri, 18 Apr 2008) | 3 lines For consistency sake, ensure that the values that ${CALLINGPRES} returns are valid as an input to SetCallingPres. (Closes issue #12472) ........ ................ 2008-04-17 23:09 +0000 [r114232-114241] Russell Bryant * /, channels/chan_sip.c: Merged revisions 114151 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114151 | oej | 2008-04-15 15:39:29 -0500 (Tue, 15 Apr 2008) | 10 lines Merged revisions 114148 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114148 | oej | 2008-04-15 22:26:05 +0200 (Tis, 15 Apr 2008) | 2 lines Handle subscribe queues in all situations... Thanks to festr_ on irc for telling me about this bug. ........ ................ * /, channels/chan_sip.c: Merged revisions 114150 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114150 | oej | 2008-04-15 15:31:08 -0500 (Tue, 15 Apr 2008) | 2 lines Adding chanvar to SIPPEER from 1.4 branch ........ * main/autoservice.c, /: Merged revisions 114233 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114233 | russell | 2008-04-17 17:24:00 -0500 (Thu, 17 Apr 2008) | 14 lines Merged revisions 114230 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114230 | russell | 2008-04-17 17:15:43 -0500 (Thu, 17 Apr 2008) | 6 lines Remove redundant safety net. The check for the autoservice channel list state accomplishes the same goal in a better way. (issue #12470) Reported By: atis ........ ................ 2008-04-17 21:05 +0000 [r114228] Mark Michelson * apps/app_chanspy.c, /: Merged revisions 114227 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114227 | mmichelson | 2008-04-17 16:04:40 -0500 (Thu, 17 Apr 2008) | 17 lines Merged revisions 114226 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114226 | mmichelson | 2008-04-17 16:03:29 -0500 (Thu, 17 Apr 2008) | 9 lines Declaration of the peer channel in this scope was making it so the peer variable defined in the outer scope was never set properly, therefore making iterating through the channel list always restart from the beginning. This bug would have affected anyone who called chanspy without specifying a first argument. (closes issue #12461) Reported by: stever28 ........ ................ 2008-04-17 16:51 +0000 [r114210-114213] Mark Michelson * main/dsp.c, main/frame.c, /, include/asterisk/dsp.h, include/asterisk/frame.h: Merged revisions 114208 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114208 | mmichelson | 2008-04-17 11:40:12 -0500 (Thu, 17 Apr 2008) | 20 lines Merged revisions 114207 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114207 | mmichelson | 2008-04-17 11:28:03 -0500 (Thu, 17 Apr 2008) | 12 lines It was possible for a reference to a frame which was part of a freed DSP to still be referenced, leading to memory corruption and eventual crashes. This code change ensures that the dsp is freed when we are finished with the frame. This change is very similar to a change Russell made with translators back a month or so ago. (closes issue #11999) Reported by: destiny6628 Patches: 11999.patch uploaded by putnopvut (license 60) Tested by: destiny6628, victoryure ........ ................ 2008-04-17 16:26 +0000 [r114206] Russell Bryant * Makefile, /: Merged revisions 114205 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114205 | russell | 2008-04-17 11:25:29 -0500 (Thu, 17 Apr 2008) | 11 lines Merged revisions 114204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114204 | russell | 2008-04-17 11:23:45 -0500 (Thu, 17 Apr 2008) | 3 lines Fix the bininstall target to install from subdirs, as well. (closes issue AST-8, patch from bmd at switchvox) ........ ................ 2008-04-17 15:17 +0000 [r114203] Tilghman Lesher * doc/CODING-GUIDELINES, /: Merged revisions 114202 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114202 | tilghman | 2008-04-17 10:12:52 -0500 (Thu, 17 Apr 2008) | 2 lines fileio.h does not exist; io.h does, though. ........ 2008-04-17 13:55 +0000 [r114200] Philippe Sultan * /, res/res_jabber.c: Merged revisions 114199 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114199 | phsultan | 2008-04-17 15:46:17 +0200 (Thu, 17 Apr 2008) | 10 lines Merged revisions 114198 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114198 | phsultan | 2008-04-17 15:42:23 +0200 (Thu, 17 Apr 2008) | 2 lines Use keepalives effectively in order diagnose bug #12432. ........ ................ 2008-04-17 12:59 +0000 [r114197] Tilghman Lesher * /, res/res_agi.c: Merged revisions 114196 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114196 | tilghman | 2008-04-17 07:59:04 -0500 (Thu, 17 Apr 2008) | 16 lines Merged revisions 114195 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114195 | tilghman | 2008-04-17 07:56:38 -0500 (Thu, 17 Apr 2008) | 8 lines Add special case for when the agi cannot be executed, to comply with the documentation that we return failure in that case. (closes issue #12462) Reported by: fmueller Patches: 20080416__bug12462.diff.txt uploaded by Corydon76 (license 14) Tested by: fmueller ........ ................ 2008-04-17 10:56 +0000 [r114193] Sean Bright * apps/app_chanspy.c, /: Merged revisions 114192 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114192 | seanbright | 2008-04-17 06:55:05 -0400 (Thu, 17 Apr 2008) | 9 lines Merged revisions 114191 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114191 | seanbright | 2008-04-17 06:51:20 -0400 (Thu, 17 Apr 2008) | 1 line Make sure we have enough room for the recording's filename. ........ ................ 2008-04-16 20:48 +0000 [r114186] Kevin P. Fleming * channels/chan_zap.c, /: Merged revisions 114185 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114185 | kpfleming | 2008-04-16 15:47:30 -0500 (Wed, 16 Apr 2008) | 14 lines Merged revisions 114184 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114184 | kpfleming | 2008-04-16 15:46:38 -0500 (Wed, 16 Apr 2008) | 6 lines use the ZT_SET_DIALPARAMS ioctl properly by initializing the structure to all zeroes in case it contains fields that we don't write values into (which it does as of Zaptel 1.4.10) (closes issue #12456) Reported by: fnordian ........ ................ 2008-04-15 20:53 +0000 [r114153] Tilghman Lesher * /, cdr/cdr_pgsql.c: Merged revisions 114152 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114152 | tilghman | 2008-04-15 15:51:08 -0500 (Tue, 15 Apr 2008) | 2 lines Oops, buffer wasn't long enough for query ........ 2008-04-15 20:09 +0000 [r114147] Steve Murphy * main/pbx.c, /: Merged revisions 114146 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114146 | murf | 2008-04-15 13:59:50 -0600 (Tue, 15 Apr 2008) | 8 lines These changes: a. fix a self-found problem with SPAWN-ing an extension, where matches were not being found b. correct some wording in a comment c. Add some debug for future debugging. ........ 2008-04-15 17:22 +0000 [r114132-114142] Jason Parker * channels/chan_unistim.c, /: Merged revisions 114141 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114141 | qwell | 2008-04-15 12:21:58 -0500 (Tue, 15 Apr 2008) | 8 lines Shorten the mac address pattern, since some phones use different identifiers (such as the i2050 softphone). (closes issue #12398) Reported by: c_hans Patches: chan_unistim_svn.diff uploaded by c (license 460) Tested by: c_hans ........ * contrib/scripts/autosupport, /: Merged revisions 114139 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114139 | qwell | 2008-04-15 12:17:37 -0500 (Tue, 15 Apr 2008) | 15 lines Merged revisions 114138 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114138 | qwell | 2008-04-15 12:17:18 -0500 (Tue, 15 Apr 2008) | 7 lines Update Digium autosupport script, for more useful information. (closes issue #12452) Reported by: angler Patches: autosupport.diff uploaded by angler (license 106) ........ ................ * /, apps/app_queue.c: Merged revisions 114134 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114134 | qwell | 2008-04-15 11:18:38 -0500 (Tue, 15 Apr 2008) | 16 lines Merged revisions 114133 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114133 | qwell | 2008-04-15 11:18:08 -0500 (Tue, 15 Apr 2008) | 8 lines Allow autofill to work in the general section of queues.conf. Additionally, don't try to (re)set options when they have empty values in realtime (all unset columns would have an empty value). (closes issue #12445) Reported by: atis Patches: 12445-autofill.diff uploaded by qwell (license 4) ........ ................ 2008-04-14 18:34 +0000 [r114122] Jason Parker * /, channels/chan_h323.c: Merged revisions 114121 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114121 | qwell | 2008-04-14 13:34:17 -0500 (Mon, 14 Apr 2008) | 15 lines Merged revisions 114120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114120 | qwell | 2008-04-14 13:31:57 -0500 (Mon, 14 Apr 2008) | 7 lines The call_token on the pvt can occasionally be NULL, causing a crash. If it is NULL, we can skip this channel, since it can't the one we're looking for. (closes issue #9299) Reported by: vazir ........ ................ 2008-04-14 17:42 +0000 [r114119] Mark Michelson * main/channel.c, /: Merged revisions 114118 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114118 | mmichelson | 2008-04-14 12:42:20 -0500 (Mon, 14 Apr 2008) | 19 lines Merged revisions 114117 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114117 | mmichelson | 2008-04-14 12:41:03 -0500 (Mon, 14 Apr 2008) | 11 lines Increase the retry count when attempting to show channels. This apparently cleared an issue someone was seeing when attempting to show channels when the load was high. (closes issue #11667) Reported by: falves11 Patches: 11677.txt uploaded by russell (license 2) Tested by: falves11 ........ ................ 2008-04-14 16:33 +0000 [r114116] Tilghman Lesher * /, contrib/scripts/astcli: Merged revisions 114115 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114115 | tilghman | 2008-04-14 11:32:59 -0500 (Mon, 14 Apr 2008) | 2 lines Make tab-completion work for all cases ........ 2008-04-14 16:25 +0000 [r114114] Mark Michelson * apps/app_dial.c, /, apps/app_queue.c: Merged revisions 114113 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114113 | mmichelson | 2008-04-14 11:25:09 -0500 (Mon, 14 Apr 2008) | 17 lines Merged revisions 114112 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114112 | mmichelson | 2008-04-14 11:24:22 -0500 (Mon, 14 Apr 2008) | 9 lines If the datastore has been moved to another channel due to a masquerade, then freeing the datastore here causes an eventual double free when the new channel hangs up. We should only free the datastore if we were able to successfully remove it from the channel we are referencing (i.e. the datastore was not moved). (closes issue #12359) Reported by: pguido ........ ................ 2008-04-14 15:02 +0000 [r114108] Mark Michelson * main/channel.c, /: Merged revisions 114107 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114107 | mmichelson | 2008-04-14 10:01:36 -0500 (Mon, 14 Apr 2008) | 13 lines Merged revisions 114106 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114106 | mmichelson | 2008-04-14 09:58:02 -0500 (Mon, 14 Apr 2008) | 5 lines Save a local copy of the generate callback prior to unlocking the channel in case the generate callback goes NULL on us after the channel is unlocked. Thanks to Russell for pointing this need out to me. ........ ................ 2008-04-14 14:54 +0000 [r114102-114105] Joshua Colp * /, channels/chan_sip.c: Merged revisions 114104 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114104 | file | 2008-04-14 11:53:33 -0300 (Mon, 14 Apr 2008) | 12 lines Merged revisions 114103 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114103 | file | 2008-04-14 11:52:46 -0300 (Mon, 14 Apr 2008) | 4 lines It is possible for the remote side to say they want T38 but not give any capabilities. (closes issue #12414) Reported by: MVF ........ ................ * main/rtp.c, /: Merged revisions 114101 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114101 | file | 2008-04-14 10:53:33 -0300 (Mon, 14 Apr 2008) | 12 lines Merged revisions 114100 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114100 | file | 2008-04-14 10:52:49 -0300 (Mon, 14 Apr 2008) | 4 lines Don't change the SSRC when a new source comes into play, this might happen quite often and depending on the remote side... they might not like this. (closes issue #12353) Reported by: dimas ........ ................ 2008-04-14 02:59 +0000 [r114097-114099] Tilghman Lesher * /, contrib/scripts/astcli: Merged revisions 114098 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114098 | tilghman | 2008-04-13 21:55:41 -0500 (Sun, 13 Apr 2008) | 3 lines Add tab command-line completion (Closes issue #12428) ........ * /, apps/app_meetme.c: Merged revisions 114096 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114096 | tilghman | 2008-04-13 09:35:43 -0500 (Sun, 13 Apr 2008) | 3 lines Use ast_mkdir instead of mkdir (Closes issue #12430) ........ 2008-04-12 16:22 +0000 [r114094-114095] Matthew Fredrickson * channels/chan_zap.c: Make sure linkset is locked exiting ss7_start_call * channels/chan_zap.c: Make sure we start incoming calls on SS7 with echo cancellation enabled. Also make sure when completing a COT we call ss7_start_call with the proper locks held. Lastly, make sure if we fail to get a channel from zt_new that we don't assume it's there. 2008-04-11 23:27 +0000 [r114089-114091] Tilghman Lesher * /, cdr/cdr_pgsql.c: Merged revisions 114090 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114090 | tilghman | 2008-04-11 18:26:56 -0500 (Fri, 11 Apr 2008) | 3 lines If any field is not null, but has no default, then it must be set or the insert will fail. (Closes issue #12285) ........ * /, configs/res_ldap.conf.sample: Merged revisions 114088 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114088 | tilghman | 2008-04-11 18:21:54 -0500 (Fri, 11 Apr 2008) | 3 lines Make the sample config match the contributed LDAP schema (Closes issue #12421) ........ 2008-04-11 23:21 +0000 [r114087] Terry Wilson * /, channels/chan_iax2.c: Merged revisions 114084 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114084 | twilson | 2008-04-11 17:48:52 -0500 (Fri, 11 Apr 2008) | 15 lines Merged revisions 114083 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114083 | twilson | 2008-04-11 17:32:51 -0500 (Fri, 11 Apr 2008) | 7 lines Several places in the code called find_callno() (which releases the lock on the pvt structure) and then immediately locked the call and did things with it. Unfortunately, the call can disappear between the find_callno and the lock, causing Bad Stuff(tm) to happen. Added find_callno_locked() function to return the callno withtout unlocking for instances that it is needed. (issue #12400) Reported by: ztel ........ ................ 2008-04-11 23:13 +0000 [r114086] Tilghman Lesher * /, res/res_config_ldap.c: Merged revisions 114085 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114085 | tilghman | 2008-04-11 18:12:16 -0500 (Fri, 11 Apr 2008) | 7 lines Use the correct function for free'ing objects, and maybe we won't crash. (closes issue #12163) Reported by: gservat Patches: 20080411__bug12163.diff.txt uploaded by Corydon76 (license 14) Tested by: gservat ........ 2008-04-11 15:51 +0000 [r114065] Mark Michelson * /, main/features.c: Merged revisions 114064 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114064 | mmichelson | 2008-04-11 10:49:35 -0500 (Fri, 11 Apr 2008) | 19 lines Merged revisions 114063 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114063 | mmichelson | 2008-04-11 10:44:28 -0500 (Fri, 11 Apr 2008) | 11 lines Fix a race condition that may happen between a sip hangup and a "core show channel" command. This patch adds locking to prevent the resulting crash. (closes issue #12155) Reported by: tsearle Patches: show_channels_crash2.patch uploaded by tsearle (license 373) Tested by: tsearle ........ ................ 2008-04-11 14:56 +0000 [r114062] Tilghman Lesher * /, res/res_config_ldap.c: Merged revisions 114061 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114061 | tilghman | 2008-04-11 09:54:22 -0500 (Fri, 11 Apr 2008) | 6 lines Errors are all greater than 0 (closes issue #12422) Reported by: nito Patches: res_config_ldap_result_check_patch.diff uploaded by nito (license 340) ........ 2008-04-10 22:23 +0000 [r114056] Mark Michelson * utils/conf2ael.c, utils/check_expr.c, utils/Makefile, main/manager.c, /, utils/astman.c, utils/hashtest.c, main/utils.c, include/asterisk/lock.h, utils/ael_main.c, utils/hashtest2.c: Merged revisions 114052 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114052 | mmichelson | 2008-04-10 17:02:32 -0500 (Thu, 10 Apr 2008) | 11 lines Merged revisions 114051 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114051 | mmichelson | 2008-04-10 15:59:49 -0500 (Thu, 10 Apr 2008) | 3 lines Fix 1.4 build when LOW_MEMORY is enabled. ........ ................ 2008-04-10 19:59 +0000 [r114047] Mark Michelson * /, channels/chan_sip.c: Merged revisions 114046 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114046 | mmichelson | 2008-04-10 14:58:36 -0500 (Thu, 10 Apr 2008) | 14 lines Merged revisions 114045 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114045 | mmichelson | 2008-04-10 14:55:33 -0500 (Thu, 10 Apr 2008) | 6 lines Be sure that we're not about to set bridgepvt NULL prior to dereferencing it. (closes issue #11775) Reported by: fujin ........ ................ 2008-04-10 19:09 +0000 [r114043] Tilghman Lesher * /, contrib/scripts/astcli: Merged revisions 114042 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114042 | tilghman | 2008-04-10 14:04:29 -0500 (Thu, 10 Apr 2008) | 7 lines The hydra grows yet another head... (closes issue #12401) Reported by: davevg Patches: astcli.diff2 uploaded by davevg (license 209) Tested by: davevg, Corydon76 ........ 2008-04-10 17:27 +0000 [r114037] Jason Parker * /, main/file.c: Merged revisions 114036 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114036 | qwell | 2008-04-10 12:27:16 -0500 (Thu, 10 Apr 2008) | 18 lines Merged revisions 114035 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114035 | qwell | 2008-04-10 12:26:10 -0500 (Thu, 10 Apr 2008) | 10 lines Only try to prefix language if we are not using an absolute path (suffix it otherwise). en/var/lib/asterisk/sounds/blah.gsm is a very silly path. (closes issue #12379) Reported by: kuj Patches: 12379-absolutepath.diff uploaded by qwell (license 4) Tested by: kuj, qwell ........ ................ 2008-04-10 16:00 +0000 [r114023-114034] Joshua Colp * /, apps/app_meetme.c: Merged revisions 114030 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114030 | file | 2008-04-10 12:10:47 -0300 (Thu, 10 Apr 2008) | 14 lines Merged revisions 114029 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114029 | file | 2008-04-10 12:09:04 -0300 (Thu, 10 Apr 2008) | 6 lines Create the directory where name recordings will go if it does not exist. (closes issue #12311) Reported by: rkeene Patches: 12311-mkdir.diff uploaded by qwell (license 4) ........ ................ * apps/app_voicemail.c, /: Merged revisions 114027 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114027 | file | 2008-04-10 11:53:19 -0300 (Thu, 10 Apr 2008) | 6 lines Don't hardcode ru into the digits filename so that languageprefix can work. (closes issue #12404) Reported by: IgorG Patches: voicemail_ru_hardcoded-v1.patch uploaded by IgorG (license 20) ........ * main/rtp.c, channels/chan_unistim.c, /, channels/chan_skinny.c: Merged revisions 114024 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114024 | file | 2008-04-10 10:45:45 -0300 (Thu, 10 Apr 2008) | 4 lines Fix spelling of existent in a few places. (closes issue #12409) Reported by: candlerb ........ * /, channels/chan_sip.c: Merged revisions 114022 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114022 | file | 2008-04-10 10:28:30 -0300 (Thu, 10 Apr 2008) | 14 lines Merged revisions 114021 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114021 | file | 2008-04-10 10:27:11 -0300 (Thu, 10 Apr 2008) | 6 lines Don't add custom URI options if they don't exist OR they are empty. (closes issue #12407) Reported by: homesick Patches: uri_options-1.4.diff uploaded by homesick (license 91) ........ ................ 2008-04-09 22:34 +0000 [r113929-113982] Mark Michelson * /, apps/app_queue.c: Merged revisions 113980 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r113980 | mmichelson | 2008-04-09 17:32:32 -0500 (Wed, 09 Apr 2008) | 8 lines Fix a crash that happened due to accessing free'd memory (closes issue #12396) Reported by: tcalosi Patches: 12396.patch uploaded by putnopvut (license 60) Tested by: tcalosi ........ * /, channels/chan_sip.c: Merged revisions 113928 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113928 | mmichelson | 2008-04-09 15:56:14 -0500 (Wed, 09 Apr 2008) | 16 lines Merged revisions 113927 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113927 | mmichelson | 2008-04-09 15:54:31 -0500 (Wed, 09 Apr 2008) | 8 lines We need to set the persistant_route [sic] parameter for the sip_pvt during the initial INVITE, no matter if we're building the route set from an INVITE request or response. (closes issue #12391) Reported by: benjaminbohlmann Tested by: benjaminbohlmann ........ ................ 2008-04-09 19:02 +0000 [r113876] Tilghman Lesher * cdr/cdr_csv.c, /, configs/cdr.conf.sample: Merged revisions 113875 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113875 | tilghman | 2008-04-09 14:00:40 -0500 (Wed, 09 Apr 2008) | 12 lines Merged revisions 113874 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113874 | tilghman | 2008-04-09 13:57:33 -0500 (Wed, 09 Apr 2008) | 4 lines If the [csv] section does not exist in cdr.conf, then an unload/load sequence is needed to correct the problem. Track whether the load succeeded with a variable, so we can fix this with a simple reload event, instead. ........ ................ 2008-04-09 17:56 +0000 [r113839] Jason Parker * /, contrib/scripts/astcli: Merged revisions 113838 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r113838 | qwell | 2008-04-09 12:56:07 -0500 (Wed, 09 Apr 2008) | 2 lines Fix a small file handle "leak" pointed out by jjshoe on #asterisk. ........ 2008-04-09 17:50 +0000 [r113837] Mark Michelson * main/pbx.c, /: Merged revisions 113836 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r113836 | mmichelson | 2008-04-09 12:48:33 -0500 (Wed, 09 Apr 2008) | 14 lines There was a subtle logical difference between 1.4 and trunk with regards to how timeouts were handled. In 1.4, if the absolute timeout were reached on a call, no matter what the return value of ast_spawn_extension was, the pbx would attempt to go to the 'T' extension or hangup otherwise. The rearrangement of this function in trunk made this check only happen in the case that ast_spawn_extension returned 0. If ast_spawn_extension returned 1, then the fact that the timeout expired resulted in a no-op, and would cause an infinite loop to occur in __ast_pbx_run. This change fixes this problem. Now timeouts will behave as they did in 1.4 (closes issue #11550) Reported by: pj Tested by: putnopvut ........ 2008-04-09 16:53 +0000 [r113786] Joshua Colp * /, channels/chan_iax2.c: Merged revisions 113785 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113785 | file | 2008-04-09 13:52:04 -0300 (Wed, 09 Apr 2008) | 12 lines Merged revisions 113784 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113784 | file | 2008-04-09 13:50:45 -0300 (Wed, 09 Apr 2008) | 4 lines If we receive an AUTHREQ from the remote server and we are unable to reply (for example they have a secret configured, but we do not) then queue a hangup frame on the Asterisk channel. This will cause the channel to hangup and a HANGUP to be sent via IAX2 to the remote side which is the proper thing to do in this scenario. (closes issue #12385) Reported by: viraptor ........ ................ 2008-04-09 14:42 +0000 [r113683] Mark Michelson * /, channels/chan_sip.c: Merged revisions 113682 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113682 | mmichelson | 2008-04-09 09:41:58 -0500 (Wed, 09 Apr 2008) | 17 lines Merged revisions 113681 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113681 | mmichelson | 2008-04-09 09:40:05 -0500 (Wed, 09 Apr 2008) | 9 lines If Asterisk receives a 488 on an INVITE (not a reinvite), then we should not send a BYE. (closes issue #12392) Reported by: fnordian Patches: chan_sip.patch uploaded by fnordian (license 110) with small modification from me ........ ................ 2008-04-09 13:56 +0000 [r113648-113650] Tilghman Lesher * /, contrib/scripts/astcli: Merged revisions 113647 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r113647 | tilghman | 2008-04-09 08:23:44 -0500 (Wed, 09 Apr 2008) | 6 lines Additional enhancements (closes issue #12390) Reported by: tzafrir Patches: astcli_fixes.diff uploaded by tzafrir (license 46) ........ 2008-04-09 01:40 +0000 [r113598] Terry Wilson * /, channels/chan_iax2.c: Merged revisions 113597 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113597 | twilson | 2008-04-08 20:36:58 -0500 (Tue, 08 Apr 2008) | 10 lines Merged revisions 113596 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113596 | twilson | 2008-04-08 20:34:25 -0500 (Tue, 08 Apr 2008) | 2 lines Initialize fr->cacheable to make valgrind happy ........ ................ 2008-04-08 21:34 +0000 [r113560] Tilghman Lesher * /, contrib/scripts/astcli (added): Merged revisions 113559 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r113559 | tilghman | 2008-04-08 16:33:11 -0500 (Tue, 08 Apr 2008) | 6 lines Add commandline tool for doing CLI commands through AMI (instead of using asterisk -rx) (closes issue #12389) Reported by: davevg Patches: astcli uploaded by davevg (license 209) ........ 2008-04-08 18:49 +0000 [r113404-113506] Jason Parker * /, channels/chan_skinny.c: Merged revisions 113505 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113505 | qwell | 2008-04-08 13:49:21 -0500 (Tue, 08 Apr 2008) | 9 lines Merged revisions 113504 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113504 | qwell | 2008-04-08 13:48:55 -0500 (Tue, 08 Apr 2008) | 1 line Add a little more that is required for previously added devices. ........ ................ * /, channels/chan_skinny.c: Merged revisions 113455 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113455 | qwell | 2008-04-08 13:08:35 -0500 (Tue, 08 Apr 2008) | 12 lines Merged revisions 113454 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113454 | qwell | 2008-04-08 13:07:49 -0500 (Tue, 08 Apr 2008) | 4 lines Add support for several new(ish) devices - most notably, 7942/7945, 7962/7965, 7975. Thanks to Greg Oliver for providing me the required information. ........ ................ * main/asterisk.c, /: Merged revisions 113403 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113403 | qwell | 2008-04-08 12:00:55 -0500 (Tue, 08 Apr 2008) | 9 lines Merged revisions 113402 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113402 | qwell | 2008-04-08 11:56:52 -0500 (Tue, 08 Apr 2008) | 1 line Work around some silliness caused by sys/capability.h - this should fix compile errors a number of users have been experiencing. ........ ................ 2008-04-08 16:56 +0000 [r113350-113401] Tilghman Lesher * /, contrib/scripts/astgenkey.8: Merged revisions 113400 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113400 | tilghman | 2008-04-08 11:54:21 -0500 (Tue, 08 Apr 2008) | 14 lines Merged revisions 113399 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113399 | tilghman | 2008-04-08 11:51:28 -0500 (Tue, 08 Apr 2008) | 6 lines Add security note on astgenkey's manpage. (closes issue #12373) Reported by: lmamane Patches: 20080406__bug12373.diff.txt uploaded by Corydon76 (license 14) ........ ................ * /, channels/chan_sip.c: Merged revisions 113349 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113349 | tilghman | 2008-04-08 10:48:58 -0500 (Tue, 08 Apr 2008) | 15 lines Merged revisions 113348 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113348 | tilghman | 2008-04-08 10:39:16 -0500 (Tue, 08 Apr 2008) | 7 lines Move check for still-bridged channels out a little further, to avoid possible deadlocks. (Closes issue #12252) Reported by: callguy Patches: 20080319__bug12252.diff.txt uploaded by Corydon76 (license 14) Tested by: callguy ........ ................ 2008-04-08 15:10 +0000 [r113298-113299] Joshua Colp * /, main/audiohook.c: Merged revisions 113297 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113297 | file | 2008-04-08 12:05:35 -0300 (Tue, 08 Apr 2008) | 12 lines Merged revisions 113296 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113296 | file | 2008-04-08 12:03:43 -0300 (Tue, 08 Apr 2008) | 4 lines If audio suddenly gets fed into one side of a channel after a lapse of frames flush the other factory so that old audio does not remain in the factory causing the sync code to not execute. (closes issue #12296) Reported by: jvandal ........ ................ 2008-04-07 22:17 +0000 [r113246] Tilghman Lesher * /, configs/manager.conf.sample: Merged revisions 113245 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r113245 | tilghman | 2008-04-07 17:16:46 -0500 (Mon, 07 Apr 2008) | 2 lines Additional note ........ 2008-04-07 21:49 +0000 [r113244] Jason Parker * /, configs/manager.conf.sample: Merged revisions 113243 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r113243 | qwell | 2008-04-07 16:49:27 -0500 (Mon, 07 Apr 2008) | 1 line Document 'originate' permission in manager sample config. ........ 2008-04-07 21:36 +0000 [r113242] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 113241 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113241 | jpeeler | 2008-04-07 16:35:48 -0500 (Mon, 07 Apr 2008) | 23 lines Merged revisions 113013 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113013 | jpeeler | 2008-04-07 10:18:10 -0500 (Mon, 07 Apr 2008) | 15 lines Merged revisions 113012 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008) | 7 lines (closes issue #12362) (closes issue #12372) Reported by: vinsik Tested by: tecnoxarxa This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one. ........ ................ ................ 2008-04-07 19:10 +0000 [r113174] Jason Parker * /, channels/chan_skinny.c, configs/skinny.conf.sample: Merged revisions 113119 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113119 | qwell | 2008-04-07 13:02:51 -0500 (Mon, 07 Apr 2008) | 16 lines Merged revisions 113118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113118 | qwell | 2008-04-07 13:00:09 -0500 (Mon, 07 Apr 2008) | 8 lines Allow playback with noanswer (and add earlyrtp option). (closes issue #9077) Reported by: pj Patches: earlyrtp.diff uploaded by wedhorn (license 30) Tested by: pj, qwell, DEA, wedhorn ........ ................ 2008-04-07 19:08 +0000 [r113173] Tilghman Lesher * /, funcs/func_strings.c: Merged revisions 113172 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113172 | tilghman | 2008-04-07 14:06:46 -0500 (Mon, 07 Apr 2008) | 11 lines Merged revisions 113117 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113117 | tilghman | 2008-04-07 12:51:49 -0500 (Mon, 07 Apr 2008) | 3 lines Force ast_mktime() to check for DST, since strptime(3) does not. (Closes issue #12374) ........ ................ 2008-04-07 16:13 +0000 [r113067] Mark Michelson * main/channel.c, /: Merged revisions 113066 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113066 | mmichelson | 2008-04-07 11:12:30 -0500 (Mon, 07 Apr 2008) | 21 lines Merged revisions 113065 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113065 | mmichelson | 2008-04-07 11:08:45 -0500 (Mon, 07 Apr 2008) | 13 lines This fix prevents a deadlock that was experienced in chan_local. There was deadlock prevention in place in chan_local, but it would not work in a specific case because the channel was recursively locked. By unlocking the channel prior to calling the generator's generate callback in ast_read_generator_actions(), we prevent the recursive locking, and therefore the deadlock. (closes issue #12307) Reported by: callguy Patches: 12307.patch uploaded by putnopvut (license 60) Tested by: callguy ........ ................ 2008-04-07 15:28 +0000 [r113042] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 113013 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113013 | jpeeler | 2008-04-07 10:18:10 -0500 (Mon, 07 Apr 2008) | 15 lines Merged revisions 113012 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008) | 7 lines (closes issue #12362) (closes issue #12372) Reported by: vinsik Tested by: tecnoxarxa This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one. ........ ................ 2008-04-05 13:30 +0000 [r112973-112975] Tilghman Lesher * /, res/res_agi.c: Merged revisions 112972 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r112972 | tilghman | 2008-04-05 08:24:12 -0500 (Sat, 05 Apr 2008) | 6 lines AsyncAGI should not close the manager session on error. (closes issue #12370) Reported by: srt Patches: asterisk-12370.diff uploaded by srt (license 378) ........ 2008-04-04 19:30 +0000 [r112786-112822] Philippe Sultan * /, channels/chan_gtalk.c: Merged revisions 112821 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112821 | phsultan | 2008-04-04 21:28:49 +0200 (Fri, 04 Apr 2008) | 9 lines Merged revisions 112820 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112820 | phsultan | 2008-04-04 21:26:15 +0200 (Fri, 04 Apr 2008) | 1 line Free newly allocated channel before returning ........ ................ * /, channels/chan_gtalk.c: Merged revisions 112785 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112785 | phsultan | 2008-04-04 19:32:46 +0200 (Fri, 04 Apr 2008) | 15 lines Merged revisions 112766 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112766 | phsultan | 2008-04-04 19:16:59 +0200 (Fri, 04 Apr 2008) | 7 lines Prevent call connections when codecs don't match. (closes issue #10604) Reported by: keepitcool Patches: branch-1.4-10604-2.diff uploaded by phsultan (license 73) Tested by: phsultan ........ ................ 2008-04-04 01:08 +0000 [r112715] Dwayne M. Hubbard * main/asterisk.c, /: Merged revisions 112653,112656,112714 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r112653 | dhubbard | 2008-04-03 17:13:11 -0500 (Thu, 03 Apr 2008) | 1 line add a Zaptel timer check to verify the timer is responding when Zaptel support is compiled into Asterisk and Zaptel drivers are loaded. This will help people not waste their valuable time debugging side effects. ........ r112656 | dhubbard | 2008-04-03 17:19:43 -0500 (Thu, 03 Apr 2008) | 1 line satisfy buildbot ........ r112714 | dhubbard | 2008-04-03 19:57:33 -0500 (Thu, 03 Apr 2008) | 1 line sleep long enough for the zaptel timer error message to display before exit ........ 2008-04-04 00:54 +0000 [r112713] Joshua Colp * /, main/Makefile: Merged revisions 112712 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112712 | file | 2008-04-03 21:53:19 -0300 (Thu, 03 Apr 2008) | 10 lines Merged revisions 112711 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112711 | file | 2008-04-03 21:52:36 -0300 (Thu, 03 Apr 2008) | 2 lines Pass in the path to Zaptel for systems that install Zaptel headers in a separate location. ........ ................ 2008-04-03 14:42 +0000 [r112601] Mark Michelson * channels/chan_zap.c, /: Merged revisions 112600 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112600 | mmichelson | 2008-04-03 09:35:47 -0500 (Thu, 03 Apr 2008) | 17 lines Merged revisions 112599 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112599 | mmichelson | 2008-04-03 09:32:20 -0500 (Thu, 03 Apr 2008) | 9 lines Fix the testing of the "res" variable so that it is more logically correct and makes the correct warning and debug messages print. (closes issue #12361) Reported by: one47 Patches: chan_zap_deferred_digit.patch uploaded by one47 (license 23) ........ ................ 2008-04-02 17:37 +0000 [r112470] Mark Michelson * main/manager.c, /: Merged revisions 112469 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112469 | mmichelson | 2008-04-02 12:36:49 -0500 (Wed, 02 Apr 2008) | 21 lines Merged revisions 112468 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112468 | mmichelson | 2008-04-02 12:36:04 -0500 (Wed, 02 Apr 2008) | 13 lines Fix a race condition in the manager. It is possible that a new manager event could be appended during a brief time when the manager is not waiting for input. If an event comes during this period, we need to set an indicator that there is an event pending so that the manager doesn't attempt to wait forever for an event that already happened. (closes issue #12354) Reported by: bamby Patches: manager_race_condition.diff uploaded by bamby (license 430) (comments added by me) ........ ................ 2008-04-02 15:27 +0000 [r112436] Joshua Colp * /, channels/chan_sip.c: Merged revisions 112431 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r112431 | file | 2008-04-02 12:26:51 -0300 (Wed, 02 Apr 2008) | 7 lines Since the SIP request structure gets reused multiple times with TCP handling we have to clear the debug state or else we will keep spitting out debug even after it has been turned off. (closes issue #12169) Reported by: pj Patches: 12169-debugoff-2.diff uploaded by qwell (license 4) Tested by: pj ........ 2008-04-02 14:33 +0000 [r112395] Mark Michelson * /, apps/app_queue.c: Merged revisions 112394 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112394 | mmichelson | 2008-04-02 09:32:43 -0500 (Wed, 02 Apr 2008) | 14 lines Merged revisions 112393 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112393 | mmichelson | 2008-04-02 09:32:00 -0500 (Wed, 02 Apr 2008) | 6 lines Ensure that there is no timeout if none is specified. (closes issue #12349) Reported by: johnlange ........ ................ 2008-04-01 22:48 +0000 [r112359] Steve Murphy * main/pbx.c, /: Merged revisions 112357 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r112357 | murf | 2008-04-01 16:45:10 -0600 (Tue, 01 Apr 2008) | 1 line Bumped across another test set for the new exten pattern matcher, which revealed a problem with the CANMATCH/MATCHMORE modes. Direct matches were getting in the way. Fixed. ........ 2008-04-01 20:20 +0000 [r112299] Steve Murphy * main/pbx.c, /: Merged revisions 112289 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r112289 | murf | 2008-04-01 14:02:19 -0600 (Tue, 01 Apr 2008) | 21 lines (closes issue #12298) Reported by: falves11 Patches: 12298.patch1 uploaded by murf (license 17) Tested by: murf I have hopes that the changes made over the last few days will finalize and solidify this code. While there are bound to be small tweaks still needed, I feel that the job (at last) is somewhat completed. Finally, I had a chance to comprehend how the scoring of extension patterns was done in the previous version, and I've come very close to using the exact same criteria in the new pattern matching code. The left-right sorting is now replicated in the trie structure itself, such that the first match found will the 'best' match. Compared the results against 1.4 for several extensions. Replicated falves11's setup and it works. Used some devious patterns provided by jsmith, supplemented with a few of my own. Looks good. ........ 2008-04-01 18:09 +0000 [r112211] Joshua Colp * main/rtp.c, /: Merged revisions 112210 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112210 | file | 2008-04-01 15:06:13 -0300 (Tue, 01 Apr 2008) | 12 lines Merged revisions 112209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112209 | file | 2008-04-01 15:02:43 -0300 (Tue, 01 Apr 2008) | 4 lines Disable Packet2Packet bridging when we need to feed DTMF frames into the core. Some implementations do not like how we switch between things. (closes issue #12212) Reported by: bamby ........ ................ 2008-04-01 17:52 +0000 [r112170-112206] Joshua Colp * /, channels/chan_sip.c: Merged revisions 112205 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112205 | file | 2008-04-01 14:48:52 -0300 (Tue, 01 Apr 2008) | 12 lines Merged revisions 112204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112204 | file | 2008-04-01 14:43:46 -0300 (Tue, 01 Apr 2008) | 4 lines Do not pass audio until the remote side has indicated they are providing early media, or if the channel has been answered. (closes issue #11823) Reported by: SDamm ........ ................ 2008-04-01 17:25 +0000 [r112157] Mark Michelson * main/dns.c, /: Merged revisions 112148 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112148 | mmichelson | 2008-04-01 12:23:19 -0500 (Tue, 01 Apr 2008) | 18 lines Merged revisions 112138 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112138 | mmichelson | 2008-04-01 12:21:21 -0500 (Tue, 01 Apr 2008) | 10 lines Initialize the __res_state structure used for dns purposes to all 0's prior to using it. This is due to valgrind's complaints on issue #12284 as well as an excerpt found in "Description" portion of the online man page found here: http://www.iti.cs.tu-bs.de/cgi-bin/UNIXhelp/man-cgi?res_nquery+3RESOLV (pertains to issue #12284 but does not necessarily close it) ........ ................ 2008-04-01 16:57 +0000 [r112127] Joshua Colp * include/asterisk/slinfactory.h, /, main/slinfactory.c: Merged revisions 112126 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112126 | file | 2008-04-01 13:50:37 -0300 (Tue, 01 Apr 2008) | 13 lines Merged revisions 112125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112125 | file | 2008-04-01 13:45:14 -0300 (Tue, 01 Apr 2008) | 5 lines Ensure that we do not exceed the hold's maximum size with a single frame. (closes issue #12047) Reported by: fabianoheringer Tested by: fabianoheringer ........ ................ 2008-03-31 22:17 +0000 [r112070-112072] Jason Parker * apps/app_voicemail.c, /: Merged revisions 112069 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112069 | qwell | 2008-03-31 16:48:30 -0500 (Mon, 31 Mar 2008) | 13 lines Merged revisions 112068 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112068 | qwell | 2008-03-31 16:48:05 -0500 (Mon, 31 Mar 2008) | 5 lines Fix a silly infinite loop when choosing an invalid option. (closes issue #12315) Reported by: jmls ........ ................ 2008-03-31 21:03 +0000 [r112034-112036] Terry Wilson * /, main/http.c: Merged revisions 112033 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r112033 | twilson | 2008-03-31 15:45:05 -0500 (Mon, 31 Mar 2008) | 2 lines Handle blank prefix= in http.conf ........ 2008-03-31 17:15 +0000 [r111997-111999] Russell Bryant * Makefile, /: Merged revisions 111998 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r111998 | russell | 2008-03-31 12:14:58 -0500 (Mon, 31 Mar 2008) | 7 lines Ensure configure gets run on a clean checkout. (closes issue #12197) Reported by: juggie Patches: 12197.diff uploaded by juggie (license 24) ........ 2008-03-31 14:22 +0000 [r111962] Joshua Colp * res/res_config_sqlite.c, /: Merged revisions 111961 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r111961 | file | 2008-03-31 11:20:39 -0300 (Mon, 31 Mar 2008) | 4 lines Initialize all these here tmp pointers at declaration. They confused some compilers a wee bit. (closes issue #12333) Reported by: ovi ........ 2008-03-29 Russell Bryant * Asterisk 1.6.0-beta7.1 released. Asterisk 1.6.0-beta7 was tagged against trunk, instead of the 1.6.0 branch. 2008-03-28 21:46 +0000 [r111858] Jason Parker * codecs/gsm/inc/private.h, /: Merged revisions 111857 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111857 | qwell | 2008-03-28 16:46:02 -0500 (Fri, 28 Mar 2008) | 20 lines Merged revisions 111856 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111856 | qwell | 2008-03-28 16:45:35 -0500 (Fri, 28 Mar 2008) | 12 lines Allow gsm to compile correctly on x86 with gcc4 optimizations. (closes issue #11243) Reported by: whiskerp Patches: 11243-maybe-asm.diff uploaded by qwell (license 4) Tested by: Seggy (IRC) Note: While I did write this patch, I would not have found this if fossil had not reported and fixed issue #12253. A huge thanks to him for helping to (indirectly) find the problem here. ........ ................ 2008-03-28 19:11 +0000 [r111722-111776] Jason Parker * /, channels/chan_skinny.c: Merged revisions 111721 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111721 | qwell | 2008-03-28 12:57:12 -0500 (Fri, 28 Mar 2008) | 9 lines Merged revisions 111720 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111720 | qwell | 2008-03-28 12:55:05 -0500 (Fri, 28 Mar 2008) | 1 line Remove unimplemented softkeys. Prompted by issue #12325. ........ ................ 2008-03-28 16:21 +0000 [r111660] Jason Parker * /, formats/format_wav_gsm.c: Merged revisions 111659 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111659 | qwell | 2008-03-28 11:20:59 -0500 (Fri, 28 Mar 2008) | 16 lines Merged revisions 111658 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111658 | qwell | 2008-03-28 11:19:56 -0500 (Fri, 28 Mar 2008) | 8 lines The file size of WAV49 does not need to be an even number. (closes issue #12128) Reported by: mdu113 Patches: 12128-noevenlength.diff uploaded by qwell (license 4) Tested by: qwell, mdu113 ........ ................ 2008-03-28 14:43 +0000 [r111607-111608] Tilghman Lesher * doc/valgrind.txt, /: Merged revisions 111606 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111606 | tilghman | 2008-03-28 09:37:28 -0500 (Fri, 28 Mar 2008) | 11 lines Merged revisions 111605 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111605 | tilghman | 2008-03-28 09:35:45 -0500 (Fri, 28 Mar 2008) | 3 lines Update debugging text, since Valgrind eliminated the --log-file-exactly option. (Closes issue #12320) ........ ................ 2008-03-28 00:56 +0000 [r111566] Joshua Colp * /, apps/app_queue.c: Merged revisions 111565 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r111565 | file | 2008-03-27 21:55:47 -0300 (Thu, 27 Mar 2008) | 2 lines Forgetting to unregister a manager action is bad, mmmk? ........ 2008-03-28 00:17 +0000 [r111534] Mark Michelson * /, apps/app_queue.c: Merged revisions 111533 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r111533 | mmichelson | 2008-03-27 19:12:52 -0500 (Thu, 27 Mar 2008) | 10 lines Fix a crash that would happen when attempting to unload the app_queue module. The problem was that when the refcount on the queue hit 0, the destructor was called, and inside the destructor, another function was called which would increase the refcount back to 1 again and then decrease it again back to 0 for every member in the queue. This meant that the destructor was being recursively called, leading to a double free of the queue. This is now fixed by making sure to unlink the queue from the queues container prior to the final unref of the queue. ........ 2008-03-27 21:28 +0000 [r111498] Steve Murphy * main/pbx.c, /: Merged revisions 111497 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r111497 | murf | 2008-03-27 15:25:55 -0600 (Thu, 27 Mar 2008) | 1 line comment cleanup and iron out a really dumb mistake in handling the '.'-wildcard in the new exten pattern matcher. ........ 2008-03-27 19:30 +0000 [r111444] Tilghman Lesher * /, main/acl.c: Merged revisions 111443 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111443 | tilghman | 2008-03-27 14:26:45 -0500 (Thu, 27 Mar 2008) | 14 lines Merged revisions 111442 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111442 | tilghman | 2008-03-27 14:23:12 -0500 (Thu, 27 Mar 2008) | 6 lines For FreeBSD, at least, the ifa_addr element could be NULL. (closes issue #12300) Reported by: festr Patches: acl.c.patch uploaded by festr (license 443) ........ ................ 2008-03-27 13:42 +0000 [r111361-111411] Steve Murphy * apps/app_playback.c, main/pbx.c, /: Merged revisions 111410 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111410 | murf | 2008-03-27 07:29:41 -0600 (Thu, 27 Mar 2008) | 17 lines Merged revisions 111391 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111391 | murf | 2008-03-27 07:03:28 -0600 (Thu, 27 Mar 2008) | 9 lines These small documentation updates made in response to a query in asterisk-users, where a user was using Playback, but needed the features of Background, and had no idea that Background existed, or that it might provide the features he needed. I thought the best way to avert these kinds of queries was to provide "See Also" references in all three of "Background", "Playback", "WaitExten". Perhaps a project to do this with all related apps is in order. ........ ................ * res/ael/pval.c, /, include/asterisk/pval.h, pbx/pbx_ael.c, include/asterisk/ael_structs.h: Merged revisions 111360 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111360 | murf | 2008-03-26 22:47:12 -0600 (Wed, 26 Mar 2008) | 23 lines Merged revisions 111341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111341 | murf | 2008-03-26 21:21:05 -0600 (Wed, 26 Mar 2008) | 15 lines (closes issue #12302) Reported by: pj Tested by: murf These changes will set a channel variable ~~EXTEN~~ just before generating code for a switch, with the value of ${EXTEN}. The exten is marked as having a switch, and ever after that, till the end of the exten, we substitute any ${EXTEN} with ${~~EXTEN~~} instead in application arguments; (and the ${EXTEN: also). The reason for this, is that because switches are coded using separate extensions to provide pattern matching, and jumping to/from these switch extensions messes up the ${EXTEN} value, which blows the minds of users. ........ ................ 2008-03-27 00:36 +0000 [r111247-111339] Jason Parker * main/frame.c, /: Merged revisions 111285 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111285 | qwell | 2008-03-26 19:25:56 -0500 (Wed, 26 Mar 2008) | 9 lines Merged revisions 111280 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111280 | qwell | 2008-03-26 19:25:13 -0500 (Wed, 26 Mar 2008) | 1 line Put this flag back so we don't change the API. ........ ................ * main/frame.c, /: Merged revisions 111246 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111246 | qwell | 2008-03-26 18:27:33 -0500 (Wed, 26 Mar 2008) | 17 lines Merged revisions 111245 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111245 | qwell | 2008-03-26 18:26:33 -0500 (Wed, 26 Mar 2008) | 9 lines Remove excessive smoother optimization that was causing audio glitches (small "pops") after (about 200ms later) an "incorrectly" sized frame was received. While it would be very nice to keep this as optimized as possible, it makes no sense for the smoother to be dropping random bits of audio like this. Isn't that the whole point of a smoother? Closes issue #12093. ........ ................ 2008-03-26 19:57 +0000 [r111131] Joshua Colp * contrib/scripts/autosupport, /: Merged revisions 111130 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111130 | file | 2008-03-26 16:56:40 -0300 (Wed, 26 Mar 2008) | 14 lines Merged revisions 111129 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111129 | file | 2008-03-26 16:55:08 -0300 (Wed, 26 Mar 2008) | 6 lines Update autosupport script. (closes issue #12310) Reported by: angler Patches: autosupport.diff uploaded by angler (license 106) ........ ................ 2008-03-26 19:53 +0000 [r111128] Kevin P. Fleming * /, UPGRADE.txt: Merged revisions 111127 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111127 | kpfleming | 2008-03-26 14:52:27 -0500 (Wed, 26 Mar 2008) | 18 lines Merged revisions 111126 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r111126 | kpfleming | 2008-03-26 14:51:24 -0500 (Wed, 26 Mar 2008) | 10 lines Merged revisions 111125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r111125 | kpfleming | 2008-03-26 14:49:30 -0500 (Wed, 26 Mar 2008) | 2 lines update UPGRADE notes to document usage of the script ........ ................ ................ 2008-03-26 19:41 +0000 [r111124] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 111123 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111123 | mmichelson | 2008-03-26 14:39:23 -0500 (Wed, 26 Mar 2008) | 12 lines Merged revisions 111121 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111121 | mmichelson | 2008-03-26 14:37:36 -0500 (Wed, 26 Mar 2008) | 4 lines This code change is made just for clarification. It does exactly the same thing as before. It just doesn't look as wrong. ........ ................ 2008-03-26 19:27 +0000 [r111072] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 111067 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111067 | mmichelson | 2008-03-26 14:26:23 -0500 (Wed, 26 Mar 2008) | 17 lines Merged revisions 111049 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111049 | mmichelson | 2008-03-26 14:22:16 -0500 (Wed, 26 Mar 2008) | 9 lines Add a lock to the vm_state structure and use the lock around mail_open calls to prevent concurrent access of the same mailstream. This, along with trunk's ability to configure TCP timeouts for IMAP storage will help to prevent crashes and hangs when using voicemail with IMAP storage. (closes issue #10487) Reported by: ewilhelmsen ........ ................ 2008-03-26 19:08 +0000 [r111026] Kevin P. Fleming * codecs/ilbc, /, contrib/scripts/get_ilbc_source.sh (added): Merged revisions 111025 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111025 | kpfleming | 2008-03-26 14:08:00 -0500 (Wed, 26 Mar 2008) | 18 lines Merged revisions 111024 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r111024 | kpfleming | 2008-03-26 14:06:56 -0500 (Wed, 26 Mar 2008) | 10 lines Merged revisions 111019 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r111019 | kpfleming | 2008-03-26 13:58:37 -0500 (Wed, 26 Mar 2008) | 2 lines add a script to make getting the iLBC source code simple for end users ........ ................ ................ 2008-03-26 19:06 +0000 [r111018-111023] Joshua Colp * /, channels/chan_sip.c: Merged revisions 111021 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111021 | file | 2008-03-26 16:05:42 -0300 (Wed, 26 Mar 2008) | 12 lines Merged revisions 111020 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111020 | file | 2008-03-26 16:04:35 -0300 (Wed, 26 Mar 2008) | 4 lines If we are requested to authenticate a reinvite make sure that it contains T38 SDP if need be. (closes issue #11995) Reported by: fall ........ ................ * /, channels/chan_iax2.c: Merged revisions 111017 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111017 | file | 2008-03-26 15:42:52 -0300 (Wed, 26 Mar 2008) | 12 lines Merged revisions 110628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4 lines Add an option (transmit_silence) which transmits silence during both Record() and DTMF generation. The reason this is an option is that in order to transmit silence we have to setup a translation path. This may not be needed/wanted in all cases. (closes issue #10058) Reported by: tracinet ........ ................ 2008-03-26 17:44 +0000 [r110964] Kevin P. Fleming * /, UPGRADE.txt: Merged revisions 110963 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110963 | kpfleming | 2008-03-26 12:44:09 -0500 (Wed, 26 Mar 2008) | 10 lines Merged revisions 110962 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110962 | kpfleming | 2008-03-26 12:43:02 -0500 (Wed, 26 Mar 2008) | 2 lines add note that the user will need to enable codec_ilbc to get it to build ........ ................ 2008-03-26 17:35 +0000 [r110959] Donny Kavanagh * /, doc/snmp.txt: Merged revisions 110911 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110911 | juggie | 2008-03-26 13:24:54 -0400 (Wed, 26 Mar 2008) | 8 lines update documentation to reflect the changes in the way configure detects net-snmp. (closes issue #12067) Reported by: juggie Patches: 12067_snmp_doc.patch uploaded by juggie (license 24) Tested by: juggie ........ 2008-03-26 17:15 +0000 [r110882] Kevin P. Fleming * codecs/ilbc/constants.h (removed), codecs/ilbc/iLBC_decode.h (removed), codecs/ilbc/iCBSearch.c (removed), codecs/Makefile, codecs/ilbc/filter.c (removed), codecs/ilbc/hpInput.c (removed), codecs/ilbc/gainquant.c (removed), codecs/ilbc/hpOutput.c (removed), codecs/ilbc/iCBSearch.h (removed), codecs/ilbc/filter.h (removed), codecs/ilbc/hpInput.h (removed), codecs/ilbc/gainquant.h (removed), codecs/ilbc/LPCencode.c (removed), codecs/ilbc/hpOutput.h (removed), codecs/ilbc/StateSearchW.c (removed), codecs/codec_ilbc.c, codecs/ilbc/LPCencode.h (removed), codecs/ilbc/iCBConstruct.c (removed), codecs/ilbc/StateSearchW.h (removed), codecs/ilbc/syntFilter.c (removed), /, codecs/ilbc/iCBConstruct.h (removed), codecs/ilbc/syntFilter.h (removed), codecs/ilbc/packing.c (removed), codecs/ilbc/StateConstructW.c (removed), codecs/ilbc/packing.h (removed), codecs/ilbc/libilbc.vcproj (removed), codecs/ilbc/StateConstructW.h (removed), codecs/ilbc/LPCdecode.c (removed), codecs/ilbc/getCBvec.c (removed), codecs/ilbc/enhancer.c (removed), codecs/ilbc/lsf.c (removed), codecs/ilbc/iLBC_encode.c (removed), codecs/ilbc/getCBvec.h (removed), codecs/ilbc/LPCdecode.h (removed), codecs/ilbc/enhancer.h (removed), codecs/ilbc/FrameClassify.c (removed), codecs/ilbc/iLBC_define.h (removed), codecs/ilbc/lsf.h (removed), codecs/ilbc/iLBC_encode.h (removed), codecs/ilbc/FrameClassify.h (removed), codecs/ilbc/helpfun.c (removed), codecs/ilbc/doCPLC.c (removed), codecs/ilbc/anaFilter.c (removed), codecs/ilbc/helpfun.h (removed), codecs/ilbc/createCB.c (removed), codecs/ilbc/doCPLC.h (removed), codecs/ilbc/anaFilter.h (removed), UPGRADE.txt, codecs/ilbc/constants.c (removed), codecs/ilbc/iLBC_decode.c (removed), codecs/ilbc/createCB.h (removed), CHANGES: Merged revisions 110881 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110881 | kpfleming | 2008-03-26 10:10:28 -0700 (Wed, 26 Mar 2008) | 18 lines Merged revisions 110880 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r110880 | kpfleming | 2008-03-26 09:42:35 -0700 (Wed, 26 Mar 2008) | 10 lines Merged revisions 110869 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r110869 | kpfleming | 2008-03-26 08:53:46 -0700 (Wed, 26 Mar 2008) | 2 lines due to licensing restrictions, we cannot distribute the source code for iLBC encoding and decoding... so remove it, and add instructions on how the user can obtain it themselves ........ ................ ................ 2008-03-26 15:33 +0000 [r110866-110868] Joshua Colp * /: Merged revisions 110726 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110726 | jpeeler | 2008-03-25 17:02:57 -0300 (Tue, 25 Mar 2008) | 2 lines This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one. ........ 2008-03-26 00:03 +0000 [r110832] Mark Michelson * main/manager.c, /: Merged revisions 110831 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110831 | mmichelson | 2008-03-25 19:02:31 -0500 (Tue, 25 Mar 2008) | 6 lines This ensures that the manager interface is not enabled by default. Prior to this change, it was possible to start Asterisk with the manager interface enabled, then either comment out the enabled option or make manager.conf unopenable and the manager interface would still be enabled. ........ 2008-03-25 22:52 +0000 [r110781] Jason Parker * cdr/cdr_custom.c, /: Merged revisions 110780 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110780 | qwell | 2008-03-25 17:51:55 -0500 (Tue, 25 Mar 2008) | 14 lines Merged revisions 110779 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110779 | qwell | 2008-03-25 17:51:17 -0500 (Tue, 25 Mar 2008) | 6 lines Make file access in cdr_custom similar to cdr_csv. Fixes issue #12268. Patch borrowed from r82344 ........ ................ 2008-03-25 22:11 +0000 [r110778] Jeff Peeler * channels/chan_sip.c: This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one. 2008-03-25 17:47 +0000 [r110690-110692] Tilghman Lesher * configs/extensions.conf.sample, /, configs/voicemail.conf.sample: Merged revisions 110691 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110691 | tilghman | 2008-03-25 12:46:34 -0500 (Tue, 25 Mar 2008) | 6 lines Update sample configurations to make virtual hosting more obvious. (closes issue #11969) Reported by: pprindeville Patches: acme-virtualpbx.1.6.patch uploaded by pprindeville (license 347) ........ * configs/extensions.conf.sample, /: Merged revisions 110689 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110689 | tilghman | 2008-03-25 12:40:28 -0500 (Tue, 25 Mar 2008) | 6 lines Update the sample configuration, to use Macro less (since it's now deprecated). (closes issue #12293) Reported by: pprindeville Patches: bugid-0012293.1.6.patch uploaded by pprindeville (license 347) ........ 2008-03-25 15:43 +0000 [r110637-110638] Mark Michelson * channels/chan_sip.c: Oops. * /, channels/chan_sip.c: Merged revisions 110636 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110636 | mmichelson | 2008-03-25 10:41:33 -0500 (Tue, 25 Mar 2008) | 15 lines Merged revisions 110635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110635 | mmichelson | 2008-03-25 10:40:33 -0500 (Tue, 25 Mar 2008) | 7 lines When reverting a commit, I accidentally left in this bit which was an experiment to see what would happen. It passed the compile test, and I didn't notice I had left this change in too. So this is a revert of a revert...sort of. ........ ................ 2008-03-25 15:39 +0000 [r110630-110634] Joshua Colp * include/asterisk/options.h, main/asterisk.c, Makefile, /, main/app.c: Merged revisions 110629 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110629 | file | 2008-03-25 11:39:45 -0300 (Tue, 25 Mar 2008) | 12 lines Merged revisions 110628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4 lines Add an option (transmit_silence) which transmits silence during both Record() and DTMF generation. The reason this is an option is that in order to transmit silence we have to setup a translation path. This may not be needed/wanted in all cases. (closes issue #10058) Reported by: tracinet ........ ................ 2008-03-24 20:14 +0000 [r110620-110622] Mark Michelson * /, channels/chan_sip.c: Merged revisions 110619 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110619 | mmichelson | 2008-03-24 14:19:37 -0500 (Mon, 24 Mar 2008) | 23 lines Merged revisions 110618 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110618 | mmichelson | 2008-03-24 14:17:41 -0500 (Mon, 24 Mar 2008) | 15 lines This is a revert for revision 108288. The reason is that that revision was not for an actual bug fix per se, and so it really should not have been in 1.4 in the first place. Plus, people who compile with DO_CRASH are more likely to encounter a crash due to this change. While I think the usage of DO_CRASH in ast_sched_del is a bit absurd, this sort of change is beyond the scope of 1.4 and should be done instead in a developer branch based on trunk so that all scheduler functions are fixed at once. I also am reverting the change to trunk and 1.6 since they also suffer from the DO_CRASH potential. (closes issue #12272) Reported by: qq12345 ........ ................ 2008-03-24 17:36 +0000 [r110616] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 110615 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110615 | russell | 2008-03-24 12:36:04 -0500 (Mon, 24 Mar 2008) | 10 lines Merged revisions 110614 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110614 | russell | 2008-03-24 12:34:56 -0500 (Mon, 24 Mar 2008) | 2 lines Turn a NOTICE into a DEBUG message. ........ ................ 2008-03-24 15:29 +0000 [r110611] Joshua Colp * /, channels/chan_sip.c: Merged revisions 110610 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110610 | file | 2008-03-24 12:28:25 -0300 (Mon, 24 Mar 2008) | 6 lines Only print out the set_address_from_contact host verbose message if debugging is enabled on the dialog. (closes issue #12280) Reported by: rjain Patches: chan_sip.c.diff uploaded by rjain (license 226) ........ 2008-03-21 21:52 +0000 [r110579] Jason Parker * /, sounds/Makefile: Merged revisions 110578 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110578 | qwell | 2008-03-21 16:52:06 -0500 (Fri, 21 Mar 2008) | 1 line Update to 1.4.11 core sounds. ........ 2008-03-21 15:25 +0000 [r110501] Russell Bryant * /, configs/sip.conf.sample, CHANGES: Merged revisions 110499 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110499 | russell | 2008-03-21 10:24:43 -0500 (Fri, 21 Mar 2008) | 3 lines Note that the TCP and TLS support is currently considered experimental and is subject to change while we work out the remaining issues. ........ 2008-03-21 14:36 +0000 [r110476] Jason Parker * /, codecs/gsm/Makefile: Merged revisions 110475 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110475 | qwell | 2008-03-21 09:36:17 -0500 (Fri, 21 Mar 2008) | 15 lines Merged revisions 110474 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110474 | qwell | 2008-03-21 09:32:52 -0500 (Fri, 21 Mar 2008) | 7 lines Don't attempt to do optimizations of gsm on mips platforms either. (closes issue #12270) Reported by: zandbelt Patches: 026-gsm-mips.patch uploaded by zandbelt (license 33) ........ ................ 2008-03-20 23:14 +0000 [r110304-110397] Russell Bryant * main/autoservice.c, /: Merged revisions 110396 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110396 | russell | 2008-03-20 18:14:13 -0500 (Thu, 20 Mar 2008) | 17 lines Merged revisions 110395 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110395 | russell | 2008-03-20 18:13:56 -0500 (Thu, 20 Mar 2008) | 9 lines Shorten the ast_waitfor() timeout from 500 ms to 50 ms in the autoservice thread. This really should not make a difference except in very rare cases. That case would be that all of the channels in autoservice are not generating any frames. In that case, this change reduces the potential amount of time that a thread waits in ast_autoservice_stop() for the autoservice thread to wrap back around to the beginning of its loop. (closes issue #12266, reported by dimas) ........ ................ * codecs/codec_g722.c, /: Merged revisions 110339 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110339 | russell | 2008-03-20 17:02:20 -0500 (Thu, 20 Mar 2008) | 3 lines Use the correct buffer for g722tolin16_sample. This shouldn't have caused any problems, but Qwell noticed the typo here. ........ * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions 110337 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110337 | russell | 2008-03-20 16:55:50 -0500 (Thu, 20 Mar 2008) | 22 lines Merged revisions 110336 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r110336 | russell | 2008-03-20 16:54:58 -0500 (Thu, 20 Mar 2008) | 14 lines Merged revisions 110335 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r110335 | russell | 2008-03-20 16:53:27 -0500 (Thu, 20 Mar 2008) | 6 lines Fix some very broken code that was introduced in 1.2.26 as a part of the security fix. The dnsmgr is not appropriate here. The dnsmgr takes a pointer to an address structure that a background thread continuously updates. However, in these cases, a stack variable was passed. That means that the dnsmgr thread would be continuously writing to bogus memory. ........ ................ ................ * /, main/file.c: Merged revisions 110303 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110303 | russell | 2008-03-20 15:08:26 -0500 (Thu, 20 Mar 2008) | 8 lines Fix a bug when using zaptel timing for playing back files that have a sample rate other than 8 kHz. The issue here is that format modules give a "whennext" sample value, which is used to calculate when to set a timer for to retrieve the next frame. However, the zaptel timer operates on 8 kHz samples, so this must be taken into account. (another part of issue #12164, reported by milazzo and jsmith, patch by me) ........ 2008-03-20 18:02 +0000 [r110273] Mark Michelson * main/dial.c, /: Merged revisions 110272 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110272 | mmichelson | 2008-03-20 13:01:36 -0500 (Thu, 20 Mar 2008) | 3 lines Add missing unlock ........ 2008-03-20 17:45 +0000 [r110269-110271] Russell Bryant * main/channel.c, /, res/res_musiconhold.c: Merged revisions 110268 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110268 | russell | 2008-03-20 12:41:22 -0500 (Thu, 20 Mar 2008) | 27 lines Add some fixes that I made in regards to wideband codec handling to get G.722 music on hold working for me. (issue #12164, reported by milazzo and jsmith, patches by me) res/res_musiconhold.c: - I moved a single line so that the sample queue update happened before ast_write(). The reason that this was a bug is that the G.722 frame originally says it has 320 samples in it (which is correct). However, when the frame is written to a channel that uses RTP, main/rtp.c modifies the frame to cut the number of samples in half before it sends it on the wire. This is to account for the stupid incorrect G.722 spec that makes it so we have to lie about the number of samples with RTP. I should probably go and re-work the RTP code so it doesn't modify the frame so that a bug like this won't happen in the future. However, this change to MOH is harmless. main/channel.c: - I made two fixes in regards to generator timing. Generators use samples for timing. However, this code assumed 8 kHz samples. In one case, it was a hard coded 160 samples, that is now written as the sample rate / 50. The other place was dealing with timing a generator based on frames coming from the other direction. However, that would have only worked if the sample rates for the formats in both directions were the same. The code now takes into account that the sample rates may differ, and scales the generator samples accordingly. ........ 2008-03-19 23:00 +0000 [r110165] Russell Bryant * /, apps/app_meetme.c: Merged revisions 110164 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110164 | russell | 2008-03-19 17:58:33 -0500 (Wed, 19 Mar 2008) | 13 lines Merged revisions 110163 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110163 | russell | 2008-03-19 17:57:59 -0500 (Wed, 19 Mar 2008) | 5 lines Fix a bug where when calls on the trunk side hang up while on hold, the state is not properly reflected. (closes issue #11990, reported by anakaoka, patched by me) ........ ................ 2008-03-19 21:06 +0000 [r110088] Jeff Peeler * /: marking rev 110087 from trunk as not applying 2008-03-19 20:37 +0000 [r110085] Mark Michelson * apps/app_chanspy.c, /: Merged revisions 110084 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110084 | mmichelson | 2008-03-19 15:34:13 -0500 (Wed, 19 Mar 2008) | 12 lines Merged revisions 110083 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110083 | mmichelson | 2008-03-19 15:33:03 -0500 (Wed, 19 Mar 2008) | 4 lines Add a missing unlock in the case that memory allocation fails in app_chanspy. Thanks to Russell for confirming that this was an issue. ........ ................ 2008-03-19 19:14 +0000 [r110037] Joshua Colp * /, res/res_musiconhold.c: Merged revisions 110036 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110036 | file | 2008-03-19 16:13:39 -0300 (Wed, 19 Mar 2008) | 12 lines Merged revisions 110035 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110035 | file | 2008-03-19 16:11:33 -0300 (Wed, 19 Mar 2008) | 4 lines Add sanity checking for position resuming. We *have* to make sure that the position does not exceed the total number of files present, and we have to make sure that the position's filename is the same as previous. These values can change if a music class is reloaded and give unpredictable behavior. (closes issue #11663) Reported by: junky ........ ................ 2008-03-19 19:00 +0000 [r110024-110032] Russell Bryant * Makefile, build_tools/cflags.xml, build_tools/cflags-devmode.xml (added), /: Merged revisions 109974 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109974 | qwell | 2008-03-19 12:15:14 -0500 (Wed, 19 Mar 2008) | 13 lines Merged revisions 109973 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109973 | qwell | 2008-03-19 12:12:52 -0500 (Wed, 19 Mar 2008) | 5 lines People report bugs about Asterisk crashing with DO_CRASH enabled was getting a little silly... Now we only show certain cflags when you run configure with --enable-dev-mode (corresponding menuselect change to follow) ........ ................ 2008-03-19 18:26 +0000 [r109971-110021] Joshua Colp * main/rtp.c, /: Merged revisions 110020 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110020 | file | 2008-03-19 15:25:33 -0300 (Wed, 19 Mar 2008) | 14 lines Merged revisions 110019 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110019 | file | 2008-03-19 15:20:28 -0300 (Wed, 19 Mar 2008) | 6 lines Make sure that the mark bit does not incorrectly cause video frame timestamps to be calculated as if they are audio frames. (closes issue #11429) Reported by: sperreault Patches: 11429-frametype.diff uploaded by qwell (license 4) ........ ................ 2008-03-19 16:46 +0000 [r109969] Steve Murphy * main/config.c, /: Merged revisions 109942 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109942 | murf | 2008-03-19 10:24:51 -0600 (Wed, 19 Mar 2008) | 80 lines Merged revisions 109908 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109908 | murf | 2008-03-19 09:41:13 -0600 (Wed, 19 Mar 2008) | 72 lines (closes issue #11442) Reported by: tzafrir Patches: 11442.patch uploaded by murf (license 17) Tested by: murf I didn't give tzafrir very much time to test this, but if he does still have remaining issues, he is welcome to re-open this bug, and we'll do what is called for. I reproduced the problem, and tested the fix, so I hope I am not jumping by just going ahead and committing the fix. The problem was with what file_save does with templates; firstly, it tended to print out multiple options: [my_category](!)(templateref) instead of [my_category](!,templateref) which is fixed by this patch. Nextly, the code to suppress output of duplicate declarations that would occur because the reader copies inherited declarations down the hierarchy, was not working. Thus: [master-template](!) mastervar = bar [template](!,master-template) tvar = value [cat](template) catvar = val would be rewritten as: ;! ;! Automatically generated configuration file ;! Filename: experiment.conf (/etc/asterisk/experiment.conf) ;! Generator: Manager ;! Creation Date: Tue Mar 18 23:17:46 2008 ;! [master-template](!) mastervar = bar [template](!,master-template) mastervar = bar tvar = value [cat](template) mastervar = bar tvar = value catvar = val This has been fixed. Since the config reader 'explodes' inherited vars into the category, users may, in certain circumstances, see output different from what they originally entered, but it should be both correct and equivalent. ........ ................ 2008-03-19 04:06 +0000 [r109834-109840] Russell Bryant * /, main/utils.c: Merged revisions 109839 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109839 | russell | 2008-03-18 23:06:31 -0500 (Tue, 18 Mar 2008) | 10 lines Merged revisions 109838 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109838 | russell | 2008-03-18 23:06:05 -0500 (Tue, 18 Mar 2008) | 2 lines Tweak spacing in a recent change because I'm very picky. ........ ................ * apps/app_chanspy.c, /: Merged revisions 109764 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109764 | russell | 2008-03-18 17:36:02 -0500 (Tue, 18 Mar 2008) | 11 lines Merged revisions 109763 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109763 | russell | 2008-03-18 17:34:42 -0500 (Tue, 18 Mar 2008) | 3 lines Fix one place where the chanspy datastore isn't removed from a channel. (issue #12243, reported by atis, patch by me) ........ ................ 2008-03-18 23:23 +0000 [r109779] Tilghman Lesher * /, configs/res_ldap.conf.sample, res/res_config_ldap.c: Merged revisions 109775 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109775 | tilghman | 2008-03-18 18:22:25 -0500 (Tue, 18 Mar 2008) | 3 lines Change back to using ldap_initialize() and let the user specify a URL directly, instead of trying to piece it together, badly. ........ 2008-03-18 21:03 +0000 [r109716] Mark Michelson * /, apps/app_queue.c: Merged revisions 109714 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109714 | mmichelson | 2008-03-18 15:59:02 -0500 (Tue, 18 Mar 2008) | 20 lines Merged revisions 109713 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109713 | mmichelson | 2008-03-18 15:52:15 -0500 (Tue, 18 Mar 2008) | 12 lines This patch makes it so that all queue member status changes are handled through device state code. This removes several problems people were seeing where their queue members would get into an "unknown" state. Huge props go to atis on this one since he was the one who found the code section that was causing the problem and proposed the solution. I just wrote what he suggested :) (closes issue #12127) Reported by: atis Patches: 12127v3.patch uploaded by putnopvut (license 60) Tested by: atis, jvandal ........ ................ 2008-03-18 20:14 +0000 [r109684] Tilghman Lesher * /, res/res_config_ldap.c: Merged revisions 109683 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109683 | tilghman | 2008-03-18 15:13:40 -0500 (Tue, 18 Mar 2008) | 4 lines Set protocol version, port number correctly. (closes issue #12211, closes issue #12209) Reported by: sylvain ........ 2008-03-18 19:24 +0000 [r109654] Jason Parker * /, codecs/log2comp.h: Merged revisions 109651 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109651 | qwell | 2008-03-18 14:24:15 -0500 (Tue, 18 Mar 2008) | 15 lines Merged revisions 109648 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109648 | qwell | 2008-03-18 14:23:44 -0500 (Tue, 18 Mar 2008) | 7 lines Allow codecs that use log2comp (g726) to compile correctly on x86 with gcc4 optimizations. (closes issue #12253) Reported by: fossil Patches: log2comp.patch uploaded by fossil (license 140) ........ ................ 2008-03-18 19:00 +0000 [r109546-109622] Mark Michelson * /, channels/chan_agent.c: Merged revisions 109576 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109576 | mmichelson | 2008-03-18 12:59:18 -0500 (Tue, 18 Mar 2008) | 14 lines Merged revisions 109575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109575 | mmichelson | 2008-03-18 12:58:11 -0500 (Tue, 18 Mar 2008) | 6 lines Make sure an agent doesn't try to send dtmf to a NULL channel closes issue #12242 Reported by Yourname ........ ................ * include/asterisk/astmm.h, /: Merged revisions 109545 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109545 | mmichelson | 2008-03-18 12:00:53 -0500 (Tue, 18 Mar 2008) | 3 lines Add format attribute to printf-style functions in astmm.h ........ 2008-03-18 Russell Bryant * Asterisk 1.6.0-beta6 released. 2008-03-18 17:01 +0000 [r109546] Mark Michelson * include/asterisk/astmm.h, /: Merged revisions 109545 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109545 | mmichelson | 2008-03-18 12:00:53 -0500 (Tue, 18 Mar 2008) | 3 lines Add format attribute to printf-style functions in astmm.h ........ 2008-03-18 16:26 +0000 [r109487] Kevin P. Fleming * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, channels/chan_misdn.c, /: Merged revisions 109475 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109475 | kpfleming | 2008-03-18 11:23:05 -0500 (Tue, 18 Mar 2008) | 2 lines fix up various warnings found via the addition of format string checking... some of these were really, really bad code ........ 2008-03-18 15:58 +0000 [r109454-109459] Russell Bryant * Makefile, channels/chan_misdn.c, include/asterisk/strings.h, res/res_indications.c, utils/extconf.c, main/asterisk.c, apps/app_voicemail.c, utils/check_expr.c, cdr/cdr_sqlite3_custom.c, apps/app_meetme.c, /, res/res_phoneprov.c, main/utils.c, channels/chan_iax2.c, utils/frame.c, main/cli.c, funcs/func_enum.c, main/manager.c, include/asterisk/astobj.h, res/res_agi.c, main/features.c, apps/app_minivm.c, res/res_realtime.c, res/res_config_ldap.c, include/asterisk/utils.h, channels/chan_sip.c, apps/app_festival.c, main/translate.c, main/jitterbuf.c, utils/astman.c, include/jitterbuf.h, apps/app_queue.c: Merged revisions 109447 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109447 | twilson | 2008-03-18 10:43:34 -0500 (Tue, 18 Mar 2008) | 3 lines Go through and fix a bunch of places where character strings were being interpreted as format strings. Most of these changes are solely to make compiling with -Wsecurity and -Wformat=2 happy, and were not actual problems, per se. I also added format attributes to any printf wrapper functions I found that didn't have them. -Wsecurity and -Wmissing-format-attribute added to --enable-dev-mode. ........ * configs/sip_notify.conf.sample, /: Merged revisions 109111 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109111 | qwell | 2008-03-17 11:37:31 -0500 (Mon, 17 Mar 2008) | 10 lines Add sample events for aastra phones. aastra-check-cfg is the same as the other check-cfg entries, and aastra-xml is to load a pre-configured xml script. (closes issue #12229) Reported by: gowen72 Patches: aastra.patch uploaded by gowen72 (license 432) ........ 2008-03-18 15:50 +0000 [r109453] Kevin P. Fleming * /, configure, include/asterisk/autoconfig.h.in, acinclude.m4: Merged revisions 109451 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109451 | kpfleming | 2008-03-18 10:50:29 -0500 (Tue, 18 Mar 2008) | 2 lines ensure that dependencies on AST_C_DEFINE_CHECK symbols work properly ........ 2008-03-18 15:50 +0000 [r109448-109452] Russell Bryant * main/dial.c, /: Merged revisions 108962 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108962 | mvanbaak | 2008-03-16 16:50:58 -0500 (Sun, 16 Mar 2008) | 15 lines Merged revisions 108961 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108961 | mvanbaak | 2008-03-16 22:47:10 +0100 (Sun, 16 Mar 2008) | 7 lines add missing break to case AST_CONTROL_SRCUPDATE (closes issue #12228) Reported by: andrew Patches: SRC.patch uploaded by andrew (license 240) ........ ................ 2008-03-18 15:16 +0000 [r109398] Joshua Colp * main/manager.c, /, main/logger.c: Merged revisions 109396 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109396 | file | 2008-03-18 12:13:07 -0300 (Tue, 18 Mar 2008) | 3 lines Make sure values are interpreted as character strings and not format strings. (AST-2008-004) ........ 2008-03-18 15:14 +0000 [r109397] Steve Murphy * pbx/ael/ael-test/ael-ntest23 (added), pbx/ael/ael-test/ael-ntest23/t1/a.ael, pbx/ael/ael-test/ael-ntest23/t1/b.ael, pbx/ael/ael-test/ael-ntest23/t1/c.ael, pbx/ael/ael-test/ael-ntest23/t2/d.ael, pbx/ael/ael-test/ael-ntest23/t2/e.ael, pbx/ael/ael-test/ael-ntest23/t2/f.ael, res/ael/ael_lex.c, pbx/ael/ael-test/ref.ael-ntest23 (added), pbx/ael/ael-test/ael-ntest23/t3/g.ael, pbx/ael/ael-test/ael-ntest23/t3/h.ael, pbx/ael/ael-test/ael-ntest23/t3/i.ael, res/ael/ael.flex, pbx/ael/ael-test/ael-ntest23/t3/j.ael, pbx/ael/ael-test/ael-ntest23/qq.ael, pbx/ael/ael-test/ael-ntest23/t1, pbx/ael/ael-test/ael-ntest23/t2, pbx/ael/ael-test/ael-ntest23/t3, /, pbx/ael/ael-test/ael-ntest23/extensions.ael: Merged revisions 109357 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109357 | murf | 2008-03-18 08:09:50 -0600 (Tue, 18 Mar 2008) | 25 lines Merged revisions 109309 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109309 | murf | 2008-03-18 00:37:15 -0600 (Tue, 18 Mar 2008) | 17 lines (closes issue #11903) Reported by: atis Many thanks to atis for spotting this problem and reporting it. The fix was to straighten out how items are placed on and removed from the file stack. Regressions as well as the provided test case helped to straighten out all code paths. valgrind was used to make sure all memory allocated was freed. Sorry for not solving this earlier. I got distracted. Added the ntest23 regression test, which is mainly a copy of ntest22, but with a few juicy errors thrown in, to replicate the kind of error that atis spotted. ........ ................ 2008-03-18 15:11 +0000 [r109395] Jason Parker * /, channels/chan_sip.c: Merged revisions 109389 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109389 | qwell | 2008-03-18 10:07:04 -0500 (Tue, 18 Mar 2008) | 3 lines Do not return with a successful authentication if the From header ends up empty. (AST-2008-003) ........ 2008-03-18 15:09 +0000 [r109392] Joshua Colp * main/rtp.c, /, channels/chan_sip.c: Merged revisions 109390 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109390 | file | 2008-03-18 12:08:09 -0300 (Tue, 18 Mar 2008) | 11 lines Merged revisions 109386 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109386 | file | 2008-03-18 11:58:39 -0300 (Tue, 18 Mar 2008) | 3 lines Put a maximum limit on the number of payloads accepted, and also make sure a given payload does not exceed our maximum value. (AST-2008-002) ........ ................ 2008-03-18 00:40 +0000 [r109283] Sean Bright * /, configure, configure.ac: Merged revisions 109282 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109282 | seanbright | 2008-03-17 20:28:39 -0400 (Mon, 17 Mar 2008) | 1 line Fix a typo ........ 2008-03-17 22:24 +0000 [r109254] Terry Wilson * build_tools/cflags.xml, /, build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, main/Makefile, configure.ac, main/http.c, main/minimime (removed), build_tools/make_buildopts_h, makeopts.in: Merged revisions 109229 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109229 | twilson | 2008-03-17 17:10:06 -0500 (Mon, 17 Mar 2008) | 5 lines Replace minimime with superior GMime library so that the entire contents of an http post are not read into memory. This does introduce a dependency on the GMime library for handling HTTP POSTs, but it is available in most distros. If the library is present, then the compile flag for ENABLE_UPLOADS is enabled by default in menuselect. ........ 2008-03-17 22:07 +0000 [r109228] Mark Michelson * /, main/utils.c: Merged revisions 109227 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109227 | mmichelson | 2008-03-17 17:06:44 -0500 (Mon, 17 Mar 2008) | 20 lines Merged revisions 109226 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109226 | mmichelson | 2008-03-17 17:05:49 -0500 (Mon, 17 Mar 2008) | 12 lines Fix a logic flaw in the code that stores lock info which is displayed via the "core show locks" command. The idea behind this section of code was to remove the previous lock from the list if it was a trylock that had failed. Unfortunately, instead of checking the status of the previous lock, we were referencing the index immediately following the previous lock in the lock_info->locks array. The result of this problem, under the right circumstances, was that the lock which we currently in the process of attempting to acquire could "overwrite" the previous lock which was acquired. While this does not in any way affect typical operation, it *could* lead to misleading "core show locks" output. ........ ................ 2008-03-17 18:11 +0000 [r109175] Michiel van Baak * /, channels/chan_skinny.c: Merged revisions 109168 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109168 | mvanbaak | 2008-03-17 18:43:46 +0100 (Mon, 17 Mar 2008) | 11 lines Update the directory of placed calls on skinny phones when dialing a channel that does not provide progress (analog ZAP lines) The phone does handle the double update on calls to channels that do provide progress and wont insert duplicate items (closes issue #12239) Reported by: DEA Patches: chan_skinny-call-log.txt uploaded by DEA (license 3) ........ 2008-03-17 17:42 +0000 [r109167] Kevin P. Fleming * Makefile, /, configure, configure.ac, acinclude.m4: Merged revisions 109166 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109166 | kpfleming | 2008-03-17 12:31:46 -0500 (Mon, 17 Mar 2008) | 3 lines don't define Zaptel features as libraries, they aren't, and we don't want '--with-zaptel-' configure options for them also some minor cleanups ........ 2008-03-17 16:47 +0000 [r109109-109114] Joshua Colp * /, channels/chan_sip.c: Merged revisions 109108 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109108 | file | 2008-03-17 13:26:36 -0300 (Mon, 17 Mar 2008) | 12 lines Merged revisions 109107 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109107 | file | 2008-03-17 13:24:29 -0300 (Mon, 17 Mar 2008) | 4 lines 200 OKs in response to a reinvite need to be sent reliably. If the remote side does not receive one the dialog will be torn down. (closes issue #12208) Reported by: atrash ........ ................ 2008-03-17 14:21 +0000 [r109027] Mark Michelson * apps/app_chanspy.c, /: Merged revisions 109024 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109024 | mmichelson | 2008-03-17 09:21:14 -0500 (Mon, 17 Mar 2008) | 14 lines Merged revisions 109012 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109012 | mmichelson | 2008-03-17 09:18:26 -0500 (Mon, 17 Mar 2008) | 6 lines Make sure that we release the lock on the spyee channel if the spyee or spy has hung up (closes issue #12232) Reported by: atis ........ ................ 2008-03-16 17:56 +0000 [r108928-108930] Russell Bryant * apps/app_voicemail.c, /: Merged revisions 108927 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r108927 | russell | 2008-03-16 12:53:46 -0500 (Sun, 16 Mar 2008) | 7 lines Fix polling for mailbox changes in mailboxes that are not in the default vm context. (closes issue #12223) Reported by: DEA Patches: vm-polled-imap.txt uploaded by DEA (license 3) ........ 2008-03-15 16:21 +0000 [r108741-108895] Russell Bryant * Makefile, /: Merged revisions 108799 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r108799 | russell | 2008-03-14 15:14:06 -0500 (Fri, 14 Mar 2008) | 8 lines Make sure configure is run before menuselect on a clean checkout (closes issue #12197) Reported by: juggie Patches: 12197.diff uploaded by juggie (license 24) ........ * channels/chan_oss.c, /: Merged revisions 108797 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108797 | russell | 2008-03-14 15:09:37 -0500 (Fri, 14 Mar 2008) | 13 lines Merged revisions 108796 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108796 | russell | 2008-03-14 15:09:22 -0500 (Fri, 14 Mar 2008) | 5 lines Fix a channel name issue. chan_oss registers the "Console" channel type, but it created channels with an "OSS" prefix. (closes issue #12194, reported by davidw, patched by me) ........ ................ * contrib/init.d/rc.suse.asterisk, /: Merged revisions 108793 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108793 | russell | 2008-03-14 15:04:56 -0500 (Fri, 14 Mar 2008) | 12 lines Merged revisions 108792 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108792 | russell | 2008-03-14 15:04:35 -0500 (Fri, 14 Mar 2008) | 4 lines Update the SuSE init script to start networking before asterisk, as well. (closes issue #12200, reported by and change suggested by reinerotto) ........ ................ * /, configure, acinclude.m4: Merged revisions 108740 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r108740 | russell | 2008-03-14 12:05:11 -0500 (Fri, 14 Mar 2008) | 5 lines Do a link test in AST_EXT_TOOL_CHECK() to ensure we have all the required libs reported by the tool. (closes issue #12067, reported by Juggie, patched by me) ........ 2008-03-14 16:54 +0000 [r108739] Mark Michelson * /, channels/chan_sip.c: Merged revisions 108738 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108738 | mmichelson | 2008-03-14 11:52:51 -0500 (Fri, 14 Mar 2008) | 41 lines Merged revisions 108737 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108737 | mmichelson | 2008-03-14 11:44:08 -0500 (Fri, 14 Mar 2008) | 33 lines Fix a race condition in the SIP packet scheduler which could cause a crash. chan_sip uses the scheduler API in order to schedule retransmission of reliable packets (such as INVITES). If a retransmission of a packet is occurring, then the packet is removed from the scheduler and retrans_pkt is called. Meanwhile, if a response is received from the packet as previously transmitted, then when we ACK the response, we will remove the packet from the scheduler and free the packet. The problem is that both the ACK function and retrans_pkt attempt to acquire the same lock at the beginning of the function call. This means that if the ACK function acquires the lock first, then it will free the packet which retrans_pkt is about to read from and write to. The result is a crash. The solution: 1. If the ACK function fails to remove the packet from the scheduler and the retransmit id of the packet is not -1 (meaning that we have not reached the maximum number of retransmissions) then release the lock and yield so that retrans_pkt may acquire the lock and operate. 2. Make absolutely certain that the ACK function does not recursively lock the lock in question. If it does, then releasing the lock will do no good, since retrans_pkt will still be unable to acquire the lock. (closes issue #12098) Reported by: wegbert (closes issue #12089) Reported by: PTorres Patches: 12098-putnopvutv3.patch uploaded by putnopvut (license 60) Tested by: jvandal ........ ................ 2008-03-14 14:33 +0000 [r108684] Jason Parker * /, res/res_musiconhold.c: Merged revisions 108683 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108683 | qwell | 2008-03-14 09:32:55 -0500 (Fri, 14 Mar 2008) | 12 lines Merged revisions 108682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108682 | qwell | 2008-03-14 09:29:05 -0500 (Fri, 14 Mar 2008) | 4 lines Fix a potential segfault if chan (or chan->music_state) is NULL. Closes issue #12210, credit to edantie for pointing this out. ........ ................ 2008-03-13 21:48 +0000 [r108587] Mark Michelson * main/manager.c, /: Merged revisions 108586 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r108586 | mmichelson | 2008-03-13 16:47:55 -0500 (Thu, 13 Mar 2008) | 3 lines Make this compile ........ 2008-03-13 21:41 +0000 [r108585] Russell Bryant * apps/app_chanspy.c, main/channel.c, /, include/asterisk/channel.h: Merged revisions 108584 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108584 | russell | 2008-03-13 16:40:43 -0500 (Thu, 13 Mar 2008) | 19 lines Merged revisions 108583 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108583 | russell | 2008-03-13 16:38:16 -0500 (Thu, 13 Mar 2008) | 11 lines Fix another issue that was causing crashes in chanspy. This introduces a new datastore callback, called chan_fixup(). The concept is exactly like the fixup callback that is used in the channel technology interface. This callback gets called when the owning channel changes due to a masquerade. Before this was introduced, if a masquerade happened on a channel being spyed on, the channel pointer in the datastore became invalid. (closes issue #12187) (reported by, and lots of testing from atis) (props to file for the help with ideas) ........ ................ 2008-03-13 21:31 +0000 [r108582] Mark Michelson * main/manager.c, /: Merged revisions 108529 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r108529 | mmichelson | 2008-03-13 15:59:00 -0500 (Thu, 13 Mar 2008) | 11 lines Fixing a potential buffer overflow in the manager command ModuleCheck. Though this overflow is exploitable remotely, we are NOT issuing a security advisory for this since in order to exploit the overflow, the attacker would have to establish an authenticated manager session AND have the system privilege. By gaining this privilege, the attacker already has more powerful weapons at his disposal than overflowing a buffer with a malformed manager header, so the vulnerability in this case really lies with the authentication method that allowed the attacker to gain the system privilege in the first place. ........ 2008-03-13 21:07 +0000 [r108347-108532] Russell Bryant * /, channels/chan_sip.c: Merged revisions 108531 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108531 | russell | 2008-03-13 16:06:52 -0500 (Thu, 13 Mar 2008) | 18 lines Merged revisions 108530 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108530 | russell | 2008-03-13 16:06:33 -0500 (Thu, 13 Mar 2008) | 10 lines Make a tweak that gets the LEDs on polycom phones to blink when an extension that has been subscribed to goes on hold. Otherwise, they just stay on like it does when an extension is in use. (closes issue #11263) Reported by: russell Patches: notify_hold.rev1.txt uploaded by russell (license 2) Tested by: russell ........ ................ * apps/app_voicemail.c, /: Merged revisions 108508 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r108508 | russell | 2008-03-13 15:35:28 -0500 (Thu, 13 Mar 2008) | 2 lines Fix a place where configuration values could cause an overflow of a buffer. ........ * /, apps/app_followme.c: Merged revisions 108472 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108472 | russell | 2008-03-13 15:26:59 -0500 (Thu, 13 Mar 2008) | 12 lines Merged revisions 108469 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108469 | russell | 2008-03-13 15:26:28 -0500 (Thu, 13 Mar 2008) | 4 lines Fix a couple uses of sprintf. The second one could actually cause an overflow of a stack buffer. It's not a security issue though, it only depends on your configuration. ........ ................ * /, main/features.c: Merged revisions 107465 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r107465 | file | 2008-03-11 10:05:17 -0500 (Tue, 11 Mar 2008) | 4 lines Clarify comment about masquerading and playback of the parking slot. (closes issue #12180) Reported by: davidw ........ * /, channels/chan_sip.c: Merged revisions 107157 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r107157 | file | 2008-03-10 15:00:21 -0500 (Mon, 10 Mar 2008) | 4 lines If we receive a 488 on a T38 request reinvite back to audio. As well reinvite across a bridge back to audio if one side doesn't negotiate to T38. (closes issue #8677) Reported by: alex-911 ........ * /: Merged revisions 106892 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r106892 | mattf | 2008-03-07 16:36:49 -0600 (Fri, 07 Mar 2008) | 1 line Make sure we don't start a call when we have already done so in response to a COT message ........ * /, main/editline/Makefile.in: Merged revisions 106843 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r106843 | qwell | 2008-03-07 16:15:20 -0600 (Fri, 07 Mar 2008) | 13 lines Merged revisions 106842 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106842 | qwell | 2008-03-07 16:14:45 -0600 (Fri, 07 Mar 2008) | 5 lines Fix hardcoded grep in editline, were GNU grep is required. (closes issue #12124) Reported by: dmartin ........ ................ * include/asterisk/http.h, main/tcptls.c, main/manager.c, /, channels/chan_sip.c, res/res_phoneprov.c, main/http.c, include/asterisk/tcptls.h: Merged revisions 108295 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r108295 | russell | 2008-03-12 17:13:18 -0500 (Wed, 12 Mar 2008) | 3 lines Rename ast_tcptls_server_instance to session_instance, since this pertains to server and client usage. ........ * /, main/http.c: Merged revisions 108346 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r108346 | russell | 2008-03-12 17:49:26 -0500 (Wed, 12 Mar 2008) | 4 lines Make the default prefix empty, like it was in Asterisk 1.4. (closes issue #12198, reported by bkruse, patched by me) ........ 2008-03-12 22:10 +0000 [r108246-108294] Mark Michelson * /, channels/chan_sip.c: Merged revisions 108293 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r108293 | mmichelson | 2008-03-12 17:09:52 -0500 (Wed, 12 Mar 2008) | 3 lines Let's get this to compile ........ * /, channels/chan_sip.c: Merged revisions 108289 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108289 | mmichelson | 2008-03-12 16:57:41 -0500 (Wed, 12 Mar 2008) | 22 lines Merged revisions 108288 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108288 | mmichelson | 2008-03-12 16:53:46 -0500 (Wed, 12 Mar 2008) | 14 lines Change AST_SCHED_DEL use to ast_sched_del for autocongestion in chan_sip. The scheduler callback will always return 0. This means that this id is never rescheduled, so it makes no sense to loop trying to delete the id from the scheduler queue. If we fail to remove the item from the queue once, it will fail every single time. (Yes I realize that in this case, the macro would exit early because the id is set to -1 in the callback, but it still makes no sense to use that macro in favor of calling ast_sched_del once and being done with it) This is the first of potentially several such fixes. ........ ................ * /, include/asterisk/sched.h: Merged revisions 108238 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108238 | mmichelson | 2008-03-12 16:19:30 -0500 (Wed, 12 Mar 2008) | 20 lines Merged revisions 108227 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108227 | mmichelson | 2008-03-12 16:16:28 -0500 (Wed, 12 Mar 2008) | 12 lines Added a large comment before the AST_SCHED_DEL macro to explain its purpose as well as when it is appropriate and when it is not appropriate to use it. I also removed the part of the debug message that mentions that this is probably a bug because there are some perfectly legitimate places where ast_sched_del may fail to delete an entry (e.g. when the scheduler callback manually reschedules with a new id instead of returning non-zero to tell the scheduler to reschedule with the same idea). I also raised the debug level of the debug message in AST_SCHED_DEL since it seems like it could come up quite frequently since the macro is probably being used in several places where it shouldn't be. Also removed the redundant line, file, and function information since that is provided by ast_log. ........ ................ 2008-03-12 20:29 +0000 [r108205] Kevin P. Fleming * /, channels/chan_sip.c: Merged revisions 108191 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108191 | kpfleming | 2008-03-12 15:27:01 -0500 (Wed, 12 Mar 2008) | 14 lines Merged revisions 108086 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108086 | kpfleming | 2008-03-12 14:16:07 -0500 (Wed, 12 Mar 2008) | 6 lines if we receive an INVITE with a Content-Length that is not a valid number, or is zero, then don't process the rest of the message body looking for an SDP closes issue #11475 Reported by: andrebarbosa ........ ................ 2008-03-12 19:59 +0000 [r108138] Russell Bryant * apps/app_chanspy.c, main/channel.c, /: Merged revisions 108137 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108137 | russell | 2008-03-12 14:59:05 -0500 (Wed, 12 Mar 2008) | 48 lines Merged revisions 108135 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108135 | russell | 2008-03-12 14:57:42 -0500 (Wed, 12 Mar 2008) | 40 lines (closes issue #12187, reported by atis, fixed by me after some brainstorming on the issue with mmichelson) - Update copyright info on app_chanspy. - Fix a race condition that caused app_chanspy to crash. The issue was that the chanspy datastore magic that was used to ensure that spyee channels did not disappear out from under the code did not completely solve the problem. It was actually possible for chanspy to acquire a channel reference out of its datastore to a channel that was in the middle of being destroyed. That was because datastore destruction in ast_channel_free() was done near the end. So, this left the code in app_chanspy accessing a channel that was partially, or completely invalid because it was in the process of being free'd by another thread. The following sort of shows the code path where the race occurred: ============================================================================= Thread 1 (PBX thread for spyee chan) || Thread 2 (chanspy) --------------------------------------||------------------------------------- ast_channel_free() || - remove channel from channel list || - lock/unlock the channel to ensure || that no references retrieved from || the channel list exist. || --------------------------------------||------------------------------------- || channel_spy() - destroy some channel data || - Lock chanspy datastore || - Retrieve reference to channel || - lock channel || - Unlock chanspy datastore --------------------------------------||------------------------------------- - destroy channel datastores || - call chanspy datastore d'tor || which NULL's out the ds' || - Operate on the channel ... reference to the channel || || - free the channel || || || - unlock the channel --------------------------------------||------------------------------------- ============================================================================= ........ ................ 2008-03-12 18:31 +0000 [r108085] Joshua Colp * apps/app_mixmonitor.c, /, include/asterisk/audiohook.h, main/audiohook.c: Merged revisions 108084 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108084 | file | 2008-03-12 15:29:33 -0300 (Wed, 12 Mar 2008) | 12 lines Merged revisions 108083 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108083 | file | 2008-03-12 15:26:37 -0300 (Wed, 12 Mar 2008) | 4 lines Add a trigger mode that triggers on both read and write. The actual function that returns the combined audio frame though will wait until both sides have fed in audio, or until one side stops (such as the case when you call Wait). (closes issue #11945) Reported by: xheliox ........ ................ 2008-03-12 17:03 +0000 [r108033] Russell Bryant * main/channel.c, /: Merged revisions 108032 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108032 | russell | 2008-03-12 12:02:57 -0500 (Wed, 12 Mar 2008) | 12 lines Merged revisions 108031 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108031 | russell | 2008-03-12 11:59:07 -0500 (Wed, 12 Mar 2008) | 4 lines Destroy the channel lock after the channel datastores. (inspired by issue #12187) ........ ................ 2008-03-12 07:44 +0000 [r107879-107999] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 107998 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r107998 | tilghman | 2008-03-12 02:43:03 -0500 (Wed, 12 Mar 2008) | 7 lines Deadlock fixes (closes issue #12143) Reported by: kactus Patches: 20080312__bug12143__2.diff.txt uploaded by Corydon76 (license 14) Tested by: kactus ........ * main/loader.c, /, apps/app_dumpchan.c, apps/app_zapras.c: Merged revisions 107960 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r107960 | tilghman | 2008-03-12 00:46:39 -0500 (Wed, 12 Mar 2008) | 4 lines Revert several changes from revision 102525, as the changes were not compatible, and, in fact, introduced regressions. (Closes issue #12190) ........ * contrib/scripts/iax-friends.sql, /, contrib/scripts/sip-friends.sql: Merged revisions 107878 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107878 | tilghman | 2008-03-11 20:54:00 -0500 (Tue, 11 Mar 2008) | 10 lines Merged revisions 107877 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107877 | tilghman | 2008-03-11 20:52:40 -0500 (Tue, 11 Mar 2008) | 2 lines Document all of the possible realtime fields ........ ................ 2008-03-11 23:38 +0000 [r107828] Jason Parker * /, doc/voicemail_odbc_postgresql.txt: Merged revisions 107827 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107827 | qwell | 2008-03-11 18:38:00 -0500 (Tue, 11 Mar 2008) | 15 lines Merged revisions 107826 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107826 | qwell | 2008-03-11 18:37:05 -0500 (Tue, 11 Mar 2008) | 7 lines Update documentation for pgsql ODBC voicemail. (closes issue #12186) Reported by: jsmith Patches: vm_pgsql_doc_update.patch uploaded by jsmith (license 15) ........ ................ 2008-03-11 22:59 +0000 [r107723-107793] Tilghman Lesher * res/res_config_sqlite.c, main/config.c, res/res_config_curl.c, res/res_config_pgsql.c, res/res_config_odbc.c, /, include/asterisk/config.h, res/res_config_ldap.c: Merged revisions 107791 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r107791 | tilghman | 2008-03-11 17:55:16 -0500 (Tue, 11 Mar 2008) | 5 lines An offhand comment from Russell made me realize that the configuration file caching would not work properly for users.conf and any other file read from more than one place. I needed to add the filename which requested the config file to get it to work properly. ........ 2008-03-11 20:54 +0000 [r107720] Jason Parker * channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged revisions 107718 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107718 | qwell | 2008-03-11 15:53:48 -0500 (Tue, 11 Mar 2008) | 13 lines Merged revisions 107714 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107714 | qwell | 2008-03-11 15:49:56 -0500 (Tue, 11 Mar 2008) | 5 lines Copy voicemail dependency logic for res_adsi to chan_gtalk and chan_jingle (for jabber). (closes issue #12014) Reported by: junky ........ ................ 2008-03-11 20:51 +0000 [r107716] Kevin P. Fleming * /, Makefile.rules, channels/Makefile: Merged revisions 107715 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107715 | kpfleming | 2008-03-11 15:50:57 -0500 (Tue, 11 Mar 2008) | 10 lines Merged revisions 107713 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107713 | kpfleming | 2008-03-11 15:48:58 -0500 (Tue, 11 Mar 2008) | 2 lines get chan_vpb to build properly in dev mode ........ ................ 2008-03-11 20:37 +0000 [r107584-107711] Joshua Colp * /, apps/app_page.c: Merged revisions 107710 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r107710 | file | 2008-03-11 17:36:14 -0300 (Tue, 11 Mar 2008) | 6 lines Dial a device even if it's state is unknown. (closes issue #12184) Reported by: bluecrow76 Patches: asterisk-svn-app_page.c.devicestate_unknown.diff uploaded by bluecrow76 (license 270) ........ * /, main/features.c: Merged revisions 107659 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107659 | file | 2008-03-11 16:23:28 -0300 (Tue, 11 Mar 2008) | 12 lines Merged revisions 107646 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107646 | file | 2008-03-11 16:20:01 -0300 (Tue, 11 Mar 2008) | 4 lines Make sure the visible indication is on the right channel so when the masquerade happens the proper indication is enacted. (closes issue #11707) Reported by: iam ........ ................ * /, apps/app_meetme.c: Merged revisions 107638 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107638 | file | 2008-03-11 15:48:59 -0300 (Tue, 11 Mar 2008) | 12 lines Merged revisions 107637 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107637 | file | 2008-03-11 15:47:33 -0300 (Tue, 11 Mar 2008) | 4 lines Add an additional check for setting conference parameter when using the marked user options. It was possible for it to return to a no listen/no talk state if a masquerade happened. (closes issue #12136) Reported by: aragon ........ ................ 2008-03-11 15:39 +0000 [r107374-107526] Kevin P. Fleming * channels/chan_vpb.cc, /: Merged revisions 107525 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r107525 | kpfleming | 2008-03-11 10:39:37 -0500 (Tue, 11 Mar 2008) | 2 lines fix another potential bug found by gcc 4.3 ........ * apps/app_rpt.c, channels/misdn/isdn_lib.c, codecs/Makefile, /, apps/app_sms.c: Merged revisions 107466 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107466 | kpfleming | 2008-03-11 10:13:38 -0500 (Tue, 11 Mar 2008) | 10 lines Merged revisions 107464 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107464 | kpfleming | 2008-03-11 09:53:03 -0500 (Tue, 11 Mar 2008) | 2 lines fix various other problems found by gcc 4.3 ........ ................ * /, configure, include/asterisk/autoconfig.h.in, configure.ac, apps/app_sms.c: Merged revisions 107462 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107462 | kpfleming | 2008-03-11 09:37:03 -0500 (Tue, 11 Mar 2008) | 10 lines Merged revisions 107461 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107461 | kpfleming | 2008-03-11 09:33:45 -0500 (Tue, 11 Mar 2008) | 2 lines stop checking for mktime() in the configure script... we don't use it, and the test is buggy under gcc 4.3 ........ ................ * /, configure, main/Makefile, configure.ac, makeopts.in: Merged revisions 107409 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107409 | kpfleming | 2008-03-11 09:09:49 -0500 (Tue, 11 Mar 2008) | 13 lines Merged revisions 107408 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107408 | kpfleming | 2008-03-11 09:07:59 -0500 (Tue, 11 Mar 2008) | 5 lines check for compiler support for -fno-strict-overflow before using it (tested with Debian's gcc 4.3, 4.1 and 3.4) (closes issue #12179) Reported by: Netview ........ ................ * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 107406 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107406 | kpfleming | 2008-03-11 08:58:37 -0500 (Tue, 11 Mar 2008) | 10 lines Merged revisions 107405 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107405 | kpfleming | 2008-03-11 08:57:08 -0500 (Tue, 11 Mar 2008) | 2 lines fix small bug in IMAP toolkit testing ........ ................ * main/udptl.c, utils/Makefile, /, main/Makefile, main/editline/readline.c, res/Makefile: Merged revisions 107373 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107373 | kpfleming | 2008-03-11 06:36:51 -0500 (Tue, 11 Mar 2008) | 19 lines Merged revisions 107352 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107352 | kpfleming | 2008-03-11 06:04:29 -0500 (Tue, 11 Mar 2008) | 11 lines fix up various compiler warnings found with gcc-4.3: - the output of flex includes a static function called 'input' that is not used, so for the moment we'll stop having the compiler tell us about unused variables in the flex source files (a better fix would be to improve our flex post-processing to remove the unused function) - main/stdtime/localtime.c makes assumptions about signed integer overflow, and gcc-4.3's improved optimizer tries to take advantage of handling potential overflow conditions at compile time; for now, suppress these optimizations until we can fiure out if the code needs improvement - main/udptl.c has some references to uninitialized variables; in one case there was no bug, but in the other it was certainly possibly for unexpected behavior to occur - main/editline/readline.c had an unused variable ........ ................ 2008-03-11 01:27 +0000 [r107336] Terry Wilson * /, channels/chan_sip.c: Merged revisions 107292 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107292 | twilson | 2008-03-10 20:09:46 -0500 (Mon, 10 Mar 2008) | 10 lines Merged revisions 107290 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107290 | twilson | 2008-03-10 19:59:18 -0500 (Mon, 10 Mar 2008) | 2 lines If we fail to alloc a channel, we should re-lock the pvt structure before returning. ........ ................ 2008-03-10 23:46 +0000 [r107289] Steve Murphy * main/cdr.c, /: Merged revisions 107019 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r107019 | murf | 2008-03-10 08:55:21 -0600 (Mon, 10 Mar 2008) | 1 line way back in July, in r.75706, a fix was made ot the strftime usages, which was good, but in this case, the check for a nil time was accidentally removed, and now it is restored, to keep timevals like '1969-12-31 17:00:00' from showing up in the cdrs. No idea what databases will do with this. No bugs filed as yet, but it felt like a bug. ........ 2008-03-10 20:29 +0000 [r107180] Jason Parker * channels/chan_zap.c, /: Merged revisions 107177 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107177 | qwell | 2008-03-10 15:28:33 -0500 (Mon, 10 Mar 2008) | 13 lines Merged revisions 107173 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107173 | qwell | 2008-03-10 15:27:08 -0500 (Mon, 10 Mar 2008) | 5 lines Make sure to reenable echo can after a "failed" (canceled, etc) three-way call. (closes issue #11335) Reported by: rebuild ........ ................ 2008-03-10 20:18 +0000 [r107101-107163] Russell Bryant * main/pbx.c, /: Merged revisions 107162 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107162 | russell | 2008-03-10 15:17:37 -0500 (Mon, 10 Mar 2008) | 16 lines Merged revisions 107161 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107161 | russell | 2008-03-10 15:17:11 -0500 (Mon, 10 Mar 2008) | 8 lines Fix another bug specifically related to asynchronous call origination. Once the PBX is started on the channel using ast_pbx_start(), then the ownership of the channel has been passed on to another thread. We can no longer access it in this code. If the channel gets hung up very quickly, it is possible that we could access a channel that has been free'd. (inspired by BE-386) ........ ................ * main/pbx.c, /: Merged revisions 107159 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107159 | russell | 2008-03-10 15:05:12 -0500 (Mon, 10 Mar 2008) | 17 lines Merged revisions 107158 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107158 | russell | 2008-03-10 15:04:27 -0500 (Mon, 10 Mar 2008) | 9 lines Fix some bugs related to originating calls. If the code failed to start a PBX on the channel (such as if you set a call limit based on the system's load average), then there were cases where a channel that has already been free'd using ast_hangup() got accessed. This caused weird memory corruption and crashes to occur. (fixes issue BE-386) (much debugging credit goes to twilson, final patch written by me) ........ ................ * main/channel.c, /: Merged revisions 107103 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107103 | russell | 2008-03-10 12:13:34 -0500 (Mon, 10 Mar 2008) | 10 lines Merged revisions 107102 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107102 | russell | 2008-03-10 12:13:17 -0500 (Mon, 10 Mar 2008) | 2 lines Resolve a compiler warning. ........ ................ * main/channel.c, /: Merged revisions 107100 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107100 | russell | 2008-03-10 11:59:13 -0500 (Mon, 10 Mar 2008) | 11 lines Merged revisions 107099 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107099 | russell | 2008-03-10 11:58:57 -0500 (Mon, 10 Mar 2008) | 3 lines Fix a race condition where the generator can go away (closes issue #12175, reported by edantie, patched by me) ........ ................ 2008-03-10 15:46 +0000 [r107069] Mark Michelson * /, apps/app_queue.c: Merged revisions 107068 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r107068 | mmichelson | 2008-03-10 10:45:13 -0500 (Mon, 10 Mar 2008) | 10 lines app_queue has now been doxygenified thanks to snuffy! The ony thing I changed was the way that locks are referenced, since the old 1.2 names were still used in the comments. (closes issue #11997) Reported by: snuffy Patches: bug_11997_queue_doxy.diff uploaded by snuffy (license 35) ........ 2008-03-10 14:38 +0000 [r107018] Joshua Colp * apps/app_dial.c, main/cdr.c, /, include/asterisk/cdr.h: Merged revisions 107017 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107017 | file | 2008-03-10 11:36:16 -0300 (Mon, 10 Mar 2008) | 15 lines Merged revisions 107016 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107016 | file | 2008-03-10 11:33:02 -0300 (Mon, 10 Mar 2008) | 7 lines Move where unanswered CDRs are dropped to the CDR core, not everything uses app_dial. (closes issue #11516) Reported by: ys Patches: branch_1.4_cdr.diff uploaded by ys (license 281) Tested by: anest, jcapp, dartvader ........ ................ 2008-03-08 17:54 +0000 [r106997] Matthew Fredrickson * channels/chan_zap.c: Make sure we don't start a call on a channel that has already started a call 2008-03-08 16:14 +0000 [r106947] Kevin P. Fleming * channels/chan_zap.c, /: Merged revisions 106946 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r106946 | kpfleming | 2008-03-08 10:03:48 -0600 (Sat, 08 Mar 2008) | 10 lines Merged revisions 106945 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106945 | kpfleming | 2008-03-08 09:59:42 -0600 (Sat, 08 Mar 2008) | 2 lines don't generate D-Channel "up" and "down" messages unless the channel state is actually changing; also, generate the "up" message when an implicit "up" occurs due to reception of a normal event when we thought the channel was "down" ........ ................ 2008-03-07 22:53 +0000 [r106897] Russell Bryant * /, apps/app_meetme.c: Merged revisions 106896 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r106896 | russell | 2008-03-07 16:52:46 -0600 (Fri, 07 Mar 2008) | 10 lines Merged revisions 106895 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106895 | russell | 2008-03-07 16:51:23 -0600 (Fri, 07 Mar 2008) | 2 lines Only start the SLA thread if SLA has actually been configured. ........ ................ 2008-03-07 19:34 +0000 [r106790] Joshua Colp * main/channel.c, /: Merged revisions 106789 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r106789 | file | 2008-03-07 15:33:09 -0400 (Fri, 07 Mar 2008) | 12 lines Merged revisions 106788 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106788 | file | 2008-03-07 15:32:00 -0400 (Fri, 07 Mar 2008) | 4 lines Ignore source update control frame. (closes issue #12168) Reported by: plack ........ ................ 2008-03-07 17:18 +0000 [r106686-106713] Russell Bryant * /, include/asterisk/sched.h: Merged revisions 106707 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r106707 | russell | 2008-03-07 11:17:30 -0600 (Fri, 07 Mar 2008) | 16 lines Merged revisions 106704 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106704 | russell | 2008-03-07 11:16:58 -0600 (Fri, 07 Mar 2008) | 8 lines Change a warning message to a debug message. This is happening quite frequently, and it is not worth spamming users with these messages unless we are pretty confident that it should never happen. As it stands today, it _will_ and _does_ happen and until that gets cleaned up a reasonable amount on the development side, let's not spam the logs of everyone else. (closes issue #12154) ........ ................ * doc/smdi.txt, /: Merged revisions 106684 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r106684 | russell | 2008-03-07 10:31:48 -0600 (Fri, 07 Mar 2008) | 2 lines fix example usage ........ 2008-03-07 16:27 +0000 [r106554-106662] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 106654 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r106654 | tilghman | 2008-03-07 10:26:07 -0600 (Fri, 07 Mar 2008) | 11 lines Merged revisions 106635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106635 | tilghman | 2008-03-07 10:22:11 -0600 (Fri, 07 Mar 2008) | 3 lines Warn the user when a temporary greeting exists (Closes issue #11409) ........ ................ * main/rtp.c, /: Merged revisions 106607 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r106607 | tilghman | 2008-03-07 09:22:34 -0600 (Fri, 07 Mar 2008) | 11 lines Merged revisions 106606 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106606 | tilghman | 2008-03-07 09:20:52 -0600 (Fri, 07 Mar 2008) | 3 lines Properly initialize rtp->schedid (Closes issue #12154) ........ ................ * apps/app_chanspy.c, apps/app_rpt.c, main/asterisk.c, apps/app_speech_utils.c, apps/app_voicemail.c, main/channel.c, funcs/func_enum.c, channels/chan_misdn.c, main/frame.c, /, channels/chan_sip.c, funcs/func_odbc.c, funcs/func_strings.c, utils/extconf.c: Merged revisions 106553 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r106553 | tilghman | 2008-03-07 00:54:47 -0600 (Fri, 07 Mar 2008) | 14 lines Merged revisions 106552 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106552 | tilghman | 2008-03-07 00:36:33 -0600 (Fri, 07 Mar 2008) | 6 lines Safely use the strncat() function. (closes issue #11958) Reported by: norman Patches: 20080209__bug11958.diff.txt uploaded by Corydon76 (license 14) ........ ................ 2008-03-07 01:19 +0000 [r106502-106520] Russell Bryant * doc/smdi.txt, /: Merged revisions 106518 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r106518 | russell | 2008-03-06 19:19:02 -0600 (Thu, 06 Mar 2008) | 1 line minor text changes ........ * doc/smdi.txt, /: Merged revisions 106507 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r106507 | russell | 2008-03-06 19:15:36 -0600 (Thu, 06 Mar 2008) | 2 lines Add updated SMDI documentation that I had only sitting in my email ... oops ........ * main/rtp.c, codecs/codec_g722.c, /, formats/format_pcm.c, main/file.c: Merged revisions 106501 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r106501 | russell | 2008-03-06 18:24:58 -0600 (Thu, 06 Mar 2008) | 28 lines Merge changes from team/russell/g722-sillyness ... Fix a number of other places where the number of samples in a G722 frame was not properly handled because of various reasons. main/rtp.c: - When a G722 frame is read from the smoother, the number of samples in the frame must be divided by 2 before being sent out over the network. Even though G722 is 16 kHz, an error in some previous spec has made it so that we have to list the number of samples such as if it was 8 kHz. main/file.c: - When scheduling the next time to expect a frame, take into account that the format of the file we're reading from may not be 8 kHz. codecs/codec_g722.c: - When converting from G722 to slinear, g722_decode() expects its samples parameter to be in the silly (real samples / 2) format. Make it so. - When converting from slinear to G722, properly set the number of samples in the frame to be the number of bytes of output * 2. formats/format_pcm.c: - This format module handles G722, among a number of other formats. However, the read() and seek() functions did not account for the fact that G722 has 2 samples per byte. (closes issue #12130, reported by rickross, patched by me) ........ 2008-03-06 22:16 +0000 [r106442] Mark Michelson * main/pbx.c, /: Merged revisions 106438 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r106438 | mmichelson | 2008-03-06 16:11:26 -0600 (Thu, 06 Mar 2008) | 16 lines Merged revisions 106437 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106437 | mmichelson | 2008-03-06 16:10:07 -0600 (Thu, 06 Mar 2008) | 8 lines Quell an annoying message that is likely to print every single time that ast_pbx_outgoing_app is called. The reason is that __ast_request_and_dial allocates the cdr for the channel, so it should be expected that the channel will have a cdr on it. Thanks to joetester on IRC for pointing this out ........ ................ 2008-03-06 22:15 +0000 [r106440] Jason Parker * /, main/file.c: Merged revisions 106439 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r106439 | qwell | 2008-03-06 16:11:30 -0600 (Thu, 06 Mar 2008) | 8 lines Fix file playback in many cases. (closes issue #12115) Reported by: pj Patches: v2-fileexists.patch uploaded by dimas (license 88) (with modifications by me) Tested by: dimas, qwell, russell ........ 2008-03-06 20:39 +0000 [r106433] Donny Kavanagh * /, res/res_agi.c: Merged revisions 106399 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r106399 | juggie | 2008-03-06 14:31:50 -0500 (Thu, 06 Mar 2008) | 9 lines trivial fix for an agi error when attempting to use EAGI on a dead/hungup channel, we now print an error that makes sense given our removal of deadagi as an actual application. (closes issue #12161) Reported by: explidous Patches: res_agi_12161.patch uploaded by juggie (license 24) Tested by: juggie ........ 2008-03-06 05:25 +0000 [r106330-106359] Tilghman Lesher * /, res/res_config_ldap.c: Merged revisions 106346 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r106346 | tilghman | 2008-03-05 23:21:39 -0600 (Wed, 05 Mar 2008) | 7 lines Missing braces, fix parsing (closes issue #12112) Reported by: cyrenity Patches: res_config_ldap.patch-03-03-2008 uploaded by cyrenity (license 416) Tested by: cyrenity, Corydon76 ........ * /, sounds/sounds.xml, sounds/Makefile: Merged revisions 106329 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r106329 | tilghman | 2008-03-05 22:45:16 -0600 (Wed, 05 Mar 2008) | 10 lines Merged revisions 106328 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106328 | tilghman | 2008-03-05 22:40:06 -0600 (Wed, 05 Mar 2008) | 2 lines Upgrade to the next release of sounds ........ ................ 2008-03-06 00:23 +0000 [r106299-106320] Russell Bryant * channels/chan_oss.c, main/rtp.c, main/channel.c, channels/chan_phone.c, main/dial.c, channels/chan_skinny.c, main/file.c, channels/chan_h323.c, channels/chan_alsa.c, include/asterisk/frame.h, channels/chan_mgcp.c, channels/chan_unistim.c, apps/app_dial.c, channels/chan_zap.c, /, channels/chan_sip.c, channels/chan_console.c, apps/app_followme.c: Merged revisions 106239 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r106239 | file | 2008-03-05 16:43:22 -0600 (Wed, 05 Mar 2008) | 12 lines Merged revisions 106235 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106235 | file | 2008-03-05 18:32:10 -0400 (Wed, 05 Mar 2008) | 4 lines Add a control frame to indicate the source of media has changed. Depending on the underlying technology it may need to change some things. (closes issue #12148) Reported by: jcomellas ........ ................ * /, channels/chan_iax2.c: Merged revisions 106238 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r106238 | russell | 2008-03-05 16:40:58 -0600 (Wed, 05 Mar 2008) | 11 lines Merged revisions 106237 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106237 | russell | 2008-03-05 16:37:09 -0600 (Wed, 05 Mar 2008) | 3 lines Fix a potential deadlock and a few different potential crashes. (closes issue #12145, reported by thiagarcia, patched by me) ........ ................ * /, doc/tex/realtime.tex: Merged revisions 106186 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r106186 | mvanbaak | 2008-03-05 15:19:06 -0600 (Wed, 05 Mar 2008) | 7 lines document var_metric usage to prevent bugreports that are actually configuration issues (closes issue #12151) Reported by: caio1982 Patches: DB_metric3.diff uploaded by caio1982 (license 22) ........ * main/rtp.c, /, main/translate.c, include/asterisk/frame.h: Merged revisions 105933 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r105933 | russell | 2008-03-04 19:54:16 -0600 (Tue, 04 Mar 2008) | 13 lines Merged revisions 105932 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105932 | russell | 2008-03-04 19:52:18 -0600 (Tue, 04 Mar 2008) | 5 lines Fix a bug that I just noticed in the RTP code. The calculation for setting the len field in an ast_frame of audio was wrong when G.722 is in use. The len field represents the number of ms of audio that the frame contains. It would have set the value to be twice what it should be. ........ ................ * funcs/func_global.c, /: Merged revisions 105899 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r105899 | russell | 2008-03-04 18:45:39 -0600 (Tue, 04 Mar 2008) | 3 lines Fix the SHARED() read callback to properly unlock the channel. This function could not have worked, as it left the channel locked in all cases. ........ * main/manager.c, /: Merged revisions 105864 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r105864 | mmichelson | 2008-03-04 17:24:56 -0600 (Tue, 04 Mar 2008) | 5 lines There are several places in manager.c where BUFSIZ is used for a buffer which will contain nowhere near that amount of data. This makes these buffers more reasonably sized. ........ * main/asterisk.c, channels/chan_zap.c, /, channels/console_gui.c, apps/app_queue.c: Merged revisions 105841 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r105841 | tilghman | 2008-03-04 17:10:45 -0600 (Tue, 04 Mar 2008) | 2 lines Fix minor misuses of snprintf ........ * main/rtp.c, main/netsock.c, main/cryptostub.c, main/file.c, main/callerid.c, main/alaw.c, main/dsp.c, main/dlfcn.c, main/frame.c, /, main/say.c, main/utils.c, main/enum.c, main/astobj2.c, main/config.c, main/fskmodem.c, main/poll.c, main/loader.c, main/term.c, main/cli.c, main/channel.c, main/dial.c, main/manager.c, main/tdd.c, main/strcompat.c, main/features.c, main/logger.c, main/app.c, main/image.c, main/dns.c, main/pbx.c, main/translate.c, main/jitterbuf.c: Merged revisions 105840 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r105840 | tilghman | 2008-03-04 17:04:29 -0600 (Tue, 04 Mar 2008) | 2 lines Whitespace changes only ........ * main/tcptls.c, main/manager.c, /, channels/chan_sip.c, main/http.c, include/asterisk/tcptls.h: Merged revisions 105804 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r105804 | russell | 2008-03-04 16:28:03 -0600 (Tue, 04 Mar 2008) | 2 lines add a destroy API call for a server instance ........ * main/tcptls.c, main/manager.c, /, channels/chan_sip.c, main/http.c, include/asterisk/tcptls.h: Merged revisions 105785 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r105785 | russell | 2008-03-04 16:23:21 -0600 (Tue, 04 Mar 2008) | 2 lines More public API name changes to use an appropriate ast_ prefix ........ * include/asterisk/http.h, main/tcptls.c, main/manager.c, /, channels/chan_sip.c, res/res_phoneprov.c, main/http.c, include/asterisk/tcptls.h: Merged revisions 105773 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r105773 | russell | 2008-03-04 16:15:18 -0600 (Tue, 04 Mar 2008) | 2 lines Rename public object server_instance to ast_tcptls_server_instance ........ * /, channels/chan_sip.c: Merged revisions 105734 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r105734 | russell | 2008-03-04 14:36:16 -0600 (Tue, 04 Mar 2008) | 6 lines Fix some bugs in the SIP tcp helper thread. - fix a spot where a lock wouldn't get unlocked in an error condition - call ast_mutex_destroy() on the lock before freeing its memory (related to issue #11972) ........ * /, res/res_phoneprov.c: Merged revisions 105733 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r105733 | twilson | 2008-03-04 14:32:55 -0600 (Tue, 04 Mar 2008) | 2 lines Set username to default to the category name if it isn't overridden by a usernmae= setting in users.conf ........ * main/rtp.c, /: Merged revisions 105677 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r105677 | file | 2008-03-04 12:11:38 -0600 (Tue, 04 Mar 2008) | 10 lines Merged revisions 105676 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105676 | file | 2008-03-04 14:10:34 -0400 (Tue, 04 Mar 2008) | 2 lines In addition to setting the marker bit let's change our ssrc so they know for sure it is a different source. ........ ................ * main/rtp.c, /, channels/chan_sip.c, include/asterisk/rtp.h: Merged revisions 105675 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r105675 | file | 2008-03-04 12:08:42 -0600 (Tue, 04 Mar 2008) | 16 lines Merged revisions 105674 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105674 | file | 2008-03-04 14:05:28 -0400 (Tue, 04 Mar 2008) | 8 lines When a new source of audio comes in (such as music on hold) make sure the marker bit gets set. (closes issue #10355) Reported by: wdecarne Patches: 10355.diff uploaded by file (license 11) (closes issue #11491) Reported by: kanderson ........ ................ 2008-03-05 17:42 +0000 [r106140] Tilghman Lesher * /, apps/app_talkdetect.c: Merged revisions 106139 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r106139 | tilghman | 2008-03-05 11:40:42 -0600 (Wed, 05 Mar 2008) | 3 lines Should check these values for non-NULL before scanning. (Closes issue #12147) ........ 2008-03-05 15:43 +0000 [r106041] Kevin P. Fleming * channels/chan_zap.c, /: Merged revisions 106040 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r106040 | kpfleming | 2008-03-05 09:40:40 -0600 (Wed, 05 Mar 2008) | 15 lines Merged revisions 106038 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106038 | kpfleming | 2008-03-05 09:32:35 -0600 (Wed, 05 Mar 2008) | 7 lines when a PRI call must be moved to a different B channel at the request of the other endpoint, ensure that any DSP active on the original channel is moved to the new one (closes issue #11917) Reported by: mavetju Tested by: mavetju ........ ................ 2008-03-05 15:31 +0000 [r106037] Tilghman Lesher * /, channels/chan_sip.c, include/asterisk/sched.h: Merged revisions 106036 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r106036 | tilghman | 2008-03-05 09:23:32 -0600 (Wed, 05 Mar 2008) | 15 lines Merged revisions 106015 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106015 | tilghman | 2008-03-05 09:17:16 -0600 (Wed, 05 Mar 2008) | 7 lines Correctly initialize retransid in SIP, and ensure that the warning when failing to delete a schedule entry can actually hit the log. (closes issue #12140) Reported by: slavon Patches: sch2.patch uploaded by slavon (license 288) (Patch slightly modified by me) ........ ................ 2008-03-04 Russell Bryant * Asterisk 1.6.0-beta5 released. 2008-03-04 16:55 +0000 [r105574-105597] Russell Bryant * CHANGES: Update CHANGES heading * funcs/func_version.c: Simplify a trivial snprintf() with ast_copy_string() * main/hashtab.c: Make it so you don't have to cast away const in a couple places * main/hashtab.c: remove unnecessary casts * main/pbx.c: - Add curly braces around the while loop - Properly break out of the loop on error when an included context is not found * main/pbx.c: Use ast_copy_string() instead of strncpy(), and use sizeof() instead of a magic number * channels/chan_zap.c: Fix some code that was improperly changed in revision 104866 from issue #12079. (closes issue #12129, reported by elguero, patched by me) 2008-03-03 18:08 +0000 [r105573] Jason Parker * /, res/snmp/agent.c: Merged revisions 105572 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105572 | qwell | 2008-03-03 12:06:52 -0600 (Mon, 03 Mar 2008) | 7 lines Fix types for astNumChannels and astConfigCallsProcessed. (closes issue #12114) Reported by: jeffg Patches: 12114.patch uploaded by jeffg (license 192) ........ 2008-03-03 17:17 +0000 [r105564-105571] Russell Bryant * channels/chan_local.c, /: Merged revisions 105570 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105570 | russell | 2008-03-03 11:16:53 -0600 (Mon, 03 Mar 2008) | 3 lines In the case of an ast_channel allocation failure, take the local_pvt out of the pvt list before destroying it. ........ * channels/chan_local.c, /: Merged revisions 105568 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105568 | russell | 2008-03-03 11:05:16 -0600 (Mon, 03 Mar 2008) | 3 lines Fix a potential memory leak of the local_pvt struct when ast_channel allocation fails. Also, in passing, centralize the code necessary to destroy a local_pvt. ........ * main/autoservice.c, /: Merged revisions 105565 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105565 | russell | 2008-03-03 10:01:50 -0600 (Mon, 03 Mar 2008) | 3 lines Update the copyright information for autoservice. Most of the code in this file now is stuff that I have written recently ... ........ * main/channel.c, main/autoservice.c, /, include/asterisk/_private.h, main/asterisk.c: 3) In addition to merging the changes below, change trunk back to a regular LIST instead of an RWLIST. The way this list works makes it such that a RWLIST provides no additional benefit. Also, a mutex is needed for use with the thread condition. Merged revisions 105563 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105563 | russell | 2008-03-03 09:50:43 -0600 (Mon, 03 Mar 2008) | 24 lines Merge in some changes from team/russell/autoservice-nochans-1.4 These changes fix up some dubious code that I came across while auditing what happens in the autoservice thread when there are no channels currently in autoservice. 1) Change it so that autoservice thread doesn't keep looping around calling ast_waitfor_n() on 0 channels twice a second. Instead, use a thread condition so that the thread properly goes to sleep and does not wake up until a channel is put into autoservice. This actually fixes an interesting bug, as well. If the autoservice thread is already running (almost always is the case), then when the thread goes from having 0 channels to have 1 channel to autoservice, that channel would have to wait for up to 1/2 of a second to have the first frame read from it. 2) Fix up the code in ast_waitfor_nandfds() for when it gets called with no channels and no fds to poll() on, such as was the case with the previous code for the autoservice thread. In this case, the code would call alloca(0), and pass the result as the first argument to poll(). In this case, the 2nd argument to poll() specified that there were no fds, so this invalid pointer shouldn't actually get dereferenced, but, this code makes it explicit and ensures the pointers are NULL unless we have valid data to put there. (related to issue #12116) ........ 2008-03-03 15:30 +0000 [r105558-105561] Joshua Colp * main/channel.c, /: Merged revisions 105560 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105560 | file | 2008-03-03 11:28:59 -0400 (Mon, 03 Mar 2008) | 7 lines It is possible for no audio to pass between the current digit and next digit so expand logic that clears emulation to AST_FRAME_NULL. (closes issue #11911) Reported by: edgreenberg Patches: v1-11911.patch uploaded by dimas (license 88) Tested by: tbsky ........ * /, channels/chan_sip.c: Merged revisions 105557 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105557 | file | 2008-03-03 11:15:39 -0400 (Mon, 03 Mar 2008) | 6 lines Add a comment to describe some logic. (closes issue #12120) Reported by: flefoll Patches: chan_sip.c.br14.patch-just-a-comment uploaded by flefoll (license 244) ........ 2008-03-01 03:59 +0000 [r105509] Joshua Colp * main/slinfactory.c: Add support for 16KHz signed linear. 2008-03-01 02:03 +0000 [r105479] Tilghman Lesher * /: Drop bad property 2008-03-01 01:30 +0000 [r105477] Terry Wilson * apps/app_dial.c, include/asterisk/app.h, main/global_datastores.c, /, main/features.c, main/app.c, include/asterisk/global_datastores.h: Asterisk, when parking can drop rights a caller when a parking timeout occurs. Also, when doing built-in attended transfers, sometimes incorrectly passes rights from the transferrer to the transferee. This patch tries to fixes the parking issue and lays some groundwork for later fixing the transfer issue. (closes issue #11520) Reported by: pliew Tested by: otherwiseguy 2008-03-01 00:53 +0000 [r105461] Russell Bryant * CHANGES, funcs/func_devstate.c: Add a "devstate change" CLI command to control custom device states. Also, do some additional code cleanup and improvement in passing. (closes issue #12106) Reported by: nizon Patches: devstate-patch.txt uploaded by nizon (license 415) -- Updated to trunk, and tab completion added by me 2008-02-29 23:53 +0000 [r105411] Tilghman Lesher * cdr/cdr_adaptive_odbc.c: Convert to use ast_str 2008-02-29 23:36 +0000 [r105410] Russell Bryant * main/autoservice.c, /: Merged revisions 105409 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105409 | russell | 2008-02-29 17:34:32 -0600 (Fri, 29 Feb 2008) | 23 lines Fix a major bug in autoservice. There was a race condition in the handling of the list of channels in autoservice. The problem was that it was possible for a channel to get removed from autoservice and destroyed, while the autoservice thread was still messing with the channel. This led to memory corruption, and caused crashes. This explains multiple backtraces I have seen that have references to autoservice, but do to the nature of the issue (memory corruption), could cause crashes in a number of areas. (fixes the crash in BE-386) (closes issue #11694) (closes issue #11940) The following issues could be related. If you are the reporter of one of these, please update to include this fix and try again. (potentially fixes issue #11189) (potentially fixes issue #12107) (potentially fixes issue #11573) (potentially fixes issue #12008) (potentially fixes issue #11189) (potentially fixes issue #11993) (potentially fixes issue #11791) ........ 2008-02-29 18:34 +0000 [r105378] Joshua Colp * configs/sip.conf.sample: Add documentation for setting username/password in SIP dial string. (closes issue #11587) Reported by: sobomax Patches: dialstring_doc.diff uploaded by sobomax (license 359) 2008-02-29 14:50 +0000 [r105263-105327] Philippe Sultan * /, res/res_jabber.c: Merged revisions 105326 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105326 | phsultan | 2008-02-29 15:47:10 +0100 (Fri, 29 Feb 2008) | 1 line Fix a potential memory leak ........ * channels/chan_jingle.c, channels/chan_gtalk.c, res/res_jabber.c: Remove unnecessary if statements before calling iks_delete (redundant check is done inside iks_delete), thus making the code conform with coding guidelines. 2008-02-29 13:55 +0000 [r105262] Joshua Colp * /, apps/app_voicemail.c: Merged revisions 105261 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105261 | file | 2008-02-29 09:48:13 -0400 (Fri, 29 Feb 2008) | 4 lines Bump up the size of the uniqueid variable. (closes issue #12107) Reported by: asgaroth ........ 2008-02-29 13:12 +0000 [r105210] Philippe Sultan * res/res_jabber.c: Automatically create new buddy upon reception of a presence stanza of type subscribed. (closes issue #12066) Reported by: ffadaie Patches: branch-1.4-12066-1.diff uploaded by phsultan (license 73) trunk-12066-1.diff uploaded by phsultan (license 73) Tested by: ffadaie, phsultan 2008-02-29 01:15 +0000 [r105176] Tilghman Lesher * contrib/init.d/rc.debian.asterisk, /: Merged revisions 105113 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105113 | tilghman | 2008-02-28 15:56:54 -0600 (Thu, 28 Feb 2008) | 7 lines Update init script for LSB compat (closes issue #9843) Reported by: ibc Patches: rc.debian.asterisk.patch uploaded by ibc (license 211) Tested by: paravoid ........ 2008-02-28 22:39 +0000 [r105144] Russell Bryant * /, main/utils.c, include/asterisk/lock.h, utils/check_expr.c: Merged revisions 105116 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105116 | russell | 2008-02-28 16:23:05 -0600 (Thu, 28 Feb 2008) | 8 lines Fix a bug in the lock tracking code that was discovered by mmichelson. The issue is that if the lock history array was full, then the functions to mark a lock as acquired or not would adjust the stats for whatever lock is at the end of the array, which may not be itself. So, do a sanity check to make sure that we're updating lock info for the proper lock. (This explains the bizarre stats on lock #63 in BE-396, thanks Mark!) ........ 2008-02-28 20:14 +0000 [r105060-105061] Mark Michelson * /, apps/app_queue.c: Merged revisions 105059 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105059 | mmichelson | 2008-02-28 14:11:57 -0600 (Thu, 28 Feb 2008) | 6 lines When using autofill, members who are in use should be counted towards the number of available members to call if ringinuse is set to yes. Thanks to jmls who brought this issue up on IRC ........ * main/dial.c, /: Merged revisions 104841 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104841 | mmichelson | 2008-02-27 15:49:20 -0600 (Wed, 27 Feb 2008) | 17 lines Two fixes: 1. Make the list of ast_dial_channels a lockable list. This is because in some cases, the ast_dial may exist in multiple threads due to asynchronous execution of its application, and I found some cases where race conditions could exist. 2. Check in ast_dial_join to be sure that the channel still exists before attempting to lock it, since it could have gotten hung up but the is_running_app flag on the ast_dial_channel may not have been cleared yet. (closes issue #12038) Reported by: jvandal Patches: 12038v2.patch uploaded by putnopvut (license 60) Tested by: jvandal ........ 2008-02-28 19:21 +0000 [r105006] Jason Parker * main/cdr.c, main/pbx.c, /: Merged revisions 105005 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105005 | qwell | 2008-02-28 13:20:10 -0600 (Thu, 28 Feb 2008) | 9 lines Make pbx_exec pass an empty string into applications, if we get NULL. This protects against possible segfaults in applications that may try to use data before checking length (ast_strdupa'ing it, for example) (closes issue #12100) Reported by: foxfire Patches: 12100-nullappargs.diff uploaded by qwell (license 4) ........ 2008-02-28 14:42 +0000 [r104974] Tilghman Lesher * channels/chan_vpb.cc: Fix crash when configuration does not match hardware detection. (closes issue #12096) Reported by: mmickan Patches: chan_vpb.cc.diff uploaded by mmickan (license 400) 2008-02-28 04:37 +0000 [r104921] Jason Parker * /, channels/chan_skinny.c: Merged revisions 104920 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104920 | qwell | 2008-02-27 22:31:21 -0600 (Wed, 27 Feb 2008) | 2 lines According to a video at www.cisco.com, the 7921G supports 6 line appearances. ........ 2008-02-28 00:11 +0000 [r104869] Tilghman Lesher * /, main/Makefile, build_tools/strip_nonapi: Merged revisions 104868 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104868 | tilghman | 2008-02-27 18:05:06 -0600 (Wed, 27 Feb 2008) | 7 lines Compatibility fix for PPC64 (closes issue #12081) Reported by: jcollie Patches: asterisk-1.4.18-funcdesc.patch uploaded by jcollie (license 412) Tested by: jcollie, Corydon76 ........ 2008-02-27 23:58 +0000 [r104866] Russell Bryant * channels/chan_zap.c: reduce indentation in alloc_sub (issue #12079) Reported by: tzafrir Patches: alloc_sub uploaded by tzafrir (license 46) 2008-02-27 21:02 +0000 [r104788] Joshua Colp * /, apps/app_chanspy.c: Merged revisions 104787 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104787 | file | 2008-02-27 16:56:23 -0400 (Wed, 27 Feb 2008) | 2 lines Don't loop around infinitely trying to spy on our own channel, and don't forget to free/detach the datastore upon hangup of the spy. ........ 2008-02-27 20:37 +0000 [r104784] Mark Michelson * /, main/file.c: Merged revisions 104783 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104783 | mmichelson | 2008-02-27 14:36:26 -0600 (Wed, 27 Feb 2008) | 4 lines Bump a couple of more buffers up by 2 so that annoying warnings aren't generated like crazy on every fileexists_core call. ........ 2008-02-27 19:36 +0000 [r104756] Jason Parker * apps/app_voicemail.c: Remove useless 's' and 'key' variables, in favor of 'val', which serves the exact same purpose. 2008-02-27 18:20 +0000 [r104705] Tilghman Lesher * main/manager.c, /: Merged revisions 104704 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104704 | tilghman | 2008-02-27 12:15:10 -0600 (Wed, 27 Feb 2008) | 2 lines Ensure the session ID can't be 0. ........ 2008-02-27 17:45 +0000 [r104687] Joshua Colp * /, main/file.c: Merged revisions 104665 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104665 | file | 2008-02-27 13:41:40 -0400 (Wed, 27 Feb 2008) | 2 lines Bump up the buffer by 2. ........ 2008-02-27 17:36 +0000 [r104643] Russell Bryant * /, apps/app_chanspy.c: Merged revisions 104625 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104625 | russell | 2008-02-27 11:33:04 -0600 (Wed, 27 Feb 2008) | 4 lines Fix a problem in ChanSpy where it could get stuck in an infinite loop without being able to detect that the calling channel hung up. (closes issue #12076, reported by junky, patched by me) ........ 2008-02-27 17:31 +0000 [r104617] Jason Parker * /, main/features.c: Merged revisions 104598 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104598 | qwell | 2008-02-27 11:26:55 -0600 (Wed, 27 Feb 2008) | 8 lines Inherit language from the transfering channel on a blind transfer. (closes issue #11682) Reported by: caio1982 Patches: local_atxfer_lang3-1.4.diff uploaded by caio1982 (license 22) Tested by: caio1982, victoryure ........ 2008-02-27 17:12 +0000 [r104595-104597] Joshua Colp * /, main/loader.c: Merged revisions 104596 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104596 | file | 2008-02-27 13:07:33 -0400 (Wed, 27 Feb 2008) | 4 lines Use the lock (which already existed, it just wasn't used) on the updaters list to protect the contents instead of the overall module list lock. (closes issue #12080) Reported by: ChaseVenters ........ * channels/chan_sip.c: After further discussion revert my previous commit for this. Currently in order to ensure devicestate is the expected value in another module (such as app_queue) then chan_sip must be loaded before hand. 2008-02-27 16:54 +0000 [r104594] Kevin P. Fleming * /, main/file.c: Merged revisions 104593 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104593 | kpfleming | 2008-02-27 10:53:06 -0600 (Wed, 27 Feb 2008) | 8 lines fallback to standard English prompts properly when using new prompt directory layout (closes issue #11831) Reported by: IgorG Patches: fallbacken.v1.diff uploaded by IgorG (license 20) (modified by me to improve code and conform rest of function to coding guidelines) ........ 2008-02-27 16:26 +0000 [r104537-104539] Joshua Colp * channels/chan_sip.c: When queueing up a device state change when the peer is loaded from the configuration give it a state of not in use. We have to do this because the channel technology may not yet be registered so the state could not be queried and would be considered invalid. (closes issue #12087) Reported by: liorm * res/res_smdi.c, /: Merged revisions 104536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104536 | file | 2008-02-27 11:52:02 -0400 (Wed, 27 Feb 2008) | 4 lines Only stop the MWI monitor thread if it was actually started. (closes issue #12086) Reported by: francesco_r ........ 2008-02-27 15:34 +0000 [r104534] Tilghman Lesher * utils/astcanary.c: open(2) needs a mode argument when O_CREAT is specified. (Closes issue #12083) 2008-02-27 15:31 +0000 [r104533] Joshua Colp * channels/chan_sip.c, main/rtp.c: Fix T38 passthrough regression introduced by state changes. (closes issue #12078) Reported by: dimas Patches: v1-12078.patch uploaded by dimas (license 88) (closes issue #12074) Reported by: Ivan 2008-02-27 08:20 +0000 [r104502] Tilghman Lesher * channels/chan_vpb.cc, configs/vpb.conf.sample, include/asterisk/module.h: Bring Voicetronix driver up to date with current drivers (closes issue #12084) Reported by: mmickan Patches: chan_vpb.cc.diff uploaded by mmickan (license 400) module.h.diff uploaded by mmickan (license 400) vpb.conf.sample uploaded by mmickan (license 400) 2008-02-27 04:42 +0000 [r104419-104473] Russell Bryant * doc/janitor-projects.txt: note that the chan_sip conversion is already in progress * doc/janitor-projects.txt: add another janitor project * doc/janitor-projects.txt: Add the stuff from the janitor projects page that is still relevant. I figure that if we keep this in the tree, it will be much easier to keep up to date. The page on asterisk.org just links to this on svn.digium.com/view 2008-02-27 03:52 +0000 [r104418] Jason Parker * doc/janitor-projects.txt (added): Create placeholder file...for now. 2008-02-27 02:05 +0000 [r104388] Tilghman Lesher * apps/app_voicemail.c: Whitespace changes only 2008-02-27 01:16 +0000 [r104333-104335] Russell Bryant * /, apps/app_chanspy.c: Merged revisions 104334 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104334 | russell | 2008-02-26 19:15:02 -0600 (Tue, 26 Feb 2008) | 3 lines Avoid some recursion in the cleanup code for the chanspy datastore (closes issue #12076, reported by junky, patched by me) ........ 2008-02-26 22:14 +0000 [r104301] Steve Murphy * res/snmp/agent.c: small change to allow this file to compile. No problem if you don't install the libsnmp package. 2008-02-26 20:33 +0000 [r104244-104270] Russell Bryant * main/asterisk.c: I swear I compiled this ... *cough* * res/res_phoneprov.c: fix this module, too * funcs/func_version.c: fix this module * Makefile, include/asterisk, build_tools/make_version_h (added): Re-add the automatically generated version.h, so that modules can include for making build time decisions for cross asterisk version compatibility * main/manager.c, channels/chan_sip.c, include/asterisk/version.h (removed), build_tools/make_version_c, res/res_agi.c, main/http.c, include/asterisk/ast_version.h (added): Rename version.h to ast_version.h. Next, I will be re-adding version.h as an automatically generated file like it used to be. This still needs to be there for modules that have to check it to compile against multiple asterisk versions. 2008-02-26 19:14 +0000 [r104215] Joshua Colp * main/cdr.c, main/pbx.c, include/asterisk/cdr.h, CHANGES: Add an 'e' option to ResetCDR which re-enables a CDR that has been disabled. (closes issue #11170) Reported by: kratzers Patches: ResetCDR.1.diff uploaded by kratzers (license 307) 2008-02-26 18:40 +0000 [r104176] Tilghman Lesher * doc/CODING-GUIDELINES: 1) Make braces mandatory for if/for/while, even around single statements. 2) Revise the argument parsing section, showing use of the standard macros. 3) Fix a typo. 2008-02-26 18:27 +0000 [r104140-104142] Jason Parker * Makefile, /: Merged revisions 104141 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104141 | qwell | 2008-02-26 12:26:12 -0600 (Tue, 26 Feb 2008) | 1 line Add badshell to .PHONY target (thanks Kevin) ........ * Makefile, /: Merged revisions 104139 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104139 | qwell | 2008-02-26 12:09:13 -0600 (Tue, 26 Feb 2008) | 2 lines Since all shells aren't as awesome as bash, we have to fail if somebody tries to use a literal "~" in DESTDIR. ........ 2008-02-26 16:51 +0000 [r104137] Olle Johansson * channels/chan_sip.c: Formatting and doxygen while waiting on an airport... 2008-02-26 16:36 +0000 [r104133-104136] Jason Parker * /, sounds/Makefile: Merged revisions 104135 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104135 | qwell | 2008-02-26 10:35:06 -0600 (Tue, 26 Feb 2008) | 5 lines Revert previous abspath change. ...abspath is new in GNU make 3.81. I feel so...defeated. Must find new fix! ........ * /, sounds/Makefile: Merged revisions 104132 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104132 | qwell | 2008-02-26 10:08:44 -0600 (Tue, 26 Feb 2008) | 9 lines Fix a very bizarre issue we were seeing with our buildbot when using a DESTDIR that wasn't an absolute path (such as DESTDIR=~/asterisk-1.4). Apparently what was happening, was that some of the targets were being expanded to the full path, so $@ ended up being /root/asterisk-1.4/[...]/ rather than ~/asterisk-1.4/[...]/ It appears that this may be a new "feature" in GNU make. (*cough* http://en.wikipedia.org/wiki/Principle_of_least_surprise *cough*) ........ 2008-02-26 14:51 +0000 [r104127] Mark Michelson * main/features.c: Remove more hardcoded pipe symbols and replace with commas. (closes issue #12072) Reported by: SimonSharman Patches: features.patch uploaded by SimonSharman (license 410) Tested by: SimonSharman 2008-02-26 06:43 +0000 [r104125] Tilghman Lesher * funcs/func_odbc.c: Use the readhandle for reads (closes issue #12069) 2008-02-26 00:38 +0000 [r104120-104124] Russell Bryant * res/res_smdi.c: Add a \todo to convert this module to the event system * CHANGES: Update CHANGES for SMDI stuff * channels/chan_zap.c, res/res_smdi.c, /, configs/smdi.conf.sample, include/asterisk/smdi.h, apps/app_voicemail.c: Merged revisions 104119 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104119 | russell | 2008-02-25 18:25:29 -0600 (Mon, 25 Feb 2008) | 33 lines Merge changes from team/russell/smdi-1.4 This commit brings in a significant set of changes to the SMDI support in Asterisk. There were a number of bugs in the current implementation, most notably being that it was very likely on busy systems to pop off the wrong message from the SMDI message queue. So, this set of changes fixes the issues discovered as well as introducing some new ways to use the SMDI support which are required to avoid the bugs with grabbing the wrong message off of the queue. This code introduces a new interface to SMDI, with two dialplan functions. First, you get an SMDI message in the dialplan using SMDI_MSG_RETRIEVE() and then you access details in the message using the SMDI_MSG() function. A side benefit of this is that it now supports more than just chan_zap. For example, with this implementation, you can have some FXO lines being terminated on a SIP gateway, but the SMDI link in Asterisk. Another issue with the current implementation is that it is quite common that the station ID that comes in on the SMDI link is not necessarily the same as the Asterisk voicemail box. There are now additional directives in the smdi.conf configuration file which let you map SMDI station IDs to Asterisk voicemail boxes. Yet another issue with the current SMDI support was related to MWI reporting over the SMDI link. The current code could only report a MWI change when the change was made by someone calling into voicemail. If the change was made by some other entity (such as with IMAP storage, or with a web interface of some kind), then the MWI change would never be sent. The SMDI module can now poll for MWI changes if configured to do so. This work was inspired by and primarily done for the University of Pennsylvania. (also related to issue #9260) ........ 2008-02-25 23:56 +0000 [r104103-104110] Russell Bryant * channels/chan_zap.c, UPGRADE.txt: Deprecate the "stripmsd" option in favor of dialplan substring variable syntax. (closes issue #12060) * /, apps/app_chanspy.c: Merged revisions 104106 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104106 | russell | 2008-02-25 17:42:42 -0600 (Mon, 25 Feb 2008) | 10 lines This patch fixes some pretty significant problems with how app_chanspy handles pointers to channels that are being spied upon. It was very likely that a crash would occur if the channel being spied upon hung up. This was because the current ast_channel handling _requires_ that the object is locked or else it could disappear at any time (except in the owning channel thread). So, this patch uses some channel datastore magic on the spied upon channel to be able to detect if and when the channel goes away. (closes issue #11877) (patch written by me, but thanks to kpfleming for the idea, and to file for review) ........ * /, main/utils.c: Merged revisions 104102 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104102 | russell | 2008-02-25 17:19:05 -0600 (Mon, 25 Feb 2008) | 7 lines Improve the lock tracking code a bit so that a bunch of old locks that threads failed to lock don't sit around in the history. When a lock is first locked, this checks to see if the last lock in the list was one that was failed to be locked. If it is, then that was a lock that we're no longer sitting in a trylock loop trying to lock, so just remove it. (inspired by issue #11712) ........ 2008-02-25 23:04 +0000 [r104097-104101] Tilghman Lesher * cdr/cdr_pgsql.c, CHANGES: Permit additional CDR columns to be saved in Postgres. Note that these changes are backward-compatible, so no changes to UPGRADE.txt are necessary. (closes issue #9279) Reported by: rottenroddy Patches: 20080125__bug9279.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 * funcs/func_global.c: Shared space for variables (instead of letting other channels muck with your own) (closes issue #11943) Reported by: ramonpeek Patches: 20080208__bug11943__2.diff.txt uploaded by Corydon76 (license 14) Tested by: jmls * /, apps/app_voicemail.c: Merged revisions 104094 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104094 | tilghman | 2008-02-25 15:31:47 -0600 (Mon, 25 Feb 2008) | 5 lines If the destination folder is full, don't delete a message when exiting. (closes issue #12065) Reported by: selsky Patch by: (myself) ........ 2008-02-25 21:40 +0000 [r104096] Joshua Colp * /, channels/chan_sip.c: Merged revisions 104095 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104095 | file | 2008-02-25 17:37:20 -0400 (Mon, 25 Feb 2008) | 6 lines Make it so a users.conf user creates both a SIP peer and a SIP user. The user will be used for inbound authentication for the device, and peer will be used for placing calls to the device. (closes issue #9044) Reported by: queuetue Patches: sip-gui-friend.diff uploaded by qwell (license 4) ........ 2008-02-25 20:50 +0000 [r104093] Jason Parker * /, main/config.c: Merged revisions 104092 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104092 | qwell | 2008-02-25 14:49:42 -0600 (Mon, 25 Feb 2008) | 11 lines Allow the use of #include and #exec in situations where the max include depth was only 1. Specifically, this fixes using #include and #exec in extconfig.conf. This was basically caused because the config file itself raises the include level to 1. I opted not to raise the include limit, because recursion here could cause very bizarre behavior. Pointed out, and tested by jmls (closes issue #12064) ........ 2008-02-25 19:02 +0000 [r104089] Joshua Colp * channels/chan_iax2.c: Instead of outputting a verbose message every so often let's make it a debug message. 2008-02-25 19:00 +0000 [r104088] Brett Bryant * doc/siptls.txt, configs/sip.conf.sample: Adding more tls configuration details to sip.conf sample, with a list of valid ciphers provided in both files. .. First commit since July, woot 2008-02-25 18:38 +0000 [r104087] Russell Bryant * /, channels/chan_agent.c: Merged revisions 104086 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104086 | russell | 2008-02-25 12:38:10 -0600 (Mon, 25 Feb 2008) | 4 lines Ensure that the channel doesn't disappear in agent_logoff(). If it does, it could cause a crash. (fixes the crash reported in BE-396) ........ 2008-02-25 16:18 +0000 [r104081-104085] Joshua Colp * /, channels/chan_sip.c: Merged revisions 104084 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104084 | file | 2008-02-25 12:16:13 -0400 (Mon, 25 Feb 2008) | 6 lines If a resubscription comes in for a dialog we no longer know about tell the remote side that the dialog does not exist so they subscribe again using a new dialog. (closes issue #10727) Reported by: s0l4rb03 Patches: 10727-2.diff uploaded by file (license 11) ........ * /, channels/chan_sip.c: Merged revisions 104082 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104082 | file | 2008-02-25 11:17:18 -0400 (Mon, 25 Feb 2008) | 6 lines Due to recent changes tag will no longer be NULL if not present so we have to use ast_strlen_zero to see if it's actually blank. (closes issue #12061) Reported by: flefoll Patches: chan_sip.c.br14.patch_pedantic_no_totag uploaded by flefoll (license 244) ........ * res/res_config_pgsql.c: Fix building of trunk. dbpass is always going to exist. 2008-02-24 02:37 +0000 [r104073-104074] Steve Murphy * channels/chan_sip.c: Enforce a space between function args as per code review. * res/res_config_pgsql.c: On a 64-bit machine, with dev-mode turned on, and pgsql installed, I get warnings that stops the compile. They are fixed now. 2008-02-22 23:56 +0000 [r104045] Doug Bailey * channels/chan_zap.c, configure, configure.ac: Add protection to chan_zap build when NEONMWI events are not defined 2008-02-22 22:55 +0000 [r104036-104039] Tilghman Lesher * doc/manager_1_1.txt, main/manager.c, UPGRADE.txt, CHANGES, include/asterisk/manager.h: Move Originate to a separate privilege and require the additional System privilege to call out to a subshell. * /, channels/chan_sip.c: Merged revisions 104037 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104037 | tilghman | 2008-02-22 16:45:14 -0600 (Fri, 22 Feb 2008) | 6 lines Backwards debug message. (closes issue #12052) Reported by: flefoll Patches: chan_sip.c.br14.patch_found-notfound uploaded by flefoll (license 244) ........ * res/res_config_pgsql.c: Allow database password to be NULL and several other cleanups. (closes issue #12048) Reported by: bukaj Patches: 20080222__bug12048.diff.txt uploaded by Corydon76 (license 14) Tested by: bukaj 2008-02-21 21:27 +0000 [r104031] Russell Bryant * channels/chan_sip.c: fix a typo 2008-02-21 21:09 +0000 [r104025-104029] Mark Michelson * res/res_agi.c: Instead of a notice, make the message about a hung-up channel a debug message, and revert the original logic on the if statement. Thanks to Juggie for bringing this to my attention. 2008-02-21 17:38 +0000 [r104024] Doug Bailey * channels/chan_zap.c: Added configuration distinction between neon and fsk mwi detection Add the detection for neon MWI events got rid of extraneous handle_init_event call in monitor loop 2008-02-21 16:46 +0000 [r104020] Mark Michelson * res/res_agi.c: Don't print the fact that we are using dead mode in AGI if called from the 'h' extension since it is well-known that it will be running in dead mode. (closes issue #12046) Reported by: explidous 2008-02-21 16:44 +0000 [r104019] Joshua Colp * configure, include/asterisk/autoconfig.h.in, configure.ac: Disable epoll as it has caused more obscure issues then any of my previous code. I will continue to work on it in a separate branch to make it stable for a release and test it against the following issues. (closes issue #11253) Reported by: falves11 (closes issue #11657) Reported by: davevg (closes issue #11033) Reported by: falves11 2008-02-21 14:44 +0000 [r104016] Kevin P. Fleming * main/manager.c, /: Merged revisions 104015 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104015 | kpfleming | 2008-02-21 08:33:51 -0600 (Thu, 21 Feb 2008) | 2 lines reduce the likelihood that HTTP Manager session ids will consist of primarily '1' bits ........ 2008-02-21 05:21 +0000 [r104014] Tilghman Lesher * utils/astman.c: Ignore some more unused generated events. (closes issue #12042) Reported by: junky Patches: astman_events.diff uploaded by junky (license 177) 2008-02-20 Russell Bryant * Asterisk 1.6.0-beta4 released. 2008-02-20 22:34 +0000 [r103957] Mark Michelson * /, apps/app_queue.c: Merged revisions 103956 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103956 | mmichelson | 2008-02-20 16:32:22 -0600 (Wed, 20 Feb 2008) | 8 lines Clear up confusion when viewing the QUEUE_WAITING_COUNT of a "dead" realtime queue. Since from the user's perspective, the queue does exist, we shouldn't tell them we couldn't find the queue. Instead since it is a dead queue, report a 0 waiting count This issue was brought up on IRC by jmls ........ 2008-02-20 22:29 +0000 [r103954-103955] Joshua Colp * channels/chan_h323.c: Try to do Packet2Packet bridging with chan_h323 if reinviting isn't enabled. (closes issue #11901) Reported by: pj * channels/chan_zap.c, /: Merged revisions 103953 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103953 | file | 2008-02-20 18:06:59 -0400 (Wed, 20 Feb 2008) | 6 lines Don't wait for additional digits when overlap dialing is enabled if the setup message contains the sending_complete information element. (closes issue #11785) Reported by: klaus3000 Patches: sending_complete_overlap_asterisk-1.4.17.patch.txt uploaded by klaus3000 (license 65) ........ 2008-02-20 21:41 +0000 [r103908] Mark Michelson * channels/chan_local.c, /: Merged revisions 103904 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103904 | mmichelson | 2008-02-20 15:40:08 -0600 (Wed, 20 Feb 2008) | 6 lines Fix a crash if the channel becomes NULL while attempting to lock it. (closes issue #12039) Reported by: danpwi ........ 2008-02-20 21:36 +0000 [r103903] Jason Parker * include/asterisk/dsp.h, main/dsp.c: Largely refactor DSP tone detection routines. Separate fax detection from digit detected. Added CED (called) tone detection for fax (previously, only CNG (calling) was supported). Separate DTMF/MF code paths where appropriate. Allow detection of arbitary tones. (closes issue #11796) Reported by: dimas Patches: v6-dsp-faxtones.patch uploaded by dimas (license 88) Tested by: dimas, IgorG, Cache 2008-02-20 21:08 +0000 [r103902] Mark Michelson * apps/app_voicemail.c: Fix a crash due to the wrong variable being used when building a directory string. (closes issue #12027) Reported by: jaroth Patches: forward.patch uploaded by jaroth (license 50) Tested by: jaroth 2008-02-20 18:29 +0000 [r103846-103847] Tilghman Lesher * include/asterisk/sched.h: Add some documentation fixups * /, main/stdtime/localtime.c: Merged revisions 103845 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103845 | tilghman | 2008-02-20 11:53:00 -0600 (Wed, 20 Feb 2008) | 7 lines Compat fix for Solaris (closes issue #12022) Reported by: asgaroth Patches: 20080219__bug12022.diff.txt uploaded by Corydon76 (license 14) Tested by: asgaroth ........ 2008-02-20 15:21 +0000 [r103844] Mark Michelson * res/res_monitor.c: Fix another spot where a hard-coded '|' hadn't been converted to ',' (closes issue #12034) Reported by: kowalma 2008-02-20 03:52 +0000 [r103838-103842] Joshua Colp * main/audiohook.c: *mumble* * main/audiohook.c: file not found. * main/audiohook.c: Minor test... 2008-02-20 00:49 +0000 [r103833] Mark Michelson * apps/app_voicemail.c: When using IMAP storage, if the folder you attempt to save to does not exist, create it first. (closes issue #12032) Reported by: jaroth Patches: createfolder.patch uploaded by jaroth (license 50) Tested by: jaroth 2008-02-19 22:35 +0000 [r103831-103832] Jason Parker * main/channel.c: Make sure to mask out non-audio first as well * main/channel.c: Maybe we should set the value before we test it? Fixes an issue people have been seeing (unreported?) with file playback not working. 2008-02-19 21:54 +0000 [r103824-103828] Joshua Colp * main/loader.c: Add a log message that appears when you try to unload a module that isn't loaded. (closes issue #12033) Reported by: jamesgolovich Patches: asterisk-loader.diff.txt uploaded by jamesgolovich (license 176) * main/file.c: Only output a log message saying the format does not exist if it actually does not exist, not if the file itself could not be opened. (closes issue #11828) Reported by: IgorG Patches: readfile.v1.diff uploaded by IgorG (license 20) * /, channels/h323/ast_h323.cxx: Merged revisions 103823 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103823 | file | 2008-02-19 16:28:08 -0400 (Tue, 19 Feb 2008) | 6 lines Send CallerID Name in setup message. (closes issue #11241) Reported by: tusar Patches: h323id_as_callerid_name.patch uploaded by tusar (license 344) ........ 2008-02-19 20:06 +0000 [r103822] Russell Bryant * channels/chan_local.c, /: Merged revisions 103821 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103821 | russell | 2008-02-19 14:02:49 -0600 (Tue, 19 Feb 2008) | 8 lines Account for the fact that the "other" channel can disappear while the local pvt is not locked. (fixes a problem introduced in rev 100581) (closes issue #12012) Reported by: stevedavies Patch by me ........ 2008-02-19 19:27 +0000 [r103819-103820] Joshua Colp * apps/app_authenticate.c: len already contains the position we want to examine, if we move one left again we'll actually probably be looking at a digit. (issue #12030) Reported by: alligosh * apps/app_channelredirect.c, UPGRADE.txt, CHANGES: Add CHANNELREDIRECT_STATUS variable to ChannelRedirect() dialplan application. This will either be set to NOCHANNEL if the given channel was not found or SUCCESS if it worked. (closes issue #11553) Reported by: johan Patches: UPGRADE.txt.channelredirect.patch uploaded by johan (license 334) CHANGES.channelredirect.patch uploaded by johan (license 334) app_channelredirect-20080219.patch uploaded by johan (license 334) 2008-02-19 18:14 +0000 [r103818] Jeff Peeler * channels/chan_zap.c: (closes issue #11864) Reported by: julianjm Patches: chan_zap.c-1.4-devicestate-v1.diff uploaded by julianjm (license 99) Patch fixes problem of device state incorrectly reporting idle before PBX answers incoming call on FXO channel. Device status is updated now during new channel creation. 2008-02-19 17:33 +0000 [r103808-103813] Joshua Colp * /, configure, configure.ac: Merged revisions 103812 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103812 | file | 2008-02-19 13:31:32 -0400 (Tue, 19 Feb 2008) | 4 lines Don't look for launchd when cross compiling. (closes issue #12029) Reported by: ovi ........ 2008-02-19 00:59 +0000 [r103805] Tilghman Lesher * main/say.c: Change verbosity into debug for Hebrew (and various whitespace fixes) (Closes issue #12011) 2008-02-18 23:58 +0000 [r103798-103802] Joshua Colp * main/channel.c, /: Merged revisions 103801 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103801 | file | 2008-02-18 19:56:48 -0400 (Mon, 18 Feb 2008) | 10 lines Ensure that emulated DTMFs do not get interrupted by another begin frame. (closes issue #11740) Reported by: gserra Patches: v1-11740.patch uploaded by dimas (license 88) (closes issue #11955) Reported by: tsearle (closes issue #10530) Reported by: xmarksthespot ........ * main/channel.c, main/frame.c, channels/chan_sip.c, include/asterisk/channel.h, include/asterisk/frame.h: Add a non-invasive API for application level manipulation of T38 on a channel. This uses control frames (so they can even pass across IAX2) to negotiate T38 and provided a way of getting the current status of T38 using queryoption. This should by no means cause any issues and if it does I will take responsibility for it. (closes issue #11873) Reported by: dimas Patches: v4-t38-api.patch uploaded by dimas (license 88) * main/frame.c: Add some missing control frames. 2008-02-18 22:33 +0000 [r103796] Jason Parker * channels/chan_zap.c, /: Merged revisions 103795 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103795 | qwell | 2008-02-18 16:28:56 -0600 (Mon, 18 Feb 2008) | 1 line Fix previous commit so that we actually disable echocanbridged if echocancel is off. ........ 2008-02-18 21:57 +0000 [r103794] Matthew Fredrickson * channels/chan_zap.c: Commit chan_zap portion of #11964: add the ability to get ORIG_CALLED_NUM 2008-02-18 21:30 +0000 [r103791] Jason Parker * channels/chan_zap.c, /: Merged revisions 103790 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103790 | qwell | 2008-02-18 15:23:32 -0600 (Mon, 18 Feb 2008) | 4 lines Correct a message when echocancelwhenbridged is on, but echocancel is not. Closes issue #12019 ........ 2008-02-18 20:58 +0000 [r103788] Matthew Fredrickson * channels/chan_zap.c: Make sure EC is enabled when SS7 call comes in. Also add support for multiple DPCs per linkset. #11779 2008-02-18 20:53 +0000 [r103787] Mark Michelson * /, main/app.c: Merged revisions 103786 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103786 | mmichelson | 2008-02-18 14:52:09 -0600 (Mon, 18 Feb 2008) | 10 lines There was an invalid assumption when calculating the duration of a file that the filestream in question was created properly. Unfortunately this led to a segfault in the situation where an unknown format was specified in voicemail.conf and a voicemail was recorded. Now, we first check to be sure that the stream was written correctly or else assume a zero duration. (closes issue #12021) Reported by: jakep Tested by: putnopvut ........ 2008-02-18 19:47 +0000 [r103783] Michiel van Baak * main/asterisk.c: make the output of 'core show settings' a bit nicer. (closes issue #12020) Reported by: seanbright Patches: asterisk.c.patch uploaded by seanbright (license 71) 2008-02-18 17:45 +0000 [r103781] Tilghman Lesher * /, channels/chan_sip.c, main/rtp.c: Merged revisions 103780 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103780 | tilghman | 2008-02-18 11:31:52 -0600 (Mon, 18 Feb 2008) | 9 lines When a SIP channel is being auto-destroyed, it's possible for it to still be in bridge code. When that happens, we crash. Delay the RTP destruction until the bridge is ended. (closes issue #11960) Reported by: norman Patches: 20080215__bug11960__2.diff.txt uploaded by Corydon76 (license 14) Tested by: norman ........ 2008-02-18 Russell Bryant * Asterisk 1.6.0-beta3 released. 2008-02-18 17:12 +0000 [r103772] Olle Johansson * main/channel.c, channels/chan_sip.c: Make sure we can set up calls without audio (text+video). And ... it works! 2008-02-18 16:40 +0000 [r103771] Mark Michelson * channels/chan_zap.c, /: Merged revisions 103770 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103770 | mmichelson | 2008-02-18 10:37:31 -0600 (Mon, 18 Feb 2008) | 10 lines Fix a linked list corruption that under the right circumstances could lead to a looped list, meaning it will traverse forever. (closes issue #11818) Reported by: michael-fig Patches: 11818.patch uploaded by putnopvut (license 60) Tested by: michael-fig ........ 2008-02-18 16:13 +0000 [r103764-103769] Joshua Colp * apps/app_channelredirect.c, main/pbx.c, include/asterisk/pbx.h: Add an API call (ast_async_parseable_goto) which parses a goto string and does an async goto instead of an explicit goto. (closes issue #11753) Reported by: johan * /, channels/chan_sip.c: Merged revisions 103763 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103763 | file | 2008-02-18 11:33:14 -0400 (Mon, 18 Feb 2008) | 2 lines Don't care if the extension given doesn't exist for subscription based MWI. ........ 2008-02-18 10:10 +0000 [r103755] Olle Johansson * CHANGES, channels/chan_iax2.c: - No space in manager event names, please - Add new event to CHANGES 2008-02-18 04:43 +0000 [r103754] Tilghman Lesher * build_tools/cflags.xml, main/channel.c, main/pbx.c, funcs/func_channel.c, include/asterisk/channel.h, CHANGES, main/cli.c: Context tracing for channels (closes issue #11268) Reported by: moy Patches: chantrace-datastored-encapsulated-rev94934.patch uploaded by moy (license 222) 2008-02-16 21:22 +0000 [r103750] Michiel van Baak * channels/chan_skinny.c: move two ast_log calls to ast_debug. Now monitoring chan_skinny port with nagios or zabbix wont generate noise on the console. @ok tilghman 2008-02-15 23:32 +0000 [r103742] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 103741 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103741 | russell | 2008-02-15 17:31:39 -0600 (Fri, 15 Feb 2008) | 8 lines Fix a crash in chan_iax2 due to a race condition (closes issue #11780) Reported by: guillecabeza Patches: bug_iax2_jb_1.4.patch uploaded by guillecabeza (license 380) bug_iax2_jb_trunk.patch uploaded by guillecabeza (license 380) ........ 2008-02-15 23:20 +0000 [r103740] Mark Michelson * CHANGES: Document GotoIfTime change from svn revision 103738 2008-02-15 23:14 +0000 [r103739] Russell Bryant * include/asterisk/aes.h: Fix a regression in Asterisk 1.6 related to the use of AES encryption. 1024 was used instead of 128 when using AES from OpenSSL. Many thanks to d1mas for figuring this one out! (closes issue #11946) Reported by: bbhoss Patches: v1-11946.patch uploaded by dimas (license 88) 2008-02-15 23:07 +0000 [r103737-103738] Mark Michelson * main/pbx.c: Add proper "false" case behavior to GotoIfTime (closes issue #11719) Reported by: kshumard Patches: gotoiftime.twobranches.patch uploaded by kshumard (license 92) Tested by: kshumard * apps/app_voicemail.c: Fix redeclaration of variables when using IMAP storage (closes issue #11988) Reported by: jaroth Patches: variable_cleanup.patch uploaded by jaroth (license 50) 2008-02-15 19:50 +0000 [r103727-103729] Russell Bryant * /, main/loader.c: Merged revisions 103728 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103728 | russell | 2008-02-15 13:50:11 -0600 (Fri, 15 Feb 2008) | 4 lines In the case that you try to directly reload a module has returned AST_MODULE_LOAD_DECLINE, log a message indicating that the module is not fully initialized and must be initialized using "module load". ........ * /, main/loader.c: Merged revisions 103726 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103726 | russell | 2008-02-15 12:33:29 -0600 (Fri, 15 Feb 2008) | 6 lines Don't attempt to execute the reload callback for a module that returned AST_MODULE_LOAD_DECLINE. This fixes a crash that was reported against chan_console in trunk. (closes issue #11953, reported by junky, fixed by me) ........ 2008-02-15 17:32 +0000 [r103725] Mark Michelson * doc/tex/imapstorage.tex, /, configure, configure.ac: Merged revisions 103722 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103722 | mmichelson | 2008-02-15 11:26:37 -0600 (Fri, 15 Feb 2008) | 8 lines Final round of changes for configure script logic for IMAP Now if a directory is specified, then we will search that directory for a source installation of the IMAP toolkit. If none is found, then we will use that directory as the basis for detecting a package installation of the IMAP c-client. If that check fails, then configure will fail. ........ 2008-02-15 17:29 +0000 [r103723] Jason Parker * channels/chan_zap.c, channels/chan_sip.c, res/res_phoneprov.c, include/asterisk/extconf.h, channels/misdn/isdn_msg_parser.c, apps/app_queue.c, channels/misdn/isdn_lib.c, main/config.c, main/channel.c, res/res_config_curl.c, channels/misdn/isdn_lib.h, main/ast_expr2f.c, channels/misdn/ie.c, channels/misdn/chan_misdn_config.h, channels/misdn/portinfo.c, include/asterisk/strings.h, res/res_config_ldap.c, include/asterisk/time.h: Fix up some doxygen issues. (closes issue #11996) Patches: bug_11996_doxygen.diff uploaded by snuffy (license 35) 2008-02-15 15:45 +0000 [r103716] Tilghman Lesher * utils/conf2ael.c: Remove extraneous copy (closes issue #12002) Reported by: junky Patches: conf2ael.diff uploaded by junky (license 177) 2008-02-15 15:11 +0000 [r103699-103715] Mark Michelson * configure, configure.ac: Merging of changes from 1.4 revision 103713. * doc/tex/imapstorage.tex, configure, configure.ac: Same changes as made to 1.4 in revision 103710 * doc/tex/imapstorage.tex: Trunk version of 1.4's imap documentation updates * configure, configure.ac: See commit message for svn revision 103698. This behavior is the same as what is described there. The difference is that trunk already had the --with-imap=system option, but it only checked the include path for headers in the imap directory and not also the c-client directory. 2008-02-14 21:21 +0000 [r103694] Jason Parker * configure, include/asterisk/autoconfig.h.in, configure.ac: Modify ldap autoconf function, so that an incorrect ldap library is not found on Solaris (it is incompatible). Also removes second check for awk, which causes Solaris to find an incompatible version of awk. (closes issue #11829) Reported by: snuffy Patches: bug-11829.diff uploaded by snuffy (license 35) 2008-02-14 21:04 +0000 [r103687-103691] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 103690 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103690 | mmichelson | 2008-02-14 15:03:02 -0600 (Thu, 14 Feb 2008) | 3 lines Fix build for non-IMAP builds ........ * /, apps/app_voicemail.c: Merged revisions 103688 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103688 | mmichelson | 2008-02-14 14:55:48 -0600 (Thu, 14 Feb 2008) | 9 lines Fix the new message count if delete=yes when using IMAP storage. (closes issue #11406) Reported by: jaroth Patches: deleteflag_v2.patch uploaded by jaroth (license 50) Tested by: jaroth ........ * configs/queues.conf.sample, UPGRADE.txt, apps/app_queue.c: Change the queue holdtime announcement to happen at any interval (not just greater than two minutes). Remove the saying of less-than for holdtime announcements since it can lead to awkward holdtime announcements. Using '1' as a queue-round-seconds value is no longer valid. (closes issue #9736) Reported by: caio1982 Patches: queue_announce5.diff uploaded by caio1982 (license 22) Tested by: caio1982, putnopvut 2008-02-14 19:52 +0000 [r103685] Jason Parker * /, funcs/func_cdr.c: Merged revisions 103683 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103683 | qwell | 2008-02-14 13:51:10 -0600 (Thu, 14 Feb 2008) | 5 lines Document the 'l' option to the CDR() function. (Thanks voipgate for pointing out the option, and Leif for providing text for it.) Closes issue #11695. ........ 2008-02-14 19:47 +0000 [r103682] Jeff Peeler * apps/app_externalivr.c: a few syntax changes and safer code 2008-02-14 18:39 +0000 [r103677] Jason Parker * channels/chan_iax2.c: Add periodic jitter stats to CLI and manager. (closes issue #8188) Reported by: stevedavies Patches: jblogging-trunk.patch uploaded by stevedavies jblogging-trunk_wmgrevent.patch uploaded by johann8384 updated_jbloggin-trunk_mgrevent.patch uploaded by johann8384 (license 190) (with additional changes by me) Tested by: stevedavies, johann8384 2008-02-14 10:19 +0000 [r103668] Olle Johansson * res/res_agi.c, apps/app_externalivr.c: Formatting fixes 2008-02-13 21:04 +0000 [r103662] Jeff Peeler * apps/app_externalivr.c: (closes issue #11825) Reported by: ctooley Patches: additional_eivr_commands.patch uploaded by ctooley (license 136) Tested by: ctooley 2008-02-13 15:47 +0000 [r103658] Mark Michelson * UPGRADE.txt, res/res_musiconhold.c: 1. Deprecate SetMusicOnHold and WaitMusicOnHold. 2. Add a duration parameter to MusicOnHold (closes issue #11904) Reported by: dimas Patches: v2-moh.patch uploaded by dimas (license 88) Tested by: dimas 2008-02-13 00:55 +0000 [r103559] Mark Michelson * main/event.c: Fix a small logic error in ast_event_iterator_next. The previous logic allowed for the iterator to indicate there was more data than there really was, causing the iterator read beyond the end of the event structure. This led to invalid memory reads and potential crashes. 2008-02-12 22:26 +0000 [r103447-103506] Jason Parker * main/manager.c: Even more sane permissions. This should be handled via a umask, like in many other places. * main/manager.c: Use slight more sane permissions 2008-02-12 15:39 +0000 [r103387-103388] Russell Bryant * main/asterisk.c: Remove development version notice. * main/manager.c: Fix build on *BSD. These permissions constants are not available there. 2008-02-12 15:13 +0000 [r103386] Joshua Colp * /, channels/chan_sip.c: Merged revisions 103385 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103385 | file | 2008-02-12 11:09:24 -0400 (Tue, 12 Feb 2008) | 4 lines Even if no CallerID name or number has been provided by the remote party still use the configured sip.conf ones. (closes issue #11977) Reported by: pj ........ 2008-02-12 14:08 +0000 [r103341] Philippe Sultan * include/asterisk/jabber.h, res/res_jabber.c: Use an ast_flags structure in aji_client and aji_buddy rather than an integer. Modify calls to various ast_*_flag macros accordingly. 2008-02-12 00:24 +0000 [r103331] Jeff Peeler * main/manager.c, include/asterisk/config.h, CHANGES, main/config.c: Requested changes from Pari, reviewed by Russell. Added ability to retrieve list of categories in a config file. Added ability to retrieve the content of a particular category. Added ability to empty a context. Created new action to create a new file. Updated delete action to allow deletion by line number with respect to category. Added new action insert to add new variable to category at specified line. Updated action newcat to allow new category to be inserted in file above another existing category. 2008-02-11 22:10 +0000 [r103317-103325] Joshua Colp * /, apps/app_meetme.c: Merged revisions 103324 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103324 | file | 2008-02-11 18:09:07 -0400 (Mon, 11 Feb 2008) | 4 lines If entering a conference with the 'w' option ensure that we can't listen or speak until the marked user appears. (closes issue #11835) Reported by: alanmcmillan ........ * res/res_agi.c: Remove ast_module_user usage from res_agi. This is taken care of in the core. * main/pbx.c, main/manager.c, main/translate.c, main/logger.c, main/app.c, main/utils.c, main/indications.c, main/asterisk.c, main/rtp.c: Just some minor coding style cleanup... * main/pbx.c: Fix Manager Redirect while in an AGI. (closes issue #10661) Reported by: junky 2008-02-11 17:09 +0000 [r103316] Kevin P. Fleming * /, configs/zapata.conf.sample: Merged revisions 103315 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103315 | kpfleming | 2008-02-11 11:05:22 -0600 (Mon, 11 Feb 2008) | 2 lines improve 2BCT documentation a bit (thanks Jared) ........ 2008-02-11 16:17 +0000 [r103313-103314] Joshua Colp * main/channel.c, channels/chan_iax2.c: Add support for allowing a native bridge to happen when the L option is enabled. The RTP bridging could already handle this, it just needed to be enabled in the main bridging code. (issue #10647) Reported by: samdell3 * channels/chan_skinny.c: Change chan_skinny to use debug messages as appropriate. (closes issue #11967) Reported by: mvanbaak Patches: 2008021000-skinnydebug.diff.txt uploaded by mvanbaak (license 7) 2008-02-11 06:05 +0000 [r103306] James Golovich * channels/chan_sip.c: Don't wipe out transport and fd in chan_sip on reload (issue #11930) 2008-02-11 03:03 +0000 [r103282-103284] Mark Michelson * apps/app_queue.c: Fix improper indentation. Thanks again to snuffy for pointing it out. * apps/app_queue.c: Add a couple of comments to clarify the unreffing of queues. Thanks to snuffy for the idea. * main/event.c: Fix a problem regarding network vs. host byte order in the event API. ast_event_iterator_get_ie_type should return the ie type in host byte order. Furthermore, ast_event_get_ie_raw should already have its ie type argument in host byte order since it could be called externally (and it in fact is called in this way by ast_event_get_cached). 2008-02-09 11:27 +0000 [r103249] Michiel van Baak * apps/app_dial.c, apps/app_dictate.c, apps/app_echo.c, apps/app_authenticate.c, apps/app_disa.c, apps/app_chanisavail.c, apps/app_exec.c, apps/app_db.c, apps/app_controlplayback.c, apps/app_channelredirect.c, apps/app_directed_pickup.c, apps/app_dumpchan.c, apps/app_amd.c, apps/app_externalivr.c, apps/app_directory.c, apps/app_chanspy.c, apps/app_cdr.c: whitespace fixes only. 2008-02-09 06:33 +0000 [r103198] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 103197 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103197 | tilghman | 2008-02-09 00:23:49 -0600 (Sat, 09 Feb 2008) | 4 lines Commit fix for being unable to send voicemail from VoiceMailMain Reported by: William F Acker (via the -users mailing list) Patch by: Corydon76 (license 14) ........ 2008-02-08 21:26 +0000 [r103171] Russell Bryant * main/udptl.c, main/pbx.c, channels/chan_sip.c, channels/chan_iax2.c, res/res_jabber.c, apps/app_playback.c, main/rtp.c, channels/chan_usbradio.c, main/cdr.c, channels/chan_skinny.c, apps/app_minivm.c, res/res_agi.c, pbx/pbx_ael.c, pbx/pbx_dundi.c, funcs/func_devstate.c, apps/app_rpt.c, main/asterisk.c, channels/chan_mgcp.c, apps/app_voicemail.c: Merge changes from team/mvanbaak/cli-command-audit (closes issue #8925) About a year ago, as Leif Madsen and Jim van Meggelen were going over the CLI commands in Asterisk 1.4 for the next version of their book, they documented a lot of inconsistencies. This set of changes addresses all of these issues and has been reviewed by Leif. While this does introduce even more changes to the CLI command structure, it makes everything consistent, which is the most important thing. Thanks to all that helped with this one! 2008-02-08 18:58 +0000 [r103071-103122] Mark Michelson * apps/app_queue.c: Forgot that AST_LIST_REMOVE_CURRENT takes different arguments in trunk than 1.4. * /, apps/app_queue.c: Merged revisions 103120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103120 | mmichelson | 2008-02-08 12:48:17 -0600 (Fri, 08 Feb 2008) | 10 lines Prevent a potential three-thread deadlock. Also added a comment block to explicitly state the locking order necessary inside app_queue. (closes issue #11862) Reported by: flujan Patches: 11862.patch uploaded by putnopvut (license 60) Tested by: flujan ........ * /, channels/chan_iax2.c: Merged revisions 103070 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103070 | mmichelson | 2008-02-08 12:00:38 -0600 (Fri, 08 Feb 2008) | 6 lines Yield the thread and return -1 if the ioctl fails for Zaptel timing device. (closes issue #11891) Reported by: tzafrir ........ 2008-02-08 16:49 +0000 [r103044] Russell Bryant * UPGRADE-1.2.txt (added), UPGRADE-1.4.txt (added), UPGRADE.txt: At the request of ManxPower, include the UPGRADE.txt from 1.2 and 1.4, as well. This way, if people need to go back and review what was deprecated in previous major releases, it is readily available to them. Thanks for the suggestion! 2008-02-08 15:31 +0000 [r102969-103018] Joshua Colp * channels/chan_sip.c: Fix a network byte order issue and ensure when creating an outgoing dialog that the socket always contains information such as type and port. (closes issue #11916) Reported by: mnnojd * /, channels/chan_iax2.c: Merged revisions 102968 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r102968 | file | 2008-02-08 11:08:20 -0400 (Fri, 08 Feb 2008) | 4 lines Make sure the presence of dbsecret is factored into user scoring. (closes issue #11952) Reported by: bbhoss ........ 2008-02-07 21:37 +0000 [r102933] Mark Michelson * apps/app_chanspy.c: This is a combination new feature/bug fix for app_chanspy. New feature: Add the 'e' option, which takes as an argument a list of interfaces separated by colons. This way, you will only be able to spy on this limited list of interfaces. Bug fix: change some pointer checks to ast_strlen_zero so that spying would work properly even if no channel was specified as the first argument to chanspy. (closes issue #10072) Reported by: xmarksthespot Patches: bugfix+newfeature10072patchtotrunkrev102726.diff uploaded by xmarksthespot (license 16) Tested by: xmarksthespot, mvanbaak 2008-02-07 21:08 +0000 [r102906-102908] Michiel van Baak * apps/app_adsiprog.c: whitespace fixes only * apps/app_alarmreceiver.c: There she goes! First commit from me to trunk \o/ Make app_alarmreceiver honor code guidelines and fix whitespace errors. No functional changes. 2008-02-07 20:02 +0000 [r102859] Jason Parker * /, main/features.c: Merged revisions 102858 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r102858 | qwell | 2008-02-07 13:53:55 -0600 (Thu, 07 Feb 2008) | 7 lines Specify which digit string was matched in debug message. (closes issue #11949) Reported by: dimas Patches: v1-feature-debug.patch uploaded by dimas (license 88) ........ 2008-02-07 16:47 +0000 [r102808] Kevin P. Fleming * /, configs/zapata.conf.sample: Merged revisions 102807 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r102807 | kpfleming | 2008-02-07 10:41:55 -0600 (Thu, 07 Feb 2008) | 2 lines document usage of 'transfer' configuration option for ISDN PRI switch-side transfers ........ 2008-02-06 20:12 +0000 [r102777] Mark Michelson * apps/app_queue.c: Add the channel's unique id to the AgentCalled manager event to make it more consistent with other manager events. 2008-02-06 18:01 +0000 [r102726] Joshua Colp * /, channels/chan_sip.c: Merged revisions 102725 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r102725 | file | 2008-02-06 13:59:23 -0400 (Wed, 06 Feb 2008) | 2 lines Only consider a T.38-only INVITE compatible if we have both a joint capability between us and them and if they provided T.38. ........ 2008-02-06 16:23 +0000 [r102700] Terry Wilson * funcs/func_realtime.c: Add REALTIME_STORE and REALTIME_DESTROY dialplan functions provided by sergee. I just added the ability to set multiple fields at once after discussions with Tilghman and Russell. Currently limited to 30 fields. (closes issue #11887) Reported by: sergee Patches: rt-func-store-destroy-multivalue.diff uploaded by otherwiseguy (license 396) Tested by: sergee, otherwiseguy 2008-02-06 15:20 +0000 [r102652] Russell Bryant * /, configs/features.conf.sample: Merged revisions 102651 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r102651 | russell | 2008-02-06 09:19:41 -0600 (Wed, 06 Feb 2008) | 3 lines Clarify setting DYNAMIC_FEATURES so that it gets inherited by outbound channels. (due to a discussion between me and a user via email) ........ 2008-02-06 03:05 +0000 [r102602] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 102576 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r102576 | tilghman | 2008-02-05 18:26:02 -0600 (Tue, 05 Feb 2008) | 4 lines Move around some defines to unbreak ODBC storage. (closes issue #11932) Reported by: snuffy ........ 2008-02-06 00:08 +0000 [r102501-102550] Mark Michelson * apps/app_queue.c: Remove an extra debug message I left in * channels/chan_unistim.c, apps/app_dial.c, main/pbx.c, apps/app_privacy.c, apps/app_alarmreceiver.c, res/res_jabber.c, apps/app_followme.c, main/loader.c, channels/chan_usbradio.c, main/tcptls.c, res/res_agi.c, apps/app_minivm.c, apps/app_dumpchan.c, main/logger.c, apps/app_zapras.c, main/astmm.c: Get rid of any remaining ast_verbose calls in the code in favor of ast_verb (closes issue #11934) Reported by: mvanbaak Patches: 20080205_astverb-2.diff.txt uploaded by mvanbaak (license 7) * apps/app_voicemail.c: Change verbose messages to use the ast_verb macro. (closes issue #11931) Reported by: snuffy Patches: bug-11931.diff uploaded by snuffy (license 35) 2008-02-05 20:51 +0000 [r102500] Jason Parker * main/pbx.c: Change where priority of a goto is adjusted. Partially reverts 102272. Closes issue #11929 (credit to file for fix suggestion - we still <3 you) 2008-02-05 20:03 +0000 [r102454] Mark Michelson * /, channels/chan_mgcp.c: Merged revisions 102453 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r102453 | mmichelson | 2008-02-05 14:02:44 -0600 (Tue, 05 Feb 2008) | 8 lines Clear the DTMF buffer on hangup. (closes issue #11919) Reported by: eferro Patches: mgcp_dtmfclean_on_hangup.diff uploaded by eferro (license 337) Tested by: eferro ........ 2008-02-05 19:58 +0000 [r102379-102452] Joshua Colp * channels/chan_sip.c: Yeah yeah, I broke building on trunk. Shoot me. * /, channels/chan_sip.c: Merged revisions 102450 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r102450 | file | 2008-02-05 15:52:30 -0400 (Tue, 05 Feb 2008) | 3 lines If a REGISTER attempt comes in that is a retransmission of a previous REGISTER do not create a new nonce value. (issue #BE-381) ........ * /, res/res_clioriginate.c: Merged revisions 102378 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r102378 | file | 2008-02-05 11:09:29 -0400 (Tue, 05 Feb 2008) | 4 lines Perform dialing asynchronously when using the originate CLI command so the CLI does not appear to block. (closes issue #11927) Reported by: bbhoss ........ 2008-02-04 21:15 +0000 [r102329] Tilghman Lesher * utils/muted.c, /, configure, include/asterisk/autoconfig.h.in, configure.ac, main/asterisk.c: Merged revisions 102323 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r102323 | tilghman | 2008-02-04 15:06:09 -0600 (Mon, 04 Feb 2008) | 7 lines Cross-platform fix: OS X now deprecates the use of the daemon(3) API. (closes issue #11908) Reported by: oej Patches: 20080204__bug11908.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ 2008-02-04 18:39 +0000 [r102297] Jason Parker * channels/chan_zap.c: Add line numbers to warning/error messages (and pretty up some existing ones). (closes issue #11894) Reported by: jmls Patches: chan_zap.patch uploaded by jmls (license 141) 2008-02-04 15:16 +0000 [r102272] Joshua Colp * main/pbx.c: Update handling of asyncgoto so it properly works on channels that are currently executing a PBX. (closes issue #11914) Reported by: arnd (closes issue #11753) Reported by: johan 2008-02-04 14:37 +0000 [r102262] Jason Parker * configs/extensions.ael.sample, configs/extensions.lua.sample: Change examples to use G here also. Closes issue #11875 2008-02-04 05:32 +0000 [r102190-102238] Tilghman Lesher * /, funcs/func_strings.c: Merged revisions 102214 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r102214 | tilghman | 2008-02-03 23:10:02 -0600 (Sun, 03 Feb 2008) | 6 lines Missing braces. (closes issue #11912) Reported by: dimas Patches: sprintf.patch uploaded by dimas (license 88) ........ * main/manager.c: CoreSettings and CoreStatus are missing the terminating "\r\n". Also, some miscellaneous spacing and initialization issues. (closes issue #11909) Reported by: srt Patches: patch-11909-2.diff uploaded by srt (license 378) Tested by: srt 2008-02-03 16:46 +0000 [r102091-102143] Olle Johansson * /, channels/chan_sip.c: Merged revisions 102142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r102142 | oej | 2008-02-03 17:38:12 +0100 (Sön, 03 Feb 2008) | 8 lines Use the same CSEQ on CANCEL as on INVITE (according to RFC 3261) (closes issue #9492) Reported by: kryptolus Patches: bug9492.txt uploaded by oej (license 306) Tested by: oej ........ * /, channels/chan_sip.c: Merged revisions 102090 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r102090 | oej | 2008-02-03 11:37:32 +0100 (Sön, 03 Feb 2008) | 8 lines Handle ACK and CANCEL in an invite transaction - even if we get INFO transactions during the actual call setup. (closes issue #10567) Reported by: jacksch Tested by: oej Patch by: oej inspired by suggestions from neutrino88 in the bug tracker ........ 2008-02-03 06:43 +0000 [r102064] Russell Bryant * configure, configure.ac: Change the version number in the configure script from 1.4 to 1.6 2008-02-02 06:10 +0000 [r101990-102037] Russell Bryant * include/asterisk/event.h: The documentation page has to be in its own comment block to work, apparently. Fix it up! * /, channels/chan_sip.c: Merged revisions 101989 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101989 | russell | 2008-02-01 17:06:32 -0600 (Fri, 01 Feb 2008) | 5 lines Change the SDP_SAMPLE_RATE macro. It turns out that even though G.722 is 16 kHz, it is supposed to specified as 8 kHz in the RTP, and RTP timestamps are supposed to be calculated based on 8 kHz. (Apparently this is due to a bug in a spec, but people follow it anyway, because it's the spec ...) ........ 2008-02-01 22:12 +0000 [r101873-101943] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 101942 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101942 | tilghman | 2008-02-01 15:54:28 -0600 (Fri, 01 Feb 2008) | 8 lines Fix the VM_DUR variable for forwarded voicemail, and fixed several other bugs while I'm in the area. (closes issue #11615) Reported by: jamessan Patches: 20071226__bug11615__2.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76, jamessan ........ * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 101894 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101894 | tilghman | 2008-02-01 13:36:12 -0600 (Fri, 01 Feb 2008) | 2 lines Change detection of getifaddrs to use AST_C_COMPILE_CHECK, backported from trunk (as suggested by kpfleming) ........ * res/res_config_curl.c: Fix multi, when using the LIKE query. (closes issue #11889) Reported by: jmls Patches: res_config_curl.patch uploaded by jmls (license 141) Tested by: jmls 2008-02-01 18:24 +0000 [r101869] Jason Parker * apps/app_authenticate.c: Comparison, not set :) Thanks mvanbaak. 2008-02-01 18:08 +0000 [r101824] Tilghman Lesher * res/res_odbc.c, configs/res_odbc.conf.sample: Clarify the pooling functionality by changing the config file keyword 2008-02-01 17:44 +0000 [r101823] Jason Parker * /, apps/app_authenticate.c: Move an feof() call to before the fgets(). This would have exited the loop early if you had an authentication file with no newline at the end. 2008-02-01 17:28 +0000 [r101819-101821] Russell Bryant * /, apps/app_authenticate.c: Merged revisions 101818 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101818 | russell | 2008-02-01 11:23:47 -0600 (Fri, 01 Feb 2008) | 4 lines Don't overwrite the last character of a line if it's not a newline. This would happen if the last line in the file doesn't have a newline. (pointed out by Qwell) ........ 2008-02-01 16:01 +0000 [r101773] Tilghman Lesher * /, configure, include/asterisk/autoconfig.h.in, configure.ac, main/acl.c: Merged revisions 101772 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101772 | tilghman | 2008-02-01 09:55:58 -0600 (Fri, 01 Feb 2008) | 2 lines Compatibility fix for OpenWRT (reported by Brian Capouch via the mailing list) ........ 2008-02-01 06:27 +0000 [r101694-101746] Russell Bryant * apps/app_authenticate.c: simplify some code, tweak formatting, and reduce indentation * apps/app_authenticate.c: reduce a level of indentation * apps/app_channelredirect.c: Get rid of a goto where there was no extra cleanup happening at the exit point * /, channels/chan_iax2.c: Merged revisions 101693 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101693 | russell | 2008-01-31 18:32:49 -0600 (Thu, 31 Jan 2008) | 8 lines Add some more sanity checking on IAX2 dial strings for the case that no peer or hostname was provided, which is the one part of the dial string that is absolutely required. If it's not there, bail out. (closes issue #11897) Reported by sokhapkin Patch by me ........ 2008-02-01 00:08 +0000 [r101650] Mark Michelson * /, apps/app_amd.c: Merged revisions 101649 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101649 | mmichelson | 2008-01-31 18:06:37 -0600 (Thu, 31 Jan 2008) | 9 lines From bugtracker: "fix totalAnalysisTime to handle periods of no channel activity" (closes issue #9256) Reported by: cmaj Patches: amd-dont-wait-too-long-for-frames-take3.diff.txt uploaded by cmaj (license 111) Tested by: cmaj, skygreg, ZX81, rjain ........ 2008-01-31 23:14 +0000 [r101611] Russell Bryant * /, main/translate.c, main/file.c: Merged revisions 101601 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101601 | russell | 2008-01-31 17:10:06 -0600 (Thu, 31 Jan 2008) | 12 lines Fix a couple of places where ast_frfree() was not called on a frame that came from a translator. This showed itself by g729 decoders not getting released. Since the flag inside the translator frame never got unset by freeing the frame to indicate it was no longer in use, the translators never got destroyed, and thus the g729 licenses were not released. (closes issue #11892) Reported by: xrg Patches: 11892.diff uploaded by russell (license 2) Tested by: xrg, russell ........ 2008-01-31 22:12 +0000 [r101578-101580] Mark Michelson * apps/app_queue.c: Forgot an ! * apps/app_queue.c: A change I made to accommodate the "linear" strategy in trunk caused queue strategies to not be loaded from realtime queues. This commit fixes that. Thanks to jmls for pointing this problem out to me on IRC. This also contains some changes to S_OR where it should be used. Thanks to Qwell for pointing these out. 2008-01-31 21:33 +0000 [r101577] Russell Bryant * channels/chan_sip.c: Fix a simple deadlock that was introduced _right_ before this code got merged into trunk. (closes issue #11895, reported by pj, patched by me) 2008-01-31 21:31 +0000 [r101532-101576] Mark Michelson * apps/app_queue.c: Handle the case of a NULL state_interface when checking a realtime member. Thanks to jmls for finding this issue. * /, res/res_monitor.c: Merged revisions 101531 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101531 | mmichelson | 2008-01-31 15:00:24 -0600 (Thu, 31 Jan 2008) | 10 lines 1. Prevent the addition of an extra '/' to the beginning of an absolute pathname. 2. If ast_monitor_change_fname is called and the new filename is the same as the old, then exit early and don't set the filename_changed field in the monitor structure. Setting it in this case was causing ast_monitor_stop to erroneously delete them. (closes issue #11741) Reported by: garlew Tested by: putnopvut ........ 2008-01-31 19:54 +0000 [r101483] Jason Parker * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions 101482 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101482 | qwell | 2008-01-31 13:52:49 -0600 (Thu, 31 Jan 2008) | 4 lines Solaris compat fixes for struct in_addr funkiness. Issue #11885, patch by snuffy. ........ 2008-01-31 19:43 +0000 [r101481] Steve Murphy * main/pbx.c, /: Merged revisions 101480 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101480 | murf | 2008-01-31 12:30:37 -0700 (Thu, 31 Jan 2008) | 1 line closes issue #11845; that's the one where there's a 1004 byte cdr leak with every AMI Redirect to a zap channel ........ 2008-01-31 19:20 +0000 [r101416-101449] Russell Bryant * /, channels/chan_agent.c: Merged revisions 101433 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101433 | russell | 2008-01-31 13:17:05 -0600 (Thu, 31 Jan 2008) | 2 lines Add more missing locking of the agents list ... ........ * /, channels/chan_agent.c: Merged revisions 101413-101414 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101413 | russell | 2008-01-31 13:04:52 -0600 (Thu, 31 Jan 2008) | 2 lines Add missing locking to the find_agent() function. ........ r101414 | russell | 2008-01-31 13:07:46 -0600 (Thu, 31 Jan 2008) | 3 lines Move the locking from find_agent() into the agent dialplan function handler to ensure that the agent doesn't disappear while we're looking at it. ........ 2008-01-31 15:36 +0000 [r101393] Joshua Colp * funcs/func_realtime.c: Add missing braces. (closes issue #11886) Reported by: sergee Patches: func_realtime_fix-r101392.diff uploaded by sergee (license 138) 2008-01-31 05:28 +0000 [r101373] Russell Bryant * CHANGES: remove entry that is no longer in the tree 2008-01-30 23:10 +0000 [r101344] Mark Michelson * channels/chan_sip.c: The deprecation of "username" in favor of "defaultuser" for SIP peers unfortunately broke realtime configurations which still used the "username" field. This was taken care of properly when reading from realtime but was not handled properly when updating a realtime peer. This change also adds a deprecation NOTICE CLI message that will print if using the deprecated "username" field. (closes issue #11880) Reported by: cabal95 Patches: 11880.patch uploaded by putnopvut (license 60) Tested by: cabal95 2008-01-30 20:08 +0000 [r101322] Olle Johansson * configs/cli.conf.sample: Clarify configuration file that can be misunderstood 2008-01-30 19:03 +0000 [r101296] Jason Parker * apps/app_controlplayback.c: Allow disabling the default ffwd/rewind keys in the ControlPlayback application. This is done in a backward compat way. If the "default" key for ffwd/rew is used for another option (such as stop), the "default" is removed. (closes issue #11754) Reported by: johan Patches: app_controlplayback.c.option3.patch uploaded by johan (license 334) Tested by: johan, qwell 2008-01-30 17:12 +0000 [r101271] Olle Johansson * configs/rtppage.conf.sample (removed), apps/app_rtppage.c (removed): Removing applications that wasn't ready for svn trunk, as trunk now has pre-release status. 2008-01-30 16:54 +0000 [r101269] Tilghman Lesher * apps/app_voicemail.c: Make the VoicemailUsersList AMI command consistent with other manager list functions. (closes issue #11874) Reported by: srt Patches: voicemail_ami-11847.patch uploaded by srt (license 378) 2008-01-30 16:39 +0000 [r101267-101268] Olle Johansson * include/asterisk/rtp.h, main/rtp.c: - doxygen fixes - change function to void because it always returned the same value and no one read it. * main/rtp.c: Formatting fixes 2008-01-30 15:42 +0000 [r101224] Mark Michelson * apps/app_rtppage.c: Get trunk to compile 2008-01-30 15:42 +0000 [r101223] Joshua Colp * /, main/slinfactory.c: Merged revisions 101222 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101222 | file | 2008-01-30 11:41:04 -0400 (Wed, 30 Jan 2008) | 4 lines Fix an issue where if a frame of higher sample size preceeded a frame of lower sample size and ast_slinfactory_read was called with a sample size of the combined values or higher a crash would happen. (closes issue #11878) Reported by: stuarth ........ 2008-01-30 15:36 +0000 [r101221] Olle Johansson * CHANGES: Update CHANGES with rtppage 2008-01-30 15:35 +0000 [r101220] Jason Parker * /, configs/extensions.conf.sample: Merged revisions 101219 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11875) ........ r101219 | qwell | 2008-01-30 09:34:37 -0600 (Wed, 30 Jan 2008) | 5 lines Change default config to use descending channel order of groups, rather than ascending. Fixes a potential source of confusion in glare-type situations. Issue 11875, reported by JimVanM. ........ 2008-01-30 15:30 +0000 [r101218] Olle Johansson * configs/rtppage.conf.sample (added), apps/app_rtppage.c (added): Add rtppage() application to do multicast or unicast RTP paging to SIP phones. (closes issue #11797) Reported by: macbrody Patches: app_rtppage-20080130.c uploaded by macbrody (license 352) 2008-01-30 15:27 +0000 [r101217] Mark Michelson * /, apps/app_queue.c: Merged revisions 101216 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101216 | mmichelson | 2008-01-30 09:23:00 -0600 (Wed, 30 Jan 2008) | 5 lines Fix a logic error with regards to autofill. Prior to this change, it was possible for a caller to go out of turn if autofill were enabled and callers ahead in the queue were attempting to call a member. This change fixes this. ........ 2008-01-30 12:48 +0000 [r101196] Kevin P. Fleming * channels/chan_sip.c: simplify this code and eliminate the return value cast that is no longer necessary 2008-01-30 11:27 +0000 [r101153-101154] Olle Johansson * channels/chan_sip.c, include/asterisk/channel.h: Constifying the interface to get pvt_ids in the bridge, based on suggestion from (const char *) Kevin. Thanks! * /, channels/chan_sip.c: Merged revisions 101152 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101152 | oej | 2008-01-30 12:20:31 +0100 (Ons, 30 Jan 2008) | 7 lines Stop musiconhold on attended transfer. (closes issue #11872) Reported by: gareth Patches: svn-101018.patch uploaded by gareth (license 208) ........ 2008-01-30 00:58 +0000 [r101126] Jason Parker * CHANGES: Fix a typo 2008-01-30 00:04 +0000 [r101082] Russell Bryant * CHANGES, apps/app_speech_utils.c: Add the 'n' option to SpeechBackground, which has the application not answer the channel if it has not already been answered. (closes SPD-51) 2008-01-29 23:59 +0000 [r101081] Dwayne M. Hubbard * /, build_tools/make_version: Merged revisions 101080 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101080 | dhubbard | 2008-01-29 17:50:42 -0600 (Tue, 29 Jan 2008) | 1 line updated build_tools to handle the autotag directory structure changes; changes related to BE-353. Patch by The Russell and reviewed by The Me. ........ 2008-01-29 23:02 +0000 [r101036] Mark Michelson * /, apps/app_queue.c: Merged revisions 101035 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101035 | mmichelson | 2008-01-29 17:02:03 -0600 (Tue, 29 Jan 2008) | 3 lines Remove a memory leak from updating realtime queues ........ 2008-01-29 22:04 +0000 [r101018] Tilghman Lesher * res/res_config_curl.c: Oops, a sizeof error 2008-01-29 19:41 +0000 [r100974] Mark Michelson * /, apps/app_queue.c: Merged revisions 100973 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100973 | mmichelson | 2008-01-29 13:39:00 -0600 (Tue, 29 Jan 2008) | 6 lines Fixing an erroneous return value returned when attempting to pause or unpause a queue member fails. Fixes BE-366, thanks to John Bigelow for writing the patch. ........ 2008-01-29 17:44 +0000 [r100933] Russell Bryant * /, main/Makefile: Merged revisions 100932 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100932 | russell | 2008-01-29 11:43:41 -0600 (Tue, 29 Jan 2008) | 4 lines Fix the last couple of issues related to building from a path that contains spaces. (closes issue #11834) ........ 2008-01-29 17:42 +0000 [r100931] Jason Parker * /, channels/misdn_config.c: Merged revisions 100930 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100930 | qwell | 2008-01-29 11:41:43 -0600 (Tue, 29 Jan 2008) | 6 lines Initialize an array to 0s if config option not specified. (closes issue #11860) Patches: misdn_get_config.v1.diff uploaded by IgorG (license 20) ........ 2008-01-29 17:22 +0000 [r100900-100928] Russell Bryant * Makefile, /: Merged revisions 100922 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100922 | russell | 2008-01-29 11:21:33 -0600 (Tue, 29 Jan 2008) | 3 lines Use GNU make magic instead of shell magic to escape spaces in the working directory. (related to issue #11834) ........ * Makefile, /: Merged revisions 100882 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100882 | russell | 2008-01-29 11:06:43 -0600 (Tue, 29 Jan 2008) | 6 lines Fix building Asterisk when the working path has spaces in it. (closes issue #11834) Reported by: spendergrass Patched by: me ........ 2008-01-29 16:14 +0000 [r100843] Jason Parker * channels/chan_zap.c, /: Merged revisions 100835 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100835 | qwell | 2008-01-29 10:10:00 -0600 (Tue, 29 Jan 2008) | 5 lines Allow zap groups above 30 to work properly. (closes issue #11590) Reported by: tbsky ........ 2008-01-29 15:30 +0000 [r100833] Joshua Colp * channels/chan_sip.c: Make externip work as documented. If no port is specified it will use the value of bindport instead of always being 5060. (closes issue #11858) Reported by: hmodes 2008-01-29 10:50 +0000 [r100794-100795] Christian Richter * channels/chan_misdn.c, /: Merged revisions 100793 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100793 | crichter | 2008-01-29 11:36:19 +0100 (Di, 29 Jan 2008) | 1 line fixed potential segfault in misdn show channels CLI command ........ * channels/chan_misdn.c, /: Merged revisions 96199 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r96199 | crichter | 2008-01-03 13:12:27 +0100 (Do, 03 Jan 2008) | 1 line make sure frame is completely clean, before we send it to asterisk as DTMF. If we don't make it clean, it happens that one way audio occurs.. ........ 2008-01-29 09:18 +0000 [r100741-100767] Olle Johansson * /, channels/chan_sip.c: Merged revisions 100740 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100740 | oej | 2008-01-29 09:26:48 +0100 (Tis, 29 Jan 2008) | 8 lines (closes issue #11736) Reported by: MVF Patches: bug11736-2.diff uploaded by oej (license 306) Tested by: oej, MVF, revolution (russellb: This was the showstopper for the release.) ........ * channels/chan_sip.c: Removing code that wasn't supposed to be there at all, only at an experimental stage before I found another solution. Thanks Kevin, for reminding me. 2008-01-28 Russell Bryant * Asterisk 1.6.0-beta2 released. 2008-01-28 21:11 +0000 [r100679] Jason Parker * build_tools/menuselect-deps.in, configs/vpb.conf.sample (added), doc/tex/channelvariables.tex, makeopts.in: Reintroduce more chan_vpb stuff that was removed in r100421 and r100422 2008-01-28 21:07 +0000 [r100678] Mark Michelson * channels/chan_vpb.cc (added), configure, include/asterisk/autoconfig.h.in, configure.ac, channels/Makefile: Re-inserting chan_vpb into trunk. 2008-01-28 21:05 +0000 [r100677] Tilghman Lesher * main/pbx.c, /: Merged revisions 100675 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100675 | tilghman | 2008-01-28 15:02:02 -0600 (Mon, 28 Jan 2008) | 2 lines WaitExten didn't handle AbsoluteTimeout properly (went to 't' instead of 'T') ........ 2008-01-28 21:02 +0000 [r100676] Jason Parker * /, apps/app_voicemail.c: Merged revisions 100672 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11795) ........ r100672 | qwell | 2008-01-28 14:42:43 -0600 (Mon, 28 Jan 2008) | 7 lines When using ODBC_STORAGE, make sure we put greeting files into the database like we do with the others. Issue #11795 Reported by: dimas Patches: vmgreet.patch uploaded by dimas (license 88) ........ 2008-01-28 20:40 +0000 [r100632-100671] Joshua Colp * channels/chan_sip.c: Fix up some T38 state change issues. (closes issue #11630) Reported by: dimas Patches: v2-sip-t38state.patch uploaded by dimas (license 88) * channels/chan_sip.c: Fix up two scheduling issues. In one instance a scheduled item was not deleted when it should have been and in the other it was scheduled again when it shouldn't have been. 2008-01-28 18:41 +0000 [r100630-100631] Russell Bryant * main/features.c: Merge rev 100626 from Asterisk 1.4. The svnmerge of this commit was a NoOp, since res_features doesn't exist in trunk. Thanks to qwell for pointing it out! * /, channels/chan_sip.c: Merged revisions 100629 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100629 | russell | 2008-01-28 12:34:20 -0600 (Mon, 28 Jan 2008) | 5 lines For some reason, the use of this strdupa() is leading to memory corruption on freebsd sparc64. This trivial workaround fixes it. (closes issue #10300, closes issue #11857, reported by mattias04 and Home-of-the-Brave) ........ 2008-01-28 18:27 +0000 [r100628] Tilghman Lesher * configure, include/asterisk/autoconfig.h.in, configure.ac, main/logger.c: Normalize the detection for execinfo, so that Linux (glibc) and other platforms with libexecinfo will generate inline stack backtraces correctly. 2008-01-28 18:27 +0000 [r100627] Russell Bryant * /: Merged revisions 100626 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100626 | russell | 2008-01-28 12:26:31 -0600 (Mon, 28 Jan 2008) | 7 lines Fix a crash in ast_masq_park_call() (issue #11342) Reported by: DEA Patches: res_features-park.txt uploaded by DEA (license 3) ........ 2008-01-28 18:24 +0000 [r100625] Jason Parker * channels/chan_zap.c, /: Merged revisions 100624 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100624 | qwell | 2008-01-28 12:23:09 -0600 (Mon, 28 Jan 2008) | 1 line Correct a comment which made little/no sense. ........ 2008-01-28 17:21 +0000 [r100565-100582] Russell Bryant * main/channel.c, channels/chan_local.c, /, include/asterisk/channel.h: Merged revisions 100581 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100581 | russell | 2008-01-28 11:15:41 -0600 (Mon, 28 Jan 2008) | 9 lines Make some deadlock related fixes. These bugs were discovered and reported internally at Digium by Steve Pitts. - Fix up chan_local to ensure that the channel lock is held before the local pvt lock. - Don't hold the channel lock when executing the timing function, as it can cause a deadlock when using chan_local. This actually changes the code back to be how it was before the change for issue #10765. But, I added some other locking that I think will prevent the problem reported there, as well. ........ * main/pbx.c: Clean up some formatting, and simplify a bit of code using ast_str 2008-01-28 13:57 +0000 [r100549] Joshua Colp * channels/chan_sip.c: Don't do a network byte order conversion when setting the socket's port variable to that of bindaddr's. It is already in the correct network byte order. (closes issue #11800) Reported by: hmodes 2008-01-28 04:43 +0000 [r100514-100533] Russell Bryant * main/channel.c: Make a couple more uses of ARRAY_LEN, and convert some spaces to tabs * main/channel.c: - Simplify a line with ARRAY_LEN() - Make a few little formatting changes * main/channel.c: These readlocks always fail for me on my mac, and I saw it happen again today on another mac. We ignore the return value of locking operations almost everywhere in Asterisk. So, ignore these, as well, so Asterisk will actually work on systems where this is occurring while I look into what the issue is. 2008-01-27 23:14 +0000 [r100488-100497] Tilghman Lesher * channels/chan_sip.c, include/asterisk/sched.h, channels/chan_iax2.c: With the switch to the ast_sched_replace* API in trunk, we lose the correction that was just merged from 1.4, so this is a changeover to those APIs to use the macro versions, so that we properly detect errors from ast_sched_del, instead of simply ignoring the return values. * main/cdr.c, channels/chan_misdn.c, main/dnsmgr.c, /, channels/chan_sip.c, channels/chan_h323.c, include/asterisk/sched.h, main/file.c, pbx/pbx_dundi.c, channels/chan_iax2.c, main/rtp.c, channels/chan_mgcp.c: Merged revisions 100465 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100465 | tilghman | 2008-01-27 15:59:53 -0600 (Sun, 27 Jan 2008) | 11 lines When deleting a task from the scheduler, ignoring the return value could possibly cause memory to be accessed after it is freed, which causes all sorts of random memory corruption. Instead, if a deletion fails, wait a bit and try again (noting that another thread could change our taskid value). (closes issue #11386) Reported by: flujan Patches: 20080124__bug11386.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76, flujan, stuarth` ........ 2008-01-25 22:54 +0000 [r100421-100422] Jason Parker * doc/tex/channelvariables.tex: Get rid of that last little bit. * build_tools/menuselect-deps.in, configs/vpb.conf.sample (removed), makeopts.in: Remove more remnants of chan_vpb 2008-01-25 22:39 +0000 [r100419-100420] Mark Michelson * channels/chan_vpb.cc (removed), configure, include/asterisk/autoconfig.h.in, configure.ac, channels/Makefile, .cleancount: Removing chan_vpb from the tree 2008-01-25 21:26 +0000 [r100379] Jason Parker * /, channels/chan_sip.c: Merged revisions 100378 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100378 | qwell | 2008-01-25 15:24:49 -0600 (Fri, 25 Jan 2008) | 2 lines This would have never been true, since we're passing (sizeof(req.data) - 1) as the len to recvfrom(). ........ 2008-01-25 20:51 +0000 [r100361] Kevin P. Fleming * apps/app_rpt.c: correct a real problem and silence an annoying compiler warning 2008-01-25 14:53 +0000 [r100344] Mark Michelson * apps/app_queue.c: Insure that we are not going to pass a NULL pointer to add_to_interfaces. (closes issue #11840) Reported by: junky 2008-01-25 02:52 +0000 [r100325] Joshua Colp * main/dial.c, include/asterisk/dial.h: Add an API call that steals the answered channel so that a destruction of the dialing structure does not hang it up. 2008-01-24 22:58 +0000 [r100307] Tilghman Lesher * Makefile, build_tools/make_defaults_h: Use the set ASTDBDIR as the default, too 2008-01-24 22:36 +0000 [r100305-100306] Kevin P. Fleming * include/asterisk/app.h: ummm... might be good if this macro argument was actually used :-) * include/asterisk/app.h: add the ability to define a structure type for argument parsing when it would be useful to be able to pass it between functions 2008-01-24 22:02 +0000 [r100266] James Golovich * channels/chan_sip.c: Fix simple whitespace issue 2008-01-24 22:01 +0000 [r100265] Kevin P. Fleming * include/asterisk/app.h, /: Merged revisions 100264 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100264 | kpfleming | 2008-01-24 15:57:41 -0600 (Thu, 24 Jan 2008) | 2 lines make these macros not assume that the only other field in the structure is 'argc'... this is true when someone uses AST_DECLARE_APP_ARGS, but it's perfectly reasonable to define your own structure as long as it has the right fields ........ 2008-01-24 20:32 +0000 [r100245] Joshua Colp * main/features.c: Minor cosmetic change... 2008-01-24 18:35 +0000 [r100224] James Golovich * main/astmm.c: Increase the size of filenames stored when astmm is used. If the path length was long they would be truncated and grouped together with whatever matches 2008-01-24 17:47 +0000 [r100206] Joshua Colp * configs/rtp.conf.sample, CHANGES, main/rtp.c: Merge in strictrtp branch. This adds a strictrtp option to rtp.conf which drops packets that do not come from the remote party. (closes issue #8952) Reported by: amorsen 2008-01-24 17:24 +0000 [r100169] Russell Bryant * /, main/asterisk.c: Merged revisions 100164 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100164 | russell | 2008-01-24 11:22:09 -0600 (Thu, 24 Jan 2008) | 2 lines Update main Asterisk copyright info to 2008 ........ 2008-01-24 16:47 +0000 [r100121-100139] Jason Parker * /, res/res_phoneprov.c, main/acl.c: Merged revisions 100138 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100138 | qwell | 2008-01-24 10:41:29 -0600 (Thu, 24 Jan 2008) | 6 lines Fix compilation on Solaris. (closes issue #11832) Patches: bug-11832.diff uploaded by snuffy (license 35) ........ * channels/chan_sip.c, main/features.c: Move chan_local dependency into places (only one) that previously depended on res_features, and used local channels 2008-01-24 15:54 +0000 [r100076-100112] Joshua Colp * channels/chan_zap.c, channels/chan_sip.c, channels/chan_iax2.c, channels/chan_mgcp.c: Remove dependency on res_features from some channel drivers. It is now part of the core and no longer exists as a module. * main/channel.c: Some more cosmetic changes. * main/channel.c: Add some spacing. * main/dial.c: Test hopefully over. * main/dial.c: Testing something... 2008-01-24 00:04 +0000 [r100057] Kevin P. Fleming * channels/chan_sip.c: fix flag bit definitions to make code from issue #11049 actually work; along the way, clarify comments and add some dummy flag definitions for other multi-bit flags to hopefully stop this from happening in the future (closes issue #11049) 2008-01-23 23:09 +0000 [r100039] Jason Parker * res/res_features.c (removed), main/Makefile, main/features.c (added), include/asterisk/_private.h, CHANGES, .cleancount, main/asterisk.c, main/loader.c, include/asterisk/features.h: Move code from res_features into (new file) main/features.c 2008-01-23 22:00 +0000 [r100021] Russell Bryant * CREDITS: Add Sergey Tamkovich to CREDITS. Thank you for your contributions! 2008-01-23 21:11 +0000 [r99979-99980] Olle Johansson * /, channels/chan_sip.c: Merged revisions 99978 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99978 | oej | 2008-01-23 22:07:16 +0100 (Ons, 23 Jan 2008) | 7 lines Second attempt. Don't change invitestate when receiving 18x messages in CANCEL state. (issue #11736) Reported by: MVF Patch by oej. ........ * /, channels/chan_sip.c: Merged revisions 99977 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99977 | oej | 2008-01-23 21:58:20 +0100 (Ons, 23 Jan 2008) | 9 lines Make sure we don't cancel destruction on calls in CANCEL state, even if we get 183 while waiting for answer on our CANCEL. (issue #11736) Reported by: MVF Patches: bug11736.txt uploaded by oej (license 306) Tested by: MVF ........ 2008-01-23 20:26 +0000 [r99976] Mark Michelson * /, apps/app_externalivr.c: Merged revisions 99975 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99975 | mmichelson | 2008-01-23 14:25:00 -0600 (Wed, 23 Jan 2008) | 3 lines Fixing a typo. ........ 2008-01-23 17:48 +0000 [r99922-99924] Russell Bryant * /, apps/app_chanspy.c: Merged revisions 99923 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99923 | russell | 2008-01-23 11:46:55 -0600 (Wed, 23 Jan 2008) | 8 lines ChanSpy issues a beep when it starts at the beginning of a list of channels to potentially spy on. However, if there were no matching channels, it would beep at you over and over, which is pretty annoying. Now, it will only beep once in the case that there are no channels to spy on, but it will still beep again once it reaches the beginning of the channel list again. (closes issue #11738, patched by me) ........ * main/tcptls.c: Fix tcptls build when openssl isn't installed (closes issue #11813) Reported by: tzafrir Patches: asterisk-tcptls.diff.txt uploaded by jamesgolovich (license 176) 2008-01-23 17:27 +0000 [r99920] Kevin P. Fleming * channels/chan_zap.c: since echo canceler parameters in Zaptel are now signed integers, allow them during parsing 2008-01-23 15:23 +0000 [r99860] Tilghman Lesher * channels/chan_h323.c: Progress messages don't work (closes issue #10497) Reported by: pj Patches: h323-announces-r99483.diff uploaded by sergee (license 138) Tested by: pj 2008-01-23 10:18 +0000 [r99839] Olle Johansson * channels/chan_sip.c: - Add a few comments to sip_xmit - Make sure that we are aware of a pending INVITE even if we're using TCP 2008-01-23 05:29 +0000 [r99696-99818] Tilghman Lesher * apps/app_voicemail.c: Coding guidelines fixups * /, apps/app_voicemail.c: Merged revisions 99777 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99777 | tilghman | 2008-01-22 22:31:51 -0600 (Tue, 22 Jan 2008) | 8 lines When we reset the password via an external command, we should also reset the password stored in the in-memory list, too (otherwise it doesn't really take effect). (closes issue #11809) Reported by: davetroy Patches: fix_externpass.diff uploaded by davetroy (license 384) ........ * /, res/res_odbc.c: Merged revisions 99775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99775 | tilghman | 2008-01-22 22:20:15 -0600 (Tue, 22 Jan 2008) | 2 lines Oops, should have checked for a NULL obj, here, too ........ * res/res_config_ldap.c: Coding guidelines cleanup * /, main/acl.c: Merged revisions 99718 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99718 | tilghman | 2008-01-22 18:56:06 -0600 (Tue, 22 Jan 2008) | 2 lines Just confirmed that all current platforms need this header file ........ * /: Oops * /, build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, doc/ldap.txt (added), configure.ac, configs/res_ldap.conf.sample (added), res/res_config_ldap.c (added), CHANGES, makeopts.in, contrib/scripts/asterisk.ldap-schema (added), contrib/scripts/asterisk.ldif (added): Add res_config_ldap for realtime LDAP engine. (closes issue #5768) Reported by: mguesdon Patches: res_config_ldap-v0.7.tar.gz uploaded by mguesdon (license 121) res_ldap.conf.sample uploaded by suretec (license 70) asterisk-v3.1.4.ldif uploaded by suretec (license 70) asterisk-v3.1.4.schema uploaded by suretec (license 70) Tested by: oej, mguesdon, suretec, cthorner 2008-01-22 21:09 +0000 [r99647-99653] Olle Johansson * /, channels/chan_sip.c: Merged revisions 99652 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99652 | oej | 2008-01-22 21:56:09 +0100 (Tis, 22 Jan 2008) | 4 lines Thanks to Russell's education I realize that BUFSIZ has changed since I learned the C language over 20 years ago... Resetting chan_sip to the size of BUFSIZ that I expected in my old head to avoid too heavy memory allocations on some systems. ........ * doc/tex/channelvariables.tex, CHANGES: Documentation updates for BRIDGEPVTCALLID 2008-01-22 20:42 +0000 [r99646] Tilghman Lesher * /, main/acl.c: Merged revisions 99643 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99643 | tilghman | 2008-01-22 14:34:55 -0600 (Tue, 22 Jan 2008) | 2 lines Fix the defines for OS X (and Solaris, too) ........ 2008-01-22 20:41 +0000 [r99645] Russell Bryant * main/asterisk.c: Make sure the command is not just present but is also configured to be executed 2008-01-22 20:35 +0000 [r99644] Olle Johansson * main/channel.c, channels/chan_sip.c, include/asterisk/channel.h: Add a generic function to set the bridged call PVT unique id string as a channel variable BRIDGEPVTCALLID This is important for call tracing in log files and CDRs, so that the SIP callID can be traced along servers. The CHANNEL dialplan function won't work here, since the outbound channel is gone when we need the Call-ID. Other channel drivers may now implement the same function :-), but this patch only supports chan_sip.so. Inspired by (issue #11816) Reported by: ctooley Patch by oej 2008-01-22 20:33 +0000 [r99642] Russell Bryant * configs/cli.conf.sample (added), CHANGES, main/asterisk.c: Change the Asterisk CLI startup commands feature to read commands to run from cli.conf after a discussion on the -dev list. 2008-01-22 17:46 +0000 [r99595-99596] Olle Johansson * channels/chan_local.c, /, res/res_features.c, channels/chan_agent.c, apps/app_followme.c: Merged revisions 99594 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99594 | oej | 2008-01-22 18:41:57 +0100 (Tis, 22 Jan 2008) | 3 lines Add more dependencies on chan_local and add a note to the description of chan_local so that people don't disable it in menuselect just to clean up. ........ * apps/app_dial.c, /: Merged revisions 99592 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99592 | oej | 2008-01-22 18:31:17 +0100 (Tis, 22 Jan 2008) | 5 lines Add dependency on chan_local to app_dial. Dial still runs without chan_local, but will be missing forwarding functionality. ........ 2008-01-22 17:15 +0000 [r99559] Tilghman Lesher * /, main/acl.c: Merged revisions 99540 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99540 | tilghman | 2008-01-22 10:54:06 -0600 (Tue, 22 Jan 2008) | 7 lines Ensure that we can get an address even when we don't have a default route. (closes issue #9225) Reported by: junky Patches: 20080122__bug9225.diff.txt uploaded by Corydon76 (license 14) Tested by: oej, loloski, sergee ........ 2008-01-22 16:55 +0000 [r99542] Russell Bryant * channels/chan_sip.c: Point out a bug in some debug counter handling 2008-01-22 15:25 +0000 [r99464-99521] Olle Johansson * channels/chan_sip.c: Add authentication options to the SIP dialstring. Documentation follows separately (issue #11587) Reported by: sobomax Patches: chan_sip.c-trunk.diff uploaded by sobomax (license 359) * configs/sip.conf.sample: Documentation updates * doc/siptls.txt: Small fixes * main/tcptls.c, channels/chan_zap.c, main/abstract_jb.c, include/asterisk/tcptls.h: Doxygen updates 2008-01-21 23:56 +0000 [r99427] Mark Michelson * channels/chan_local.c, /: Merged revisions 99426 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99426 | mmichelson | 2008-01-21 17:55:26 -0600 (Mon, 21 Jan 2008) | 12 lines Fixing an issue wherein monitoring local channels was not possible. During a channel masquerade, the monitors on the two channels involved are swapped. In 99% of the cases this results in the desired effect. However, if monitoring a local channel, this caused the monitor which was on the local channel to get moved onto a channel which is immediately hung up after the masquerade has completed. By swapping the monitors prior to the masquerade, we avoid the problem by tricking the masquerade into placing the monitor back onto the channel where we want it. During the investigation of the issue, the channel's monitor was the only thing that was swapped in such a manner which did not make sense to have done. All other variable swapping made sense. ........ 2008-01-21 23:25 +0000 [r99424] Jason Parker * channels/chan_zap.c: Fix distinctive ring detection. Reported by: milazzo Patches: drings.diff uploaded by milazzo (license 383) Closes issue #11799 2008-01-21 22:32 +0000 [r99406] Mark Michelson * configs/queues.conf.sample, apps/app_queue.c: Adding the QUEUENAME variable to the variables set using the setqueuevar option in queues.conf. Suggestion comes from Shaun2222 on IRC. 2008-01-21 21:11 +0000 [r99382-99384] Olle Johansson * channels/chan_console.c: Remove compiler warning for uninitialized variable * channels/chan_sip.c: Doxygen updates. The TCP/TLS code was committed without any doxygen obviously. Tss tss. * channels/chan_sip.c: Updating doxygen 2008-01-21 18:15 +0000 [r99350] Tilghman Lesher * include/asterisk/res_odbc.h, /, res/res_odbc.c, configs/res_odbc.conf.sample: Merged revisions 99341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99341 | tilghman | 2008-01-21 12:11:07 -0600 (Mon, 21 Jan 2008) | 8 lines Permit the user to specify number of seconds that a connection may remain idle, which fixes a crash on reconnect with the MyODBC driver. (closes issue #11798) Reported by: Corydon76 Patches: 20080119__res_odbc__idlecheck.diff.txt uploaded by Corydon76 (license 14) Tested by: mvanbaak ........ 2008-01-21 16:02 +0000 [r99302] Joshua Colp * /, channels/chan_sip.c: Merged revisions 99301 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99301 | file | 2008-01-21 12:01:00 -0400 (Mon, 21 Jan 2008) | 4 lines Bump the buffer size for Via headers up to 512. There are some exceptionally large Via headers out there. (closes issue #11783) Reported by: ofirroval ........ 2008-01-21 07:02 +0000 [r99280] Olle Johansson * CREDITS: Update 2008-01-21 03:54 +0000 [r99265] Joshua Colp * channels/chan_sip.c: Change over to using ast_debug so these debug messages don't always show up. 2008-01-20 07:28 +0000 [r99166-99248] Russell Bryant * channels/chan_console.c: Add a "console active" CLI command, which lets you find out which console device is currently active for the Asterisk CLI, or to set it. Also, knock multiple device support off of the to-do list. * configs/console.conf.sample: correct the name of a CLI command for getting available device names * configs/console.conf.sample, channels/chan_console.c: Merge changes from team/russell/console_devices - Add support for multiple devices. All devices are configured in console.conf. - Add "console list devices" CLI command to show configured devices. Also, changed the old "list devices" to be "list available", which queries PortAudio for all audio devices that are available for use. * /, main/slinfactory.c: Merged revisions 99187 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99187 | russell | 2008-01-19 04:05:27 -0600 (Sat, 19 Jan 2008) | 4 lines Fix a couple of memory leaks with frame handling. Specifically, ast_frame_free() needed to be called on the frame that came from the translator to signed linear. ........ * README: Add Cygwin as an "other" platform that is supported * README: Various README updates 2008-01-18 Russell Bryant * Asterisk 1.6.0-beta1 released. 2008-01-18 22:04 +0000 [r99080-99085] Russell Bryant * CREDITS, include/asterisk/http.h, main/tcptls.c (added), main/manager.c, channels/chan_sip.c, doc/siptls.txt (added), main/Makefile, main/http.c, include/asterisk/tcptls.h (added), configs/sip.conf.sample, CHANGES: Merge changes from team/group/sip-tcptls This set of changes introduces TCP and TLS support for chan_sip. There are various new options in configs/sip.conf.sample that are used to enable these features. Also, there is a document, doc/siptls.txt that describes some things in more detail. This code was implemented by Brett Bryant and James Golovich. It was reviewed by Joshua Colp and myself. A number of other people participated in the testing of this code, but since it was done outside of the bug tracker, I do not have their names. If you were one of them, thanks a lot for the help! (closes issue #4903, but with completely different code that what exists there.) * main/frame.c, /, include/asterisk/translate.h: Merged revisions 99081 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99081 | russell | 2008-01-18 15:37:21 -0600 (Fri, 18 Jan 2008) | 9 lines Revert adding the packed attribute, as it really doesn't make sense why that would do any good. Fix the real bug, which is to do the check to see if the frame came from a translator at the beginning of ast_frame_free(), instead of at the end. This ensures that it always gets checked, even if none of the parts of the frame are malloc'd, and also ensures that we aren't looking at free'd memory in the case that it is a malloc'd frame. (closes issue #11792, reported by explidous, patched by me) ........ * /, include/asterisk/translate.h: Merged revisions 99079 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99079 | russell | 2008-01-18 15:22:21 -0600 (Fri, 18 Jan 2008) | 4 lines Since we're relying on the offset between the frame and the beginning of the translator pvt struct, set the packed attribute to make sure we get to the right place. (potential fix for issue #11792) ........ 2008-01-18 16:58 +0000 [r99026] Terry Wilson * res/res_features.c: This should at least temporarily fix a problem where the 't' Dial option is incorrectly passed to the transferee when built-in attended transfers are used. There is still a problem with 'T', but better to fix some problems than no problems while we work on it. (closes issue #7904) Reported by: k-egg Patches: transfer-fix-trunk-r97657.diff uploaded by sergee (license 138) Tested by: sergee, otherwiseguy 2008-01-18 06:58 +0000 [r99015-99018] Tilghman Lesher * funcs/func_odbc.c: Convert func_odbc to use SQLExecDirect for speed (closes issue #10723) Reported by: mnicholson Patches: func-odbc-direct-execute1.diff uploaded by mnicholson (license 96) Tested by: Corydon76, mnicholson, falves11 * res/res_odbc.c: Permit username and password to be NULL (which enables pass-through from the layer above). Reported by: lurcher Patch by: tilghman (Closes issue #11739) * funcs/func_cut.c: Reset default CUT delimiter back to '-' 2008-01-17 23:28 +0000 [r99006-99011] Russell Bryant * channels/chan_console.c: Make the output of "console list devices" a bit prettier. * channels/chan_console.c: List which devices are inputs and outputs in "console list devices" * main/channel.c: Add AST_FORMAT_SLINEAR16 to the list for ast_best_codec() * main/frame.c, /, channels/chan_iax2.c, include/asterisk/frame.h: Merged revisions 99004 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99004 | russell | 2008-01-17 16:37:22 -0600 (Thu, 17 Jan 2008) | 10 lines Have IAX2 optimize the codec translation path just like chan_sip does it. If the caller's codec is in our codec list, move it to the top to avoid transcoding. (closes issue #10500) Reported by: stevedavies Patches: iax-prefer-current-codec.patch uploaded by stevedavies (license 184) iax-prefer-current-codec.1.4.patch uploaded by stevedavies (license 184) Tested by: stevedavies, pj, sheldonh ........ 2008-01-17 22:22 +0000 [r99002] Mark Michelson * apps/app_voicemail.c: Fixing trunk IMAP build (closes issue #11788) Reported by: DEA Patches: vm-imap-build-fix.txt uploaded by DEA (license 3) 2008-01-17 20:51 +0000 [r98998] Jason Parker * Makefile, build_tools/cflags.xml, channels/chan_zap.c, main/dsp.c, configs/zapata.conf.sample: Add several busy detection related defines to menuselect. Allow better busy detect debugging (with BUSYDETECT_DEBUG). Remove very old BUSYDETECT and BUSY_DETECT_MARTIN defines. (closes issue #11107) Patches: busydetect_enhancement.patch uploaded by agx (license 298) busydetect-r94975.diff uploaded by sergee (license 138) Additional changes/cleanup by me. 2008-01-17 16:33 +0000 [r98993-98994] Mark Michelson * apps/app_queue.c: state_interface could be NULL, so use the never-NULL cur->state_interface for this check * apps/app_queue.c: Get the device state of the state interface instead of the interface when creating a new queue member. Thanks to Atis Lezdins for bringing this up on the Asterisk-Dev mailing list. 2008-01-17 16:21 +0000 [r98992] Jason Parker * /, configs/zapata.conf.sample: Merged revisions 98991 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes issue #11784) ........ r98991 | qwell | 2008-01-17 10:19:46 -0600 (Thu, 17 Jan 2008) | 4 lines Add a clarification about the immediate= option of zapata.conf Issue 11784, patch by klaus3000. ........ 2008-01-17 16:17 +0000 [r98989-98990] Kevin P. Fleming * channels/chan_zap.c, configs/zapata.conf.sample: major reliability and performance improvement in VWMI monitoring for FXO ports (code by markster, me and dbailey) * res/res_config_curl.c: resolve (valid) compiler warning about variable that could be used before being initialized 2008-01-17 03:09 +0000 [r98988] Terry Wilson * res/res_phoneprov.c, doc/tex/phoneprov.tex, configs/phoneprov.conf.sample: Update res_phoneprov to default to setting the SERVER variable to the IP the HTTP request for the config came in on and the SERVER_PORT to the bindport setting in sip.conf. I've left in the ability to override these options, because I can't always guess how someone might decide to do something weird with what is available to them--although needing to is pretty unlikely. Documentation was updated to reflect preference for not setting serveraddr, serveriface, or serverport. Tested on Linux and OS X. 2008-01-17 00:13 +0000 [r98987] Tilghman Lesher * cdr/cdr_adaptive_odbc.c: Change the way the new filter feature works, by allowing it to be a column NOT logged into the database. This will allow more granularity of a decision evaluated in the dialplan, then takes effect when posting the CDR. 2008-01-17 00:05 +0000 [r98986] Russell Bryant * CHANGES, main/asterisk.c: Add support for an easy way to automatically execute some Asterisk CLI commands immediately at startup. Any commands in the startup_commands file in the Asterisk config diretory will get executed. (closes issue #11781) Reported by: jamesgolovich Patches: asterisk-startupcmds.diff.txt uploaded by jamesgolovich (license 176) -- With some changes by me. 2008-01-16 23:08 +0000 [r98985] Jason Parker * configure, include/asterisk/autoconfig.h.in, configure.ac, acinclude.m4: Change AST_EXT_TOOL_CHECK to attempt to build against _LIB, per recommendations from Russell. 2008-01-16 22:36 +0000 [r98984] Tilghman Lesher * CHANGES: Info about res_config_curl 2008-01-16 22:20 +0000 [r98981] Tilghman Lesher * res/res_config_curl.c (added), main/utils.c: New module res_config_curl (closes issue #11747) Reported by: Corydon76 Patches: res_config_curl.c uploaded by Corydon76 (license 14) 20080116__bug11747.diff.txt uploaded by Corydon76 (license 14) Tested by: jmls 2008-01-16 21:53 +0000 [r98978] Russell Bryant * CREDITS, channels/chan_sip.c, configs/sip.conf.sample: Merge the changes from issue #10665 from the team/group/sip_session_timers branch. This set of changes introduces SIP session timers support (RFC 4028). In short, this prevents stuck SIP sessions that were not properly torn down due to network or endpoint failures during an established SIP session. To quote some of the documentation supplied with the patch: "The SIP Session-Timers is an extension of the SIP protocol that allows end-points and proxies to refresh a session periodically. The sessions are kept alive by sending a RE-INVITE or UPDATE request at a negotiated interval. If a session refresh fails then all the entities that support Session- Timers clear their internal session state. In addition, UAs generate a BYE request in order to clear the state in the proxies and the remote UA (this is done for the benefit of SIP entities in the path that do not support Session-Timers)." (closes issue #10665) Reported by: rjain Patches: chan_sip.c.1.diff uploaded by rjain (license 226) chan_sip.c.diff uploaded by rjain (license 226) sip.conf.sample.diff uploaded by rjain (license 226) proc_422_rsp_comment.diff uploaded by rjain (license 226) chan_sip.c.cache.diff uploaded by rjain (license 226) chan_sip.memalloc uploaded by rjain (license 226) chan_sip.memalloc.bugfix uploaded by rjain (license 226) Patches tracked in team/group/sip_session_timers, with some additional fixes by russell and oej. Tested by: jtodd, rjain, loloski 2008-01-16 19:41 +0000 [r98968-98971] Jason Parker * configure, include/asterisk/autoconfig.h.in, configure.ac: Partially revert r93898, because it broke the way netsnmp was being detected. rizzo, do you want to discuss so we can rethink this, or do you have another way? * CHANGES: Add note about new update.log to CHANGES, by request of jmls and further prodding by jsmith. * Makefile, /: Add logging for 'make update' command (also fixes updates in some places). Issue #11766, initial patch by jmls. 2008-01-16 17:51 +0000 [r98967] Joshua Colp * /, channels/chan_iax2.c: Merged revisions 98966 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98966 | file | 2008-01-16 13:50:10 -0400 (Wed, 16 Jan 2008) | 6 lines Add missing NULLs at end of two ast_load_realtimes. (closes issue #11769) Reported by: tequ Patches: chaniax.patch uploaded by dimas (license 88) ........ 2008-01-16 17:21 +0000 [r98965] Mark Michelson * channels/chan_local.c, /: Merged revisions 98964 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98964 | mmichelson | 2008-01-16 11:20:11 -0600 (Wed, 16 Jan 2008) | 10 lines Fix a deadlock in chan_local in local_hangup. There was contention because the local_pvt was held and it was attempting to lock a channel, which is the incorrect locking order. (closes issue #11730) Reported by: UDI-Doug Patches: 11730.patch uploaded by putnopvut (license 60) Tested by: UDI-Doug ........ 2008-01-16 16:06 +0000 [r98962] Terry Wilson * res/res_phoneprov.c: Make users list static 2008-01-16 15:09 +0000 [r98954-98961] Joshua Colp * main/dial.c, /: Merged revisions 98960 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98960 | file | 2008-01-16 11:08:24 -0400 (Wed, 16 Jan 2008) | 6 lines Introduce a lock into the dialing API that protects it when destroying the structure. (closes issue #11687) Reported by: callguy Patches: 11687.diff uploaded by file (license 11) ........ * /, main/rtp.c: Merged revisions 98958 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98958 | file | 2008-01-16 11:03:14 -0400 (Wed, 16 Jan 2008) | 4 lines Add two more SDP names for ulaw and alaw. (closes issue #11777) Reported by: tootai ........ * /, channels/chan_sip.c: Merged revisions 98955 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98955 | file | 2008-01-15 23:07:24 -0400 (Tue, 15 Jan 2008) | 6 lines Don't drop the old record route information when dealing with packets related to a reinvite. (closes issue #11545) Reported by: kebl0155 Patches: reinvite-patch.txt uploaded by kebl0155 (license 356) ........ * channels/chan_sip.c: Remove DNS lookup from sip_devicestate. This seems to come from way back when and I can't think of a reason for it being here, plus it could cause needless DNS lookups. (closes issue #10983) Reported by: jtodd 2008-01-16 01:35 +0000 [r98953] Steve Murphy * main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2.c: Terry found this problem with running the expr2 parser on OSX. Make the #defines come out the same between the parser & lexer. 2008-01-16 01:17 +0000 [r98952] Joshua Colp * /, build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, codecs/codec_speex.c, configure.ac, makeopts.in: Merged revisions 98951 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98951 | file | 2008-01-15 21:13:27 -0400 (Tue, 15 Jan 2008) | 4 lines Add autoconf logic for speexdsp. Later versions use a separate library for some things so we need to use it if present in codec_speex. (closes issue #11693) Reported by: yzg ........ 2008-01-15 23:53 +0000 [r98948] Russell Bryant * /, channels/chan_sip.c: Merged revisions 98946 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98946 | russell | 2008-01-15 17:50:10 -0600 (Tue, 15 Jan 2008) | 11 lines Change a buffer in check_auth() to be a thread local dynamically allocated buffer, instead of a massive buffer on the stack. This fixes a crash reported by Qwell due to running out of stack space when building with LOW_MEMORY defined. On a very related note, the usage of BUFSIZ in various places in chan_sip is arbitrary and careless. BUFSIZ is a system specific define. On my machine, it is 8192, but by definition (according to google) could be as small as 256. So, this buffer in check_auth was 16 kB. We don't even support SIP messages larger than 4 kB! Further usage of this define should be avoided, unless it is used in the proper context. ........ 2008-01-15 23:52 +0000 [r98947] Tilghman Lesher * cdr/cdr_adaptive_odbc.c, configs/cdr_adaptive_odbc.conf.sample: Add the "filter" keyword 2008-01-15 23:35 +0000 [r98944-98945] Russell Bryant * main/translate.c, include/asterisk/translate.h: Clean up something I did for ABI compatability in 1.4 * main/frame.c, /, main/translate.c, main/abstract_jb.c, channels/chan_iax2.c, codecs/codec_zap.c, include/asterisk/frame.h, main/rtp.c, include/asterisk/translate.h: Merged revisions 98943 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98943 | russell | 2008-01-15 17:26:52 -0600 (Tue, 15 Jan 2008) | 25 lines Commit a fix for some memory access errors pointed out by the valgrind2.txt output on issue #11698. The issue here is that it is possible for an instance of a translator to get destroyed while the frame allocated as a part of the translator is still being processed. Specifically, this is possible anywhere between a call to ast_read() and ast_frame_free(), which is _a lot_ of places in the code. The reason this happens is that the channel might get masqueraded during this time. During a masquerade, existing translation paths get destroyed. So, this patch fixes the issue in an API and ABI compatible way. (This one is for you, paravoid!) It changes an int in ast_frame to be used as flag bits. The 1 bit is still used to indicate that the frame contains timing information. Also, a second flag has been added to indicate that the frame came from a translator. When a frame with this flag gets released and has this flag, a function is called in translate.c to let it know that this frame is doing being processed. At this point, the flag gets cleared. Also, if the translator was requested to be destroyed while its internal frame still had this flag set, its destruction has been deffered until it finds out that the frame is no longer being processed. Admittedly, this feels like a hack. But, it does fix the issue, and I was not able to think of a better solution ... ........ 2008-01-15 20:10 +0000 [r98895-98935] Joshua Colp * /, channels/chan_sip.c: Merged revisions 98934 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98934 | file | 2008-01-15 16:08:43 -0400 (Tue, 15 Jan 2008) | 4 lines Based on the boundary found move over the correct amount. (closes issue #11750) Reported by: tasker ........ * /, channels/chan_sip.c: Merged revisions 98894 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98894 | file | 2008-01-14 18:41:55 -0400 (Mon, 14 Jan 2008) | 4 lines Accept "; boundary=" not just ";boundary=" in the multipart mixed content type. (closes issue #11750) Reported by: tasker ........ 2008-01-14 22:19 +0000 [r98889] Jason Parker * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: Add backupdeleted option to app_voicemail (closes issue #10740) Reported by: ruffle Patches: app_voicemail.diff uploaded by ruffle (license 201) 10740-voicemail.diff uploaded by qwell (license 4) 20080113_bug10740.diff.txt uploaded by mvanbaak (license 7) Tested by: blitzrage, mvanbaak, qwell 2008-01-14 22:11 +0000 [r98850-98888] Mark Michelson * apps/app_directory.c: Big improvement for app_directory. This patch breaks the do_directory function up so that it is more easily parsed by the human brain. It also fixes some errors. I'll quote dimas from the original bug description: "app_directory contained some duplicate code even before addition of 'm' option. Addition of that option doubled amount of that code. Worst of all, there are minor differences between these code block and bugs caused by these differences. 1. There is a memory leak. In the 'menu' mode, result of the convert(pos) function is not freed while it should be. 2. In the 'menu' mode check for OPT_LISTBYFIRSTNAME flag ('f' option) is not negated as result, application works in the mode opposite to what user expect (checking last name when user wants the first nd vice versa). 3. select_item function plays message for user using res = func1() || func2() || func3()... construct. This construct loses the actual value returned by ast_waitstream() for example so at the end, res does not contain digit user dialed while listening to the message. 4. (also in 1.4) application announces entries from voicemail.conf/realtime separately from entries from users.conf. I see no reason why doing so instead of building combined list. 5. Alot of duplicated code as already mentioned." This was tested by dimas and I (I tested under valgrind). A word of caution: any bug fixes that happen in app_directory in 1.4 will almost certainly not merge cleanly into trunk as a result of this, but it is well worth it. Huge thanks to dimas for this wonderful submission. (closes issue #11744) Reported by: dimas Patches: dir3.patch uploaded by dimas (license 88) Tested by: putnopvut, dimas 2008-01-14 20:01 +0000 [r98830] Joshua Colp * main/manager.c: Make sure the user's manager secret exists, even if it is blank. (closes issue #11749) Reported by: srt 2008-01-14 18:42 +0000 [r98811] Terry Wilson * CHANGES: Add description of TOUPPER and TOLOWER dialplan functions to CHANGES. 2008-01-14 17:40 +0000 [r98776] Jason Parker * channels/chan_skinny.c: Add proper call forwarding (all and busy) support for chan_skinny. Note: NoAnswer support is currently not implemented, as it would take a significant amount of work to figure out how to do correctly. Closes issue #11310, patches, testing, and support by DEA, mvanbaak, and myself. 2008-01-14 17:39 +0000 [r98775] Russell Bryant * /, main/translate.c: Merged revisions 98774 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98774 | russell | 2008-01-14 11:38:38 -0600 (Mon, 14 Jan 2008) | 3 lines Revert a change that introduces an unacceptable performance hit and is causing memory leaks ... (from rev 97973) ........ 2008-01-14 17:18 +0000 [r98773] Jason Parker * channels/chan_skinny.c: Fix for potential crash with vmexten 2008-01-14 16:36 +0000 [r98735-98738] Mark Michelson * apps/app_queue.c: Merged revisions 98737 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98737 | mmichelson | 2008-01-14 10:35:12 -0600 (Mon, 14 Jan 2008) | 3 lines Fixing another compilation error. I'm a bit off today :( ........ * /, apps/app_queue.c: Merged revisions 98733 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98733 | mmichelson | 2008-01-14 10:21:28 -0600 (Mon, 14 Jan 2008) | 8 lines Adding explicit defaults for missing options to init_queue. This is necessary because if a user either removes or comments one of these options and reloads their queues, the option will not reset to its default, instead maintaining the value from prior to the reload. Thanks to John Bigelow for pointing this error out to me. ........ 2008-01-14 15:07 +0000 [r98695-98714] Joshua Colp * main/pbx.c: Print out a warning when spaces are used in the variable name in Set and MSet. It is extremely hard to debug this issue so this should make it easier. (closes issue #11759) Reported by: caio1982 Patches: setvar_space_warning1.diff uploaded by caio1982 (license 22) * apps/app_meetme.c, doc/tex/qos.tex, doc/tex/realtime.tex: Update documentation. (closes issue #11763) Reported by: IgorG Patches: docupd.v1.diff uploaded by IgorG (license 20) 2008-01-14 04:53 +0000 [r98558-98676] Russell Bryant * apps/app_jack.c: Add another small option for the JACK app and JACK_HOOK function. The 'n' option tells JACK not to start jackd automatically if it is not already running. Otherwise, the default is that jackd will get started for you if it isn't running already. * CHANGES: - Break up the Misc. section a bit with a new section for Misc. New Modules - Change spacing a bit in some places for consistent indentation * CHANGES, apps/app_jack.c (added): Bring in the code from team/russell/jack/. Add a new module, app_jack, which provides interfaces to JACK, the Jack Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are provided; there is a JACK() application, and a JACK_HOOK() function. Both interfaces create an input and output JACK port. The application makes these ports the endpoint of the call. The audio coming from the channel goes out the output port and whatever comes back in on the input port is what gets sent to the channel. The JACK_HOOK() function turns on a JACK audiohook on the channel. This lets you run the audio coming from a channel through JACK, and whatever comes back in is what gets forwarded on as the channel's audio. This is very useful for building custom vocoders or doing recording or analysis of the channel's audio in another application. In case anyone is curious, the platform that inspired me to write this is PureData (http://puredata.info/). I wrote these JACK interfaces so that I could use Pd to do interesting things with the audio of phone calls ... * build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add configure script check for JACK. * build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Remove KDE configure script check that isn't used * main/audiohook.c: Remove a duplicate lock of the audiohook lock when destroying manipulate audiohooks. This causes an error when we attempt to destroy the lock later when freeing the audiohook. * main/pbx.c, CHANGES: Add a new CLI command, "core set chanvar", which allows you to set a channel variable (or function) on an active channel from the CLI. 2008-01-12 18:12 +0000 [r98536] Tilghman Lesher * main/manager.c: Conversion to load manager.conf into memory did not convert the password functions correctly. (Closes issue #11749) 2008-01-12 05:13 +0000 [r98514] Pari Nannapaneni * /, main/http.c: merging a comment added in 1.4 2008-01-12 00:20 +0000 [r98488] Kevin P. Fleming * channels/chan_zap.c, CHANGES: Add 'zap set dnd' CLI command, and ensure that the AMI DNDState event always gets generated. (closes issue #11212) Reported by: tzafrir Patches: zap_dnd.diff uploaded by tzafrir (modified by me) (license 46) 2008-01-12 00:17 +0000 [r98487] Tilghman Lesher * /, res/res_odbc.c: Merged revisions 98467 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98467 | tilghman | 2008-01-11 18:05:08 -0600 (Fri, 11 Jan 2008) | 4 lines Add a connection timeout attribute, as that was what was intended with the login timeout, but ODBC divides it up into 2 different timeouts. (Closes issue #11745) ........ 2008-01-11 23:57 +0000 [r98454] Russell Bryant * configure, doc/tex/Makefile, configure.ac, makeopts.in: Add some extra checking to help out with a potential error when trying to run "make asterisk.pdf" when not all of the right packages are installed. (closes issue #10763) Reported by: Corydon76 Patches: 20070919__bug10763.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 2008-01-11 23:10 +0000 [r98436] Kevin P. Fleming * channels/chan_zap.c, CHANGES, configs/zapata.conf.sample: Add 'auto' signalling mode for Zaptel channels. (closes issue #11690) Reported by: tzafrir Patches: signaling_to_signalling.diff uploaded by tzafrir (license 46) signalling_cleanup.diff uploaded by tzafrir (license 46) zap_auto_default.diff uploaded by tzafrir (license 46) zap_no_default_sig.diff uploaded by tzafrir (license 46) zap_signal_auto.diff uploaded by tzafrir (license 46) 2008-01-11 23:09 +0000 [r98424-98435] Joshua Colp * main/event.c: Goodbye again drumkilla. * main/event.c: drumkilla ftw. * main/audiohook.c: I am no longer Rockin' * main/audiohook.c: Testing something... 2008-01-11 22:52 +0000 [r98400] Russell Bryant * /, pbx/pbx_dundi.c: Merged revisions 98390 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98390 | russell | 2008-01-11 16:46:21 -0600 (Fri, 11 Jan 2008) | 9 lines Fix up setting the EID on BSD based systems. (closes issue #11646) Reported by: caio1982 Patches: dundi_osx_eid6.diff.txt uploaded by caio1982 (license 22) dundi_osx_eid6-1.4.diff uploaded by caio1982 (license 22) Tested by: caio1982, mvanbaak ........ 2008-01-11 19:53 +0000 [r98318-98334] Joshua Colp * /, main/rtp.c: Merged revisions 98325 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98325 | file | 2008-01-11 15:51:10 -0400 (Fri, 11 Jan 2008) | 6 lines If the incoming RTP stream changes codec force the bridge to break if the other side does not support it. (closes issue #11729) Reported by: tsearle Patches: new_codec_patch_udiff.patch uploaded by tsearle (license 373) ........ * /, res/res_agi.c: Merged revisions 98317 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98317 | file | 2008-01-11 15:28:30 -0400 (Fri, 11 Jan 2008) | 6 lines If the channel is hungup during RECORD FILE send a result code of -1 to be uniform with everything else. (closes issue #11743) Reported by: davevg Patches: res_agi.diff uploaded by davevg (license 209) ........ 2008-01-11 19:12 +0000 [r98316] Mark Michelson * main/channel.c, /: Merged revisions 98315 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98315 | mmichelson | 2008-01-11 13:10:57 -0600 (Fri, 11 Jan 2008) | 5 lines Properly report the hangup cause as no answer when someone does not answer (closes issue #10574, reported by boch, patched by moy) ........ 2008-01-11 19:05 +0000 [r98270-98308] Russell Bryant * codecs/codec_resample.c: Kevin noted that the thing that I _actually_ changed here was that I converted a value from a double, to a float, back to a double. Sure enough, when I changed my interim variable back to a double, it still blows up. Switching all of these to a float fixes the problem. This seems like a compiler bug where a double passed as an argument isn't getting properly aligned, so I'll have to see if I can replicate it with a small test program. (related to issue #11725) * codecs/codec_resample.c: Fix a bus error that happened when asterisk was built with optimizations on with platforms that explode on unaligned access. I'm not exactly sure why this fixes it, but it fixed it on the machine I was testing on. If it makes sense to you, feel free to enlighten me. :) (closes issue #11725, patched by me) 2008-01-11 18:35 +0000 [r98268-98269] Tilghman Lesher * cdr/cdr_adaptive_odbc.c: Port Nick Gorham's timestamp patch to adaptive_odbc, too * cdr/cdr_odbc.c: Commit Nick Gorham's suggestion for timestamp fix 2008-01-11 17:27 +0000 [r98220] Joshua Colp * /, apps/app_followme.c: Merged revisions 98219 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98219 | file | 2008-01-11 13:22:53 -0400 (Fri, 11 Jan 2008) | 4 lines Ensure the return value of ast_bridge_call is passed back up as the application return value. This is needed for transfers to function so the PBX core knows to continue execution. (closes issue #10327) Reported by: kkiely ........ 2008-01-11 17:17 +0000 [r98218] Russell Bryant * codecs/codec_g722.c: At one point during working on this module, I had the lin/lin16 versions of the framein callbacks different. However, they are now the same again, so remove the duplicate code and use the same functions for the lin/lin16 versions. 2008-01-11 16:08 +0000 [r98152-98193] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 98164 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98164 | tilghman | 2008-01-11 09:52:31 -0600 (Fri, 11 Jan 2008) | 2 lines Back out changes from revision 97077, since it wasn't perfect ........ * doc/manager_1_1.txt: Documentation updates 2008-01-11 12:51 +0000 [r98124] Kevin P. Fleming * channels/chan_sip.c: Ascom phones send Flash events as SIP INFO using '!' as the 'digit' 2008-01-11 03:40 +0000 [r98081-98083] Russell Bryant * codecs/codec_g722.c, main/frame.c: - Fix the last set of places where incorrect assumptions were made about the sample length with g722. It is _2_ samples per byte, not 1. This was all over the place, and I believed it, and it is what caused me to take so long to figure out what was broken. - Update copyright information on codec_g722. 2008-01-11 00:54 +0000 [r98047] Mark Michelson * main/translate.c: Fix "core show translation" to not output information for "unknown" codecs. This fix was made in favor of the proposed patch since it doesn't involve changing a core codec define. (closes issue #11722, reported and initially patched by caio1982, final patch by me) 2008-01-11 00:38 +0000 [r98024-98027] Russell Bryant * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add a new global and per-peer option to chan_sip, qualifyfreq, which allows you to set the qualify frequency. (closes issue #11597) Reported by: wilder Patches: qualifyfreq5.patch uploaded by wilder (license 362) -- with some mods by me * main/translate.c: Simplify this code with a suggestion from Luigi on the asterisk-dev list. Instead of using is16kHz(), implement a format_rate() function. 2008-01-10 23:40 +0000 [r97978] Tilghman Lesher * /, channels/chan_sip.c, main/translate.c: Merged revisions 97973 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97973 | tilghman | 2008-01-10 17:08:36 -0600 (Thu, 10 Jan 2008) | 6 lines 1) When we get a translated frame out, clone it, because if the translator pvt is freed before we use the frame, bad things happen. 2) Getting a failure from ast_sched_delete means that the schedule ID is currently running. Don't just ignore it. (Closes issue #11698) ........ 2008-01-10 23:33 +0000 [r97974-97977] Russell Bryant * /, main/translate.c: Merged revisions 97976 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97976 | russell | 2008-01-10 17:30:40 -0600 (Thu, 10 Jan 2008) | 3 lines Fix various timing calculations that made assumptions that the audio being processed was at a sample rate of 8 kHz. ........ * codecs/codec_g722.c: Fix various issues in codec_g722. - The most common fix being made here is to fix all of the places where the number of output samples and output bytes gets updated in the translator state structure. - Fix a number of other places where the number of samples provided as an initialization value to a struct was incorrect. * codecs/codec_resample.c: Fix the buffer_samples value. For signed linear, the number of samples needed to fill the buffer is half the buffer size. 2008-01-10 21:58 +0000 [r97933] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 97925 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97925 | mmichelson | 2008-01-10 15:57:06 -0600 (Thu, 10 Jan 2008) | 6 lines Let us leave a voicemail for ourself if we have logged into VoiceMailMain and chosen to leave a message. (closes issue #11735, reported and patched by jamessan) ........ 2008-01-10 21:46 +0000 [r97850-97890] Steve Murphy * /, res/ael/ael_lex.c, res/Makefile, res/ael/ael.flex: Merged revisions 97889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97889 | murf | 2008-01-10 14:37:10 -0700 (Thu, 10 Jan 2008) | 1 line Applied the same fixes for ael.flex as was done in 97849 for ast_expr2.fl; overrode the normally generate yyfree func with our own version that checks the pointer for non-null before passing to free(). Also takes care of a little problem with 2.5.33 and the use of the __STDC_VERSION__ macro. ........ * /, main/Makefile, main/ast_expr2f.c, main/ast_expr2.fl: Merged revisions 97849 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97849 | murf | 2008-01-10 13:21:27 -0700 (Thu, 10 Jan 2008) | 1 line This is a fix for 2 things: a problem Terry was having in OSX with null pointers, which was my fault, as I probably forgot to run the sed script last time I made mods. So, I moved the fix into the flex input itself. Then, I found when I used flex 2.5.33, that it was using __STDC_VERSION__, and that's not real good; so I added back in a DIFFERENT sed script to fix that little mess. Tested everything, a couple different ways. Hope I did no harm, at the least. ........ 2008-01-10 20:13 +0000 [r97848] Jason Parker * /, include/asterisk/frame.h: Merged revisions 97847 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97847 | qwell | 2008-01-10 14:12:37 -0600 (Thu, 10 Jan 2008) | 1 line Fix a comment that is no longer true. ........ 2008-01-10 20:05 +0000 [r97846] Mark Michelson * apps/app_voicemail.c: Use the appropriate line ending for the X-Asterisk-VM-Message-Type header. (closes issue #11734, reported and patched by jaroth) 2008-01-10 19:07 +0000 [r97825-97826] Terry Wilson * main/ast_expr2f.c: heh, remove patch to generated file. * main/ast_expr2f.c, main/cli.c: Check pointers before freeing (was getting WARNINGS under MALLOC_DEBUG) 2008-01-10 17:38 +0000 [r97805] Tilghman Lesher * cdr/cdr_odbc.c: Fix problem with timestr going out of scope (Closes issue #11726, closes issue #11731) 2008-01-10 17:30 +0000 [r97745-97804] Russell Bryant * formats/format_sln16.c: minor formatting changes * main/translate.c: spaces to tabs * configure, configure.ac: Use AST_EXT_TOOL_CHECK() for the GTK check again. I changed this to an inline implementation to fix a small bug, but after a discussion with rizzo, I went to change it back. Also, it turns out that the implementation of the macro already supported what was needed to fix the problem. * pbx/pbx_kdeconsole.h (removed), /, configs/modules.conf.sample, pbx/kdeconsole_main.cc (removed): Merged revisions 97753 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97753 | russell | 2008-01-10 10:19:47 -0600 (Thu, 10 Jan 2008) | 2 lines Remove other remnants of pbx_kdeconsole ........ * /, build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, configure.ac, makeopts.in, pbx/pbx_kdeconsole.cc (removed): Merged revisions 97734 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97734 | russell | 2008-01-10 10:10:09 -0600 (Thu, 10 Jan 2008) | 4 lines Remove pbx_kdeconsole from the tree. It hasn't worked in ages, and nobody has complained. (closes issue #11706, reported by caio1982) ........ 2008-01-10 15:12 +0000 [r97698] Joshua Colp * funcs/func_groupcount.c, /: Merged revisions 97697 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97697 | file | 2008-01-10 11:07:12 -0400 (Thu, 10 Jan 2008) | 6 lines Don't try to copy the category from the group if no category exists. (closes issue #11724) Reported by: IgorG Patches: group_count.v1.patch uploaded by IgorG (license 20) ........ 2008-01-10 00:54 +0000 [r97657] Russell Bryant * include/asterisk.h: These prototypes are not supposed to be in asterisk.h. They are already in version.h. 2008-01-10 00:50 +0000 [r97656] Steve Murphy * include/asterisk.h, channels/console_video.c, utils/astman.c, channels/console_board.c, channels/vgrabbers.c: The fixes in this commit are mainly to allow compiling of trunk with --enable-dev-mode, mutex profiling, lock debugging, etc. Mainly, the version.c needs to be in the OBJS line; asterisk.h was chosen to have the prototypes for ast_get_version, ast_get_version_num; and the ASTERISK_FILE_VERSION macro needs to be used after including asterisk.h in a few files. I hope I did the right thing. If not, let me know. 2008-01-10 00:39 +0000 [r97655] Tilghman Lesher * main/manager.c: oops, missed the case of a 0 permission (which should mean everybody is allowed, not nobody) 2008-01-10 00:22 +0000 [r97653] Terry Wilson * res/res_phoneprov.c: Attempt at making lookup_iface work under FreeBSD. Not yet tested, but it compiles under OS X. And still works under linux. 2008-01-10 00:17 +0000 [r97652] Russell Bryant * codecs/Makefile: Fix this so it doesn't force codec_g722 to get relinked every time 2008-01-10 00:12 +0000 [r97651] Tilghman Lesher * main/pbx.c, main/manager.c, channels/chan_sip.c, res/res_features.c, pbx/pbx_realtime.c, configs/manager.conf.sample, CHANGES, channels/chan_iax2.c, include/asterisk/manager.h, apps/app_stack.c, main/db.c, apps/app_voicemail.c: Several manager changes: 1) Add the Dialplan class, for NewExten and VarSet events, which should cut down on the volume of traffic in the Call class. 2) Permit some commands to be run from multiple classes, such as allowing DBGet to be run from either the System or the Reporting class. 3) Heavily document each class in the sample config, as there were several that made no sense to be in the write= line, and two that made no sense to be in the read= line (since they controlled no permissions there). (Closes issue #10386) 2008-01-10 00:11 +0000 [r97641-97650] Russell Bryant * codecs/Makefile: Ensure that libg722.a gets rebuilt if one of the files changes * /, pbx/pbx_gtkconsole.c: Merged revisions 97645 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97645 | russell | 2008-01-09 17:01:48 -0600 (Wed, 09 Jan 2008) | 2 lines Strip terminal sequences from the verbose messages ........ * configure: re-gen configure * configure.ac: re-add check for gtk1, which is used for pbx_gtkconsole (related to issue #11706) * /, pbx/pbx_gtkconsole.c: Merged revisions 97640 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97640 | russell | 2008-01-09 16:26:33 -0600 (Wed, 09 Jan 2008) | 3 lines Make pbx_gtkconsole build ... but doesn't actually load on my system still (related to issue #11706) ........ 2008-01-09 21:37 +0000 [r97634] Terry Wilson * phoneprov/000000000000.cfg, phoneprov/000000000000-directory.xml, phoneprov/polycom.xml, res/res_phoneprov.c (added), funcs/func_strings.c, phoneprov/000000000000-phone.cfg, configs/modules.conf.sample, main/acl.c, include/asterisk/localtime.h, CHANGES, configs/phoneprov.conf.sample (added), Makefile, phoneprov (added), doc/tex/phoneprov.tex (added), main/stdtime/localtime.c, doc/tex/asterisk.tex: Added a new module, res_phoneprov, which allows auto-provisioning of phones based on configuration templates that use Asterisk dialplan function and variable substitution. It should be possible to create phone profiles and templates that work for the majority of phones provisioned over http. It is currently only intended to provision a single user account per phone. An example profile and set of templates for Polycom phones is provided. NOTE: Polycom firmware is not included, but should be placed in AST_DATA_DIR/phoneprov/configs to match up with the included templates. 2008-01-09 20:30 +0000 [r97620-97623] Jason Parker * /, main/cli.c: Merged revisions 97622 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11718) ........ r97622 | qwell | 2008-01-09 14:28:43 -0600 (Wed, 09 Jan 2008) | 5 lines Correctly display a message if a command could not be found. Also fix a comment which may have led to this happening. Issue 11718, reported by kshumard. ........ * /, main/cli.c: Merged revisions 97618 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97618 | qwell | 2008-01-09 14:05:45 -0600 (Wed, 09 Jan 2008) | 1 line Fix some locking and return value funkiness. We really shouldn't be unlocking this lock inside of a function, unless we locked it there too. ........ 2008-01-09 18:53 +0000 [r97577] Mark Michelson * /, apps/app_queue.c: Merged revisions 97575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97575 | mmichelson | 2008-01-09 12:48:15 -0600 (Wed, 09 Jan 2008) | 3 lines Part 2 of app_queue doxygen improvements. Some smaller functions this time ........ 2008-01-09 18:12 +0000 [r97532-97533] Luigi Rizzo * channels/console_gui.c: remove a wrong 'const' * images/kpad2.jpg: add annotations for the two message windows we use. 2008-01-09 18:04 +0000 [r97531] Russell Bryant * /, res/res_features.c: Merged revisions 97529 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97529 | russell | 2008-01-09 12:02:08 -0600 (Wed, 09 Jan 2008) | 2 lines Fix saying the parking space number to the caller doing the parking ... ........ 2008-01-09 18:03 +0000 [r97530] Luigi Rizzo * channels/console_gui.c, channels/console_board.c, channels/console_video.h: Two changes: - support scrolling of message window; - simplify the code for creating a message window, and try it using a second one in the top of the keypad (where we echo the dialed number). The 'skin' that supports these two windows will be committed separately. 2008-01-09 17:30 +0000 [r97495] Kevin P. Fleming * /, codecs/codec_zap.c: Merged revisions 97491 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97491 | kpfleming | 2008-01-09 11:21:14 -0600 (Wed, 09 Jan 2008) | 2 lines report the same message whether Zaptel does not have transcoder support loaded or no transcoders were found ........ 2008-01-09 16:59 +0000 [r97490] Philippe Sultan * /, channels/chan_gtalk.c: Merged revisions 97489 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97489 | phsultan | 2008-01-09 17:44:24 +0100 (Wed, 09 Jan 2008) | 7 lines Set the caller id within the gtalk_alloc function. As underlined in issue #10437 by Josh, we need to prevent a possible memory leak. We only set the name part of the caller id, the number part is not relevant when dealing with JIDs. Closes issue #11549. ........ 2008-01-09 16:44 +0000 [r97488] Luigi Rizzo * channels/console_gui.c, channels/console_video.c, channels/console_board.c, channels/console_video.h: Implement keyboard handling, and use it to enter a number to dial in the 'message' area under the keypad. Now you can make calls using the keypad as a regular phone (or the keyboard for chars not present on the keypad) 2008-01-09 16:13 +0000 [r97451] Joshua Colp * /, apps/app_meetme.c: Merged revisions 97450 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97450 | file | 2008-01-09 12:11:17 -0400 (Wed, 09 Jan 2008) | 6 lines Don't do conferencing totally in Zaptel if Monitor is running on the channel. (closes issue #11709) Reported by: BigJimmy Patches: patch-meetmerec uploaded by BigJimmy (license 371) ........ 2008-01-09 15:45 +0000 [r97421-97449] Kevin P. Fleming * channels/chan_zap.c, /: Merged revisions 97448 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97448 | kpfleming | 2008-01-09 09:43:19 -0600 (Wed, 09 Jan 2008) | 2 lines pass the right variable to get an error string... oops ........ * channels/chan_zap.c, /: Merged revisions 97410 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97410 | kpfleming | 2008-01-09 09:26:23 -0600 (Wed, 09 Jan 2008) | 2 lines add error number output to ioctl failure messages to help with debugging ........ 2008-01-09 12:23 +0000 [r97389-97390] Luigi Rizzo * channels/console_video.c, channels/console_video.h: implement the "console startgui" and "console stopgui" commands so you can start and stop the gui even outside of a call. This is convenient for testing, and also for using the keypad to pick up a call, and to dial a number (the latter not yet implemented, but should be close). * channels/chan_oss.c: make get_video_desc() return the active console if passed a null argument (channel). 2008-01-09 00:58 +0000 [r97364-97365] Tilghman Lesher * main/asterisk.c: New option in trunk, needs strdupa to be safe, too * /, main/editline/readline.c, main/cli.c: Merged revisions 97350 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97350 | tilghman | 2008-01-08 18:44:14 -0600 (Tue, 08 Jan 2008) | 5 lines Allow filename completion on zero-length modules, remove a memory leak, remove a file descriptor leak, and make filename completion thread-safe. Patched and tested by tilghman. (Closes issue #11681) ........ 2008-01-09 00:18 +0000 [r97307-97309] Mark Michelson * /, apps/app_queue.c: Merged revisions 97308 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97308 | mmichelson | 2008-01-08 18:17:40 -0600 (Tue, 08 Jan 2008) | 3 lines use the \retval doxygen command properly ........ * /, apps/app_queue.c: Merged revisions 97304 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97304 | mmichelson | 2008-01-08 17:49:11 -0600 (Tue, 08 Jan 2008) | 5 lines Part 1 of N of adding doxygen comments to app_queue. I picked some of the most common functions used (which also happen to be some the biggest/ugliest functions too) to document first. I'm pretty new to doxygen so criticism is welcome. ........ 2008-01-08 23:51 +0000 [r97305] Tilghman Lesher * apps/app_voicemail.c: Add a new flag 'd' (with optional context) permitting any extension within that context to be entered as a new extension during the playback of a voicemail greeting. Patch inspired by bluecrow76, by tilghman. (Closes issue #7063) 2008-01-08 23:35 +0000 [r97280-97303] Luigi Rizzo * channels/console_board.c: add copyright (most of this code was written by Marta Carbone), remove some unused code, add/clarify some comments. * images/kpad2.jpg: Add the annotation for the textarea used for messages, and also change the background from white to something different to show that we can make use of fonts with transparent background. * images/font.png (added): add a font suitable for use with the console GUI. The background of this particular image is transparent so we can preserve the original background when we draw strings. * channels/console_gui.c, channels/console_video.c, channels/console_board.c (added), channels/Makefile: add support for textareas, used for various dialog windows on the gui. The main code to implement the textarea is in console_board.c, and uses a simple png image with the font, blitting characters on the designated areas of the main screen. Additionally we provide some annotations in the image used as a skin to indicate which areas are used for text messages. (images will be committed separately). At the moment the dialog area is only used to display a running counter, just as a proof of concept. 2008-01-08 21:56 +0000 [r97248] Terry Wilson * apps/app_queue.c: Initialize new variable to NULL 2008-01-08 21:28 +0000 [r97203-97208] Mark Michelson * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Adding the option of specifying a second interface in a member definition for a queue. app_queue will monitor this second device's state for the member, even though it actually calls the first interface. This ability has been added for statically defined queue members, realtime queue members, and dynamic queue members added through the CLI, dialplan, or manager. (closes issue #11603, reported by acidv) 2008-01-08 21:01 +0000 [r97199-97200] Olle Johansson * channels/chan_console.c: Change reference to external library so it appears on the extref listing http://www.asterisk.org/doxygen/trunk/extref.html * res/res_jabber.c: Iksemel is alive in a new home. Release 1.3 is out with bug fixes. 2008-01-08 20:56 +0000 [r97198] Tilghman Lesher * main/autoservice.c, /, main/utils.c: Merged revisions 97194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97194 | tilghman | 2008-01-08 14:47:07 -0600 (Tue, 08 Jan 2008) | 3 lines Increase constants to where we're less likely to hit them while debugging. (Closes issue #11694) ........ 2008-01-08 20:52 +0000 [r97196-97197] Joshua Colp * channels/chan_sip.c: One line documentation ftw! * /, channels/chan_mgcp.c: Merged revisions 97195 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97195 | file | 2008-01-08 16:48:20 -0400 (Tue, 08 Jan 2008) | 6 lines Fix various DTMF issues in chan_mgcp. (closes issue #11443) Reported by: eferro Patches: dtmf_control_hybrid-inband-mode.patch uploaded by eferro (license 337) ........ 2008-01-08 20:45 +0000 [r97193] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 97192 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97192 | mmichelson | 2008-01-08 14:42:07 -0600 (Tue, 08 Jan 2008) | 9 lines Making some changes designed to not allow for a corrupted mailstream for a vm_state. 1. Add locking to the vm_state retrieval functions so that no linked list corruption occurs. 2. Make sure to always grab the persistent vm_state when mailstream access is necessary. 3. Correct an incorrect return value in the init_mailstream function. (closes issue #11304, reported by dwhite) ........ 2008-01-08 20:06 +0000 [r97153-97154] Joshua Colp * channels/chan_sip.c: Move common code for setting T38 capabilities and fix a bug with fax detection in the SIP RTP read callback. It's still sort of silly... but more on that later. (closes issue #11239) Reported by: dimas Patches: sipt38prop.patch uploaded by dimas (license 88) * funcs/func_groupcount.c, /: Merged revisions 97152 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97152 | file | 2008-01-08 15:53:52 -0400 (Tue, 08 Jan 2008) | 4 lines If no group has been provided to the GROUP_COUNT dialplan function then use the first one specific to the channel. (closes issue #11077) Reported by: m4him ........ 2008-01-08 19:06 +0000 [r97125] Tilghman Lesher * /, channels/chan_sip.c, main/asterisk.c: Merged revisions 97077 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97077 | tilghman | 2008-01-08 12:02:13 -0600 (Tue, 08 Jan 2008) | 3 lines Apply multiple crash fixes, found in issue #11386, but not completely closing that issue. ........ 2008-01-08 18:42 +0000 [r97041-97103] Joshua Colp * /, apps/app_queue.c: Merged revisions 97093 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97093 | file | 2008-01-08 14:36:40 -0400 (Tue, 08 Jan 2008) | 4 lines Make app_queue calls work with directed pickup. (closes issue #11700) Reported by: jbauer ........ * utils/extconf.c: Make ast_atomic_fetchadd_int_slow magically appear in extconf. (closes issue #11703) Reported by: dmartin 2008-01-07 23:03 +0000 [r96988] Luigi Rizzo * channels/console_gui.c: add support for cropping the keypad image while displaying it. This way it can contain additional elements (e.g. fonts, buttons, widgets) without having to use a zillion files to store them. 2008-01-07 22:31 +0000 [r96987] Mark Michelson * apps/app_voicemail.c: Explicitly make literal constants long where they are expected to be. 2008-01-07 21:12 +0000 [r96936] Jason Parker * main/config.c: Display a message if no config mappings are found with "core show config mappings". Closes issue #11704, patch by kshumard. 2008-01-07 21:10 +0000 [r96934-96935] Mark Michelson * apps/app_voicemail.c: Document some weird casting magic that's necessary to interface with the c-client * doc/tex/imapstorage.tex, apps/app_voicemail.c: Adding user-configurable TCP timeout settings to IMAP voicemail. This could go a long way towards preventing unexplainable hangs experienced by people. In the case of MWI hangs, this also will mean that the SIP port isn't blocked anymore. (closes issue #11665, reported by yehavi) 2008-01-07 20:48 +0000 [r96885-96933] Russell Bryant * /, configs/extensions.conf.sample: Merged revisions 96932 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r96932 | russell | 2008-01-07 14:47:52 -0600 (Mon, 07 Jan 2008) | 10 lines Merged revisions 96931 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r96931 | russell | 2008-01-07 14:46:22 -0600 (Mon, 07 Jan 2008) | 2 lines Change misery.digium.com to pbx.digium.com ........ ................ * configs/http.conf.sample: Add a note about viewing the default set of documentation using the built-in http server * Makefile: If the HTML documentation exists, install it in the static-http/docs directory so that it can be viewed through the Asterisk http server if it is turned on. * build_tools/prep_tarball: Build the HTML version of the doc files for tarballs, as well * res/res_smdi.c, /: Merged revisions 96884 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r96884 | russell | 2008-01-07 10:39:23 -0600 (Mon, 07 Jan 2008) | 3 lines Don't crash if something happens when setting up an SMDI interface and it gets destroyed before the SMDI port handling thread gets created. ........ 2008-01-07 16:17 +0000 [r96862] Kevin P. Fleming * formats/format_sln16.c (added): add a file-format driver for 16KHz signed linear... which may or may not work 2008-01-07 15:52 +0000 [r96858] Joshua Colp * main/manager.c, main/loader.c: Move ModuleLoad and ModuleCheck manager commands from loader.c to manager.c. Previously they would get registered twice because of the way manager.c operates. (closes issue #11699) Reported by: caio1982 Patches: manager_module_commands1.diff uploaded by caio1982 (license 22) 2008-01-07 15:06 +0000 [r96776-96836] Luigi Rizzo * channels/console_gui.c: update comments to reflect reality (or at least planned behaviour). minor code cleanups * channels/console_gui.c: resolve a load-time problem avoiding a call to console_do_answer. On passing, fix dialling from the keypad. 2008-01-05 23:05 +0000 [r96645-96743] Russell Bryant * res/snmp/agent.c: Convert this file over the new method of getting the Asterisk version. (I don't have this building on this machine, so caio1982 on IRC is going to test it for me. :) ) * Makefile, funcs/func_version.c, main/manager.c, channels/chan_sip.c, main/Makefile, build_tools/make_version_c (added), include/asterisk/version.h (added), res/res_agi.c, main, main/http.c, build_tools/make_version_h (removed), include/asterisk, main/asterisk.c: Now that the version.h file was getting properly regenerated every time the svn revision changed, every module that used the version was getting rebuilt after every svn update. This severly annoyed me pretty quickly, so I have improved the situation. Now, instead of generating version.h, main/version.c is generated. version.c includes the version information, as well as a couple of API calls for modules to retrieve the version. So now, only version.c will get rebuilt, and the main asterisk binary relinked, which is must faster than rebuilding http.c, manager.c, asterisk.c, relinking the asterisk binary, chan_sip.c, func_version.c, res_agi ... The only minor change in behavior here is that the version information reported by chan_sip, for example, is the version of the Asterisk core, and not necessarily the Asterisk version that the chan_sip module came from. * main/pbx.c: Print out the name of a function being registered in color, just like the name of applications when they get registered. * UPGRADE.txt: Add a note about changing modules.conf since another console channel driver is now present that can not be used at the same time as chan_alsa or chan_oss. * channels/chan_console.c: Add the URL to the home page for portaudio. Also add the location of the svn repository to check out portaudio v19. * /, main/devicestate.c: Merged revisions 96644 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r96644 | russell | 2008-01-04 20:09:19 -0600 (Fri, 04 Jan 2008) | 2 lines Don't pass an empty string as the device name. ........ 2008-01-05 01:05 +0000 [r96621] Kevin P. Fleming * channels/chan_usbradio.c: improve chan_usbradio to use indications just like chan_alsa/chan_oss do now 2008-01-04 23:12 +0000 [r96576] Tilghman Lesher * /, main/devicestate.c: Merged revisions 96575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r96575 | tilghman | 2008-01-04 17:03:40 -0600 (Fri, 04 Jan 2008) | 7 lines Fix the problem of notification of a device state change to a device with a '-' in the name. Could probably do with a better fix in trunk, but this bug has been open way too long without a better solution. Reported by: stevedavies Patch by: tilghman (Closes issue #9668) ........ 2008-01-04 22:57 +0000 [r96574] Jason Parker * /, res/res_features.c: Merged revisions 96573 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes issue #11237) ........ r96573 | qwell | 2008-01-04 16:55:56 -0600 (Fri, 04 Jan 2008) | 4 lines Properly continue in the dialplan if using PARKINGEXTEN and the slot is full. Issue 11237, patch by me. ........ 2008-01-04 19:35 +0000 [r96547] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 96525 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r96525 | tilghman | 2008-01-04 13:27:25 -0600 (Fri, 04 Jan 2008) | 4 lines If you change the bindaddr in sip.conf to a non-bound address and reload, sip goes kablooie. Reported and patched by: one47 (Closes issue #11535) ........ 2008-01-04 17:21 +0000 [r96500] Kevin P. Fleming * channels/chan_zap.c, configure, include/asterisk/autoconfig.h.in, configure.ac, acinclude.m4: [commit message] (closes issue #10393) Reported by: tzafrir Patches: chan_alarm_asterisk.diff uploaded by tzafrir (license 46) (modified by me and added configure script support) 2008-01-04 17:19 +0000 [r96499] Philippe Sultan * res/res_jabber.c: Use SASL DIGEST-MD5 authentication over unsecured network connections only. This authentication mechanism is implemented under the iksemel API, which makes use of GnuTLS, whereas we use OpenSSL. Note : there's ongoing dicsussion at the SASL IETF WG in order to deprecate SASL DIGEST-MD5, see http://ietfreport.isoc.org/ids-wg-sasl.html. 2008-01-04 16:21 +0000 [r96450] Russell Bryant * channels/chan_zap.c, /: Merged revisions 96449 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r96449 | russell | 2008-01-04 10:19:22 -0600 (Fri, 04 Jan 2008) | 7 lines Make use of the temporary channel pointer while the pvt is unlocked. (closes issue #11675) Reported by: flefoll Patches: chan_zap.c.patch-store-owner-before-unlock uploaded by flefoll (license 244) ........ 2008-01-03 23:14 +0000 [r96397-96398] Kevin P. Fleming * Makefile: we have to *always* use a completely silent 'make' invocation for generating the module embedding rules * Makefile: there was no reason to add this define for non-Solaris platforms 2008-01-03 22:46 +0000 [r96395] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 96394 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r96394 | russell | 2008-01-03 16:44:22 -0600 (Thu, 03 Jan 2008) | 3 lines Don't crash if the iax2 pvt structure has been destroyed before we get to this point (closes issue #11672, reported by snuffy, patched by me) ........ 2008-01-03 21:58 +0000 [r96301-96368] Tilghman Lesher * include/asterisk/channel.h: Document recent API addition * res/res_config_pgsql.c, /: Merged revisions 96318 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r96318 | tilghman | 2008-01-03 15:37:02 -0600 (Thu, 03 Jan 2008) | 4 lines Missed initialization caused crash. Reported and fixed by: tiziano (Closes issue #11671) ........ * main/channel.c: Allow the uniqueid to be used for searching for a channel in the list. Reported and initially patched by: michael-fig (Closes issue #11340) 2008-01-03 20:04 +0000 [r96245-96272] Kevin P. Fleming * Makefile, tests/Makefile (added), tests/test_skel.c (added), tests (added): add some simple infrastructure for modules to be used for testing parts of Asterisk * channels/answer.h (removed), channels/ring10.h (removed), channels/busy.h (removed), channels/ringtone.h (removed), channels/Makefile, channels/chan_oss.c, channels/gentone.c (removed), channels: eliminiate sound_thread() and other stuff from chan_oss since Asterisk indications can handle it remove gentone and all the headers containing tones that are no longer needed * channels/chan_alsa.c: coding guidelines cleanup remove background thread and all sound generation mechanisms, as the built-in indications can handle everything that is needed 2008-01-03 14:47 +0000 [r96221] Christian Richter * channels/chan_misdn.c, /: Merged revisions 96198 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r96198 | crichter | 2008-01-03 13:08:40 +0100 (Do, 03 Jan 2008) | 1 line when overlapdial was used and no number was dialed, the call was dropped, now we just jump into the s extension, which makes a lot more sense. ........ 2008-01-03 06:16 +0000 [r96147-96174] Tilghman Lesher * res/res_agi.c: Add coordination between AMI and AGI applications, with an asyncagi method Feature proposed and patched by: moy (Closes issue #11282) * apps/app_mp3.c, apps/app_ices.c, main/asterisk.c: Compatibility fix for OpenBSD Report and fix by: mvanbaak (Closes issue #11669) 2008-01-02 23:48 +0000 [r96103] Mark Michelson * /, apps/app_queue.c: Merged revisions 96102 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r96102 | mmichelson | 2008-01-02 17:46:02 -0600 (Wed, 02 Jan 2008) | 4 lines We need to reset the membername to NULL on each iteration of this loop, otherwise the result is that multiple members can have the same name, since the variable was not reset on each iteration of the loop. ........ 2008-01-02 23:22 +0000 [r96076-96079] Russell Bryant * channels/chan_console.c: Add support for generating a ringing sound on an incoming call. This is a bit of a hack. It just asks the core to generate the same tone that it would when you hear ringback when making an outbound call. But hey, it works, and you get the localized ring tone for the appropriate language set on the channel. * channels/chan_console.c: Note that this module doesn't actually play a ringing sound for an incoming call ... oops * channels/chan_console.c: Show the correct CLI command to answer the call 2008-01-02 22:41 +0000 [r96073] Kevin P. Fleming * channels/chan_zap.c: actually parse and store echocan parameters from zapata.conf... this *should* work 2008-01-02 22:40 +0000 [r96071] Joshua Colp * configure, include/asterisk/autoconfig.h.in, configure.ac: Don't use AST_C_DEFINE_CHECK for the two pthread things that may not actually be definitions, they could be enums for example. 2008-01-02 22:29 +0000 [r96028] Mark Michelson * channels/chan_zap.c: Add curly braces around a compound if statement so that trunk will build properly 2008-01-02 21:51 +0000 [r96019] Kevin P. Fleming * channels/chan_zap.c, configs/zapata.conf.sample: another checkpoint... chan_zap can now use the new ZT_ECHOCAN_PARAMS ioctl if it is present, but doesn't parse any supplied parameters yet (this implementation is not very memory efficient as the parameters and their values will be duplicated for each channel that has the same settings, but we can worry about that later once it is working) 2008-01-02 21:49 +0000 [r96018] Russell Bryant * main/libresample/include/libresample.h: Add doxygen documentation to libresample.h while it's still fresh on my mind 2008-01-02 21:08 +0000 [r95994] Mark Michelson * funcs/func_odbc.c, channels/chan_agent.c, funcs/func_strings.c, apps/app_rpt.c: Change instances of AST_NONSTANDARD_APP_ARGS(foo, bar, ',') to AST_STANDARD_APP_ARGS(foo, bar) (closes issue #11668, reported and patched by mvanbaak) 2008-01-02 20:26 +0000 [r95947] Joshua Colp * /, channels/chan_sip.c: Merged revisions 95946 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r95946 | file | 2008-01-02 16:24:09 -0400 (Wed, 02 Jan 2008) | 4 lines Allocate a SIP refer structure when performing a transfer using BYE with Also so that the transfer information is properly stored. (AST-2008-001) (closes issue #11637) Reported by: greyvoip ........ 2008-01-02 20:23 +0000 [r95944-95945] Mark Michelson * apps/app_queue.c: Since ',' is the standard argument separator in trunk, change app_queue to use AST_STANDARD_APP_ARGS instead of AST_NONSTANDARD_APP_ARGS for determining member data. * include/asterisk/app.h: Fix a typo in a comment. AST_STANDARD_APP_ARGS uses ',' as the separator, not '|'. 2008-01-02 19:47 +0000 [r95893-95939] Kevin P. Fleming * channels/chan_zap.c: clean up hwgain CLI command and improve docs for swgain CLI command * configure, include/asterisk/autoconfig.h.in, configure.ac, acinclude.m4: improve AC_C_DEFINE_CHECK to not try to evaluate the macro being checked for, but just check for its existence finish implementation of check for Zaptel HWGAIN support add check for Zaptel ECHOCANCEL_PARAMS support * codecs/Makefile, include/asterisk/libresample.h (added), codecs/codec_resample.c: and now just to keep the libresample party going... if the functions from libresample are going to be in the main Asterisk binary, it makes sense for the header that defines them to be available without any special CFLAGS and to out-of-tree modules building against /usr/include/asterisk * channels/chan_zap.c: umm... this did not compile on x86-64, and could not possibly have worked on any platform as it was passing string pointers to a function expecting ints 2008-01-02 18:05 +0000 [r95891] Mark Michelson * /, apps/app_queue.c: Merged revisions 95890 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r95890 | mmichelson | 2008-01-02 11:51:22 -0600 (Wed, 02 Jan 2008) | 9 lines A change to improve the accuracy of queue logging in the case where a member does not answer during the specified timeout period. Prior to this change, there was a small chance that the member name recorded in this case would be blank. Also prior to this change, if using the ringall strategy, if no one answered the call during the specified timeout, the member name listed in the queue log would randomly be one of the members that was rung. (closes issue #11498, reported and tested by hloubser, patched by me) ........ 2008-01-02 17:38 +0000 [r95888] Jason Parker * apps/app_osplookup.c: Update osplookup documentation to use commas instead of pipes. Closes issue #11666, patch by Laureano. 2008-01-02 16:20 +0000 [r95864] Russell Bryant * main/Makefile, main/translate.c: For some odd reason, the last set of libresample build changes from Kevin did not work for everyone, but it did for some. This set of changes makes trunk start again for those having problems. Instead of building libresample as a static library, it just links the object files in directly with the asterisk binary. 2008-01-02 14:53 +0000 [r95816-95841] Kevin P. Fleming * channels/Makefile: fix some long-time breakage that kept chan_misdn from being embedded * channels/Makefile: use the proper technique for including submodules so that embedding will work * CHANGES: note that chan_console requires portaudio v19 * configure, configure.ac: actually check for a function present in libiconv (don't know how this test could have worked before) and don't do the check on Linux/GNU systems because libiconv is not present there and attempting to link with '-liconv' always fails (it's not necessary as the iconv functionality is always available) * main/libresample/src/filterkit.h, main/libresample/src/resample.c, main/libresample/win/libresample.dsp, main/libresample/configure, main/libresample/Makefile.in, res/Makefile, main/libresample/configure.in, main/libresample/src, main/libresample/tests/testresample.c, main/libresample/win/libresample.vcproj, main/libresample/tests/compareresample.c, main/libresample/tests, codecs/codec_resample.c, res/res_resample.c (removed), main/libresample/README.txt, main/libresample/src/resamplesubs.c, main/libresample/tests/resample-sndfile.c, main/libresample/src/configtemplate.h, main/libresample/install-sh, main/Makefile, main/translate.c, main/libresample/include, main/libresample/src/resample_defs.h, codecs/Makefile, main/libresample/config.guess, main/libresample/config.sub, main/libresample/win, main/libresample/LICENSE.txt, main/libresample (added), main/libresample/Makefile.asterisk, build_tools/strip_nonapi, res/libresample (removed), main/libresample/src/filterkit.c, main/libresample/include/libresample.h: go back to including libresample in the main Asterisk binary, but this time including a small hack to ensure that it does get linked in (and also modify the strip_nonapi script to leave the resample_ symbols alone) 2008-01-02 11:34 +0000 [r95794] Philippe Sultan * res/res_jabber.c: Set stream flags to zero upon initialization. When the XMPP over TLS/SSL connection resets for some reason, it is wrongly believed as being secured, which makes the re-connection process endlessly fail. This was reported by mvanbaak in issue #11644. 2008-01-02 09:16 +0000 [r95771-95772] Luigi Rizzo * main/loader.c: some cleanup of this code while I am trying to debug a problem with gdb dying while debugging asterisk. The problem seems to be related with a race in the handling of module_list, which in turn is triggeded by calling dlopen() on a system which uses initializers to create locks. * include/asterisk/module.h: There are three instances of the module definition macros, which make maintaining this file very error prone. This commit merges the embedded and !embedded versions, and fixes the C++ version. Eventually we should move to a single version of the macro. Too bad C++ doesn't like the C-style struct initializers .foo = some_value 2008-01-02 04:33 +0000 [r95697-95746] Russell Bryant * res/libresample/src/resample_defs.h, res/libresample/src/resample.c: Don't make libresample print out debugging output * main/translate.c: Make the translation table show slin16 * apps/app_meetme.c: fix a spacing issue introduced in revision 95443. * main/Makefile, res/libresample/README.txt, res/Makefile, res/libresample/install-sh, res/libresample/configure, res/libresample/Makefile.in, res/libresample/include, codecs/Makefile, res/libresample/configure.in, res/libresample/src, res/libresample/config.guess, main/libresample (removed), res/libresample/config.sub, res/libresample/win, codecs/codec_resample.c, res/libresample/LICENSE.txt, res/libresample (added), res/libresample/Makefile.asterisk, res/libresample/tests, res/res_resample.c (added): Instead of linking libresample into the main Asterisk binary, build it as res_resample, and mark codec_resample as dependent upon res_resample. This prevents the linker from optimizing away libresample, and also makes it so the libresample code isn't linked in to multiple places. (I have another module in a branch that needs it, too.) 2008-01-01 23:55 +0000 [r95671-95673] Luigi Rizzo * channels/console_gui.c: call directly the cli command to implement hangup. * channels/vcodecs.c: prevent a panic when destroying a channel with no incoming video. * channels/console_video.c: remove a leftover sleep(1) used for debugging 2008-01-01 23:09 +0000 [r95648] Joshua Colp * codecs/Makefile: Fix building of codec_resample on platforms other then Cygwin. On everything else it actually gets built after codec_resample, so you can't exactly link it in since it doesn't exist. 2008-01-01 22:21 +0000 [r95624-95625] Luigi Rizzo * codecs/Makefile, codecs/codec_resample.c: make codec_resample build on __CYGWIN__, and make it load on FreeBSD (and probably other systems as well). Both need libresample.a to be specified in the linking phase, and cygwin needs as other BSD. The checks for OS-specific headers should really be moved to some common header though. * build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, configure.ac, funcs/func_iconv.c, makeopts.in: implement "configure" checks for libiconv, and add the iconv dependency for func_iconv. This fixes some build issues on CYGWIN and FreeBSD and probably other platforms where libiconv is not there by default 2007-12-31 23:44 +0000 [r95578] Mark Michelson * main/pbx.c, /: Merged revisions 95577 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r95577 | mmichelson | 2007-12-31 17:43:13 -0600 (Mon, 31 Dec 2007) | 9 lines Avoiding a potentially bad locking situation. ast_merge_contexts_and_delete writelocks the conlock, then calls ast_hint_extension, which attempts to readlock the same lock. Recursion with read-write locks is dangerous, so the inner lock needs to be removed. I did this by copying the "guts" of ast_hint_extension into ast_merge_contexts_and_delete (sans the extra lock). (this change is inspired by the locking problems seen in issue #11080, but I have no idea if this is the problematic area experienced by the reporters of that issue) ........ 2007-12-31 22:41 +0000 [r95501-95550] Russell Bryant * codecs/codec_resample.c: Use float.h to fix the build on FreeBSD. Also, add some other platforms as they are likely the same. * channels/chan_console.c: Update chan_console to natively use a 16 kHz sample rate. If it is talking to an 8 kHz endpoint, then codec_resample will automatically be used to properly resample the audio before sending it to/from chan_console. * main/libresample/src/filterkit.h, main/libresample/README.txt, main/libresample/tests/resample-sndfile.c, main/libresample/src/resamplesubs.c, main/Makefile, main/libresample/install-sh, main/libresample/src/configtemplate.h, main/libresample/src/resample.c, main/libresample/win/libresample.dsp, main/libresample/configure, main/libresample/Makefile.in, main/libresample/include, CHANGES, main/libresample/src/resample_defs.h, main/libresample/configure.in, main/libresample/src, main/libresample/config.guess, codecs/Makefile, main/libresample/tests/testresample.c, codecs/slin_resample_ex.h (added), main/libresample/config.sub, main/libresample/win, main/libresample/win/libresample.vcproj, main/libresample/LICENSE.txt, main/libresample (added), main/libresample/Makefile.asterisk, main/libresample/tests, main/libresample/tests/compareresample.c, codecs/codec_resample.c (added), main/libresample/src/filterkit.c, main/libresample/include/libresample.h: Merge changes from team/russell/codec_resample This commit imports libresample for use in Asterisk. It also adds a new codec module, codec_resample. This module uses libresample to re-sample signed linear audio between 8 kHz and 16 kHz. It also provides an alternative for converting between 16 kHz G.722 and 8 kHz signed linear when using G.722, which will likely be useful as some people have complained about volume issues when the current codec_g722 converts to 8 kHz signed linear. But, to test this, you will have to disable the g722-to-slin and g722-to-slin16 translators in codec_g722.c. 2007-12-31 20:33 +0000 [r95490] Tilghman Lesher * /, funcs/func_env.c: Merged revisions 95470 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r95470 | tilghman | 2007-12-31 14:27:26 -0600 (Mon, 31 Dec 2007) | 3 lines Allow the default "0" to be returned if the STAT fails (Closes issue #11659) ........ 2007-12-31 18:46 +0000 [r95443] Mark Michelson * apps/app_meetme.c: Fix a compiler warning (closes issue #11658, reported and patched by eliel) 2007-12-31 16:13 +0000 [r95383-95412] Russell Bryant * configs/console.conf.sample (added), configs/modules.conf.sample, channels/chan_console.c (added), CHANGES: Merge the main set of changes from team/russell/chan_console. Add a new console channel driver, chan_console, which is a console channel driver that uses portaudio as a cross platform audio interface. It was written to provide a console channel driver that works with Mac CoreAudio, but it supports a number of other audio interfaces, as well, including OSS and ALSA. It could one day be the single console channel driver, but does not yet have as many features as chan_oss. * include/asterisk/channel.h: fix a spelling error in a comment * include/asterisk/config.h: Add CV_STRINGFIELD() macro. This lets you set a config variable to a string field. (from team/russell/chan_console) * configure, include/asterisk/autoconfig.h.in: Regenerate configure script to include check for portaudio. * build_tools/menuselect-deps.in, configure.ac, makeopts.in: Add configure script checking for portaudio. 2007-12-29 02:02 +0000 [r95262-95313] Luigi Rizzo * channels/vcodecs.c, channels/console_video.c, channels/Makefile, channels/console_video.h, channels/vgrabbers.c (added): Move grabbers definitions to a separate file, vgrabbers.c, so it is easier to add more entries. This required moving struct grab_desc to the common header, and adding an entry in the Makefile. On passing, cleanup some comments and file headers (some are still missing). * channels/console_gui.c, channels/console_video.c: virtualize the interface for video grabbers, which should make it easier to add support for more grabbers (V4L2, firewire, and so on). * channels/console_video.c: Add a few entries up to 1408x1152 in the table of known video resolutions. This makes it very convenient to enlarge images using the right-click on the video window. * channels/vcodecs.c, channels/console_video.c: change the interface of video encapsulation routines, they only need the buffer and mtu as input. * channels/console_gui.c, channels/vcodecs.c, channels/console_video.c, channels/console_video.h: various rearrangements and renaming of console_video stuff 2007-12-28 18:39 +0000 [r95233] Mark Michelson * apps/app_queue.c: The diff for this change looks really bad, but all I did here was decrease the indentation of most of the queue_exec function by reversing the logic of an if statement. This change makes the function comply better with the coding guidelines. Since this change is purely a cosmetic change to the code, I am only committing the change to trunk. 2007-12-28 18:26 +0000 [r95192] Russell Bryant * /, channels/chan_sip.c: Merged revisions 95191 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r95191 | russell | 2007-12-28 12:24:59 -0600 (Fri, 28 Dec 2007) | 6 lines Remove duplicate increment of the header count in the add_header() function. (closes issue #11648) Reported by: makoto Patch provided by sergee, committed patch by me, inspired by comments from putnopvut ........ 2007-12-28 16:12 +0000 [r95167] Mark Michelson * apps/app_amd.c, CHANGES: Some changes to app_amd. The channel name is printed in verbose messages maximumWordLength option added. Duration of words that do not meet the minimum word duration will be logged The duration of pre-greeting silence will be logged Only consider us in the greeting if we actually detected a valid word duration. (closes issue #11650, reported and patched by davevg) 2007-12-28 08:57 +0000 [r95139] Luigi Rizzo * channels/console_video.c: fix a small bug in printing out geometries - wrong input. 2007-12-28 00:17 +0000 [r95096] Mark Michelson * /, apps/app_queue.c: Merged revisions 95095 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r95095 | mmichelson | 2007-12-27 18:16:15 -0600 (Thu, 27 Dec 2007) | 8 lines I found a bug while browsing the queue code and managed to reproduce it in a small setup. If a queue uses the ringall strategy, it was possible through unfortunate coincidence for a single member at a given penalty level to make app_queue think that all members at that penalty level were unavailable and cause the members at the next penalty level to be rung. With this patch, we will only move to the next penalty level if ALL the members at a given penalty level are unreachable. ........ 2007-12-27 23:32 +0000 [r95073] Luigi Rizzo * apps/app_dictate.c, apps/app_mp3.c, apps/app_voicemail.c: remove more unnecessary casts for NULL. main/say.c is a big offender in this respect. 2007-12-27 23:28 +0000 [r95070] Jason Parker * doc/asterisk.8, main/asterisk.c: Fix -s socket option, and document it as well. Closes issue #11645, patch by Laureano. 2007-12-27 23:13 +0000 [r95068-95069] Luigi Rizzo * apps/app_ices.c, apps/app_queue.c, apps/app_voicemail.c: NULL does not need to be cast to (char *) * channels/chan_oss.c: remove useless casts 2007-12-27 21:41 +0000 [r95025] Russell Bryant * main/channel.c, /: Merged revisions 95024 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r95024 | russell | 2007-12-27 15:40:02 -0600 (Thu, 27 Dec 2007) | 9 lines Don't report a syntax error when an empty string is passed to ast_get_group. Just return 0. (closes issue #11540) Reported by: tzafrir Patches: group_empty.diff uploaded by tzafrir (license 46) -- slightly changed by me ........ 2007-12-27 20:11 +0000 [r94978] Mark Michelson * /, main/io.c: Merged revisions 94977 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94977 | mmichelson | 2007-12-27 14:09:06 -0600 (Thu, 27 Dec 2007) | 3 lines Fixing a typo in a comment. ........ 2007-12-27 17:34 +0000 [r94908-94934] Joshua Colp * /, channels/chan_h323.c: Merged revisions 94924 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94924 | file | 2007-12-27 13:32:15 -0400 (Thu, 27 Dec 2007) | 6 lines Include types.h in chan_h323 as without it it can not be compiled on some operating systems like FreeBSD to name one. (closes issue #11585) Reported by: sobomax Patches: chan_h323.c.diff uploaded by sobomax (license 359) ........ * /, channels/chan_sip.c: Merged revisions 94905 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94905 | file | 2007-12-27 13:27:11 -0400 (Thu, 27 Dec 2007) | 4 lines Use ast_strlen_zero to see if our_contact is set or not on the dialog. It is possible for it to be a pointer to NULL. (closes issue #11557) Reported by: FuriousGeorge ........ 2007-12-27 17:26 +0000 [r94904] Luigi Rizzo * channels/console_gui.c, channels/console_video.c: more localization of gui stuff 2007-12-27 17:18 +0000 [r94903] Mark Michelson * doc/manager_1_1.txt: Adding documentation for new manager actions and events in app_queue 2007-12-27 16:51 +0000 [r94902] Luigi Rizzo * CHANGES: clarify the type of video support in chan_oss 2007-12-27 16:11 +0000 [r94830-94877] Russell Bryant * codecs/codec_g722.c: I went looking for where we downloaded the g722 implementation and came across these two links. So, I'm adding them so they are available for reference later. * /, main/translate.c, include/asterisk/translate.h: Merged revisions 94828-94829 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94828 | russell | 2007-12-27 08:33:21 -0600 (Thu, 27 Dec 2007) | 9 lines Change ast_translator_best_choice() to only pay attention to audio formats. This fixes a problem where Asterisk claims that a translation path can not be found for channels involving video. (closes issue #11638) Reported by: cwhuang Tested by: cwhuang Patch suggested by cwhuang, with some additional changes by me. ........ r94829 | russell | 2007-12-27 08:44:29 -0600 (Thu, 27 Dec 2007) | 2 lines Use the constant that I really meant to use here ... ........ 2007-12-27 09:13 +0000 [r94826-94827] Olle Johansson * funcs/func_dialplan.c: This function checks more than just contexts... * apps/app_pickupchan.c: - Add Copyright - Doxygen fixes Note: - This application needs better documentation and a RESULT code in the dialplan. 2007-12-27 01:03 +0000 [r94825] Kevin P. Fleming * main/manager.c, /: Merged revisions 94824 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94824 | kpfleming | 2007-12-26 18:01:47 -0700 (Wed, 26 Dec 2007) | 2 lines make this comment explain the situation in an even more explicit fashion ........ 2007-12-27 00:48 +0000 [r94819-94823] Luigi Rizzo * channels/console_gui.c: more steps to decouple the gui from the rest of the code. * channels/console_gui.c, channels/console_video.c, channels/console_video.h: Enable building the code even if SDL is not present (similarly, SDL is also detected at runtime). Now we should be able to stream video even without a rendering device (useful for remote monitoring). * channels/console_gui.c, channels/console_video.c: more localizations around sdl_setup * channels/console_gui.c: use fread instead of mmap to read in the comment area from the keypad. fread is simpler and more portable, and there is no performance gain in using mmap. * images/kpad2.jpg: update the region description with an empty line at the beginning. 2007-12-26 22:38 +0000 [r94818] Tilghman Lesher * build_tools/cflags.xml, channels/chan_zap.c: Allow more spans than 32. Also, rearrange compiler flags so the most often used flags appear closer to the top. Reported by: tzafrir Patch by: tzafrir,tilghman (Closes issue #11528) 2007-12-26 22:29 +0000 [r94817] Luigi Rizzo * channels/console_gui.c, channels/console_video.c: another bunch of gui localizations 2007-12-26 22:14 +0000 [r94814] Jason Parker * apps/app_exec.c: Make 'else' argument to ExecIf optional. Clean up the description and usage text a bit. Closes issue #11564, patch by pnlarsson (with some extra cleanup by me). 2007-12-26 22:10 +0000 [r94810-94813] Luigi Rizzo * channels/console_gui.c, channels/console_video.c: more localization of sdl stuff * channels/console_gui.c, channels/console_video.c, channels/console_video.h: move more gui stuff into console_gui.c 2007-12-26 20:49 +0000 [r94809] Tilghman Lesher * main/manager.c, /: Merged revisions 94808 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94808 | tilghman | 2007-12-26 14:43:38 -0600 (Wed, 26 Dec 2007) | 6 lines Workaround for what is probably a glibc bug (but we'll see this crop up again and again, if we don't add the workaround). Reported by: rolek Patch by: tilghman (Closes issue #11601, closes issue #11426) ........ 2007-12-26 20:02 +0000 [r94806] Jason Parker * pbx/pbx_loopback.c, apps/app_zapbarge.c, pbx/pbx_spool.c, apps/app_authenticate.c, apps/app_zapscan.c, apps/app_zapras.c, apps/app_alarmreceiver.c, apps/app_amd.c, pbx/pbx_realtime.c, pbx/pbx_dundi.c, apps/app_zapateller.c, pbx/pbx_config.c, pbx/pbx_gtkconsole.c, apps/app_adsiprog.c, apps/app_cdr.c: Use defined return values in load_module in more places. (closes issue #11096) Patches: pbx_config.c.patch uploaded by moy (license 222) pbx_dundi.c.patch uploaded by moy (license 222) pbx_gtkconsole.c.patch uploaded by moy (license 222) pbx_loopback.c.patch uploaded by moy (license 222) pbx_realtime.c.patch uploaded by moy (license 222) pbx_spool.c.patch uploaded by moy (license 222) app_adsiprog.c.patch uploaded by moy (license 222) app_alarmreceiver.c.patch uploaded by moy (license 222) app_amd.c.patch uploaded by moy (license 222) app_authenticate.c.patch uploaded by moy (license 222) app_cdr.c.patch uploaded by moy (license 222) app_zapateller.c.patch uploaded by moy (license 222) app_zapbarge.c.patch uploaded by moy (license 222) app_zapras.c.patch uploaded by moy (license 222) app_zapscan.c.patch uploaded by moy (license 222) 2007-12-26 20:01 +0000 [r94805] Luigi Rizzo * channels/console_gui.c, channels/vcodecs.c, channels/console_video.c, channels/console_video.h: more preparation for untangling of the various console_video stuff 2007-12-26 19:09 +0000 [r94796-94802] Russell Bryant * main/autoservice.c, /: Merged revisions 94801 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94801 | russell | 2007-12-26 13:04:31 -0600 (Wed, 26 Dec 2007) | 4 lines Just in case the AST_FLAG_END_DTMF_ONLY flag was already set before starting autoservice, remember it and ensure that the channel has the same setting when autoservice gets stopped. (pointed out by d1mas, patched up by me) ........ * funcs/func_dialplan.c (added), CHANGES: Add a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for the existence of a dialplan target. (closes issue #11579) Reported by: irroot Patches: func_dialplan2.c uploaded by irroot (license 52) -- Additional changes by me. * main/autoservice.c, /: Merged revisions 94797 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94797 | russell | 2007-12-26 12:46:39 -0600 (Wed, 26 Dec 2007) | 4 lines When a channel is in autoservice, mark a flag on the channel that says that we only care about the END of a digit. That way, no magic digit emulation stuff will happen when all we're doing is queueing up END frames. ........ * main/channel.c: Leave a note for a minor bug that was pointed out by d1mas 2007-12-26 18:05 +0000 [r94795] Tilghman Lesher * channels/chan_zap.c: Convert raw bits for callprogress bitfield to use constants, for greater code clarity Reported by: dimas Patch by: dimas (Closes issue #11280) 2007-12-26 17:26 +0000 [r94787-94794] Russell Bryant * /, res/res_features.c: Merged revisions 94793 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94793 | russell | 2007-12-26 11:24:17 -0600 (Wed, 26 Dec 2007) | 3 lines Don't try to send a parked call back to itself. (closes issue #11622, reported by djrodman, patched by me) ........ * Makefile, /: Merged revisions 94789 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94789 | russell | 2007-12-26 11:00:03 -0600 (Wed, 26 Dec 2007) | 5 lines List include/asterisk/version.h as a .PHONY target because we want the commands listed for this target to be executed regardless of whether the file exists or not. This fixes having the version not up to date when running from svn. (closes issue #11619, reported by plack, fixed by me) ........ * main/autoservice.c, /: Merged revisions 94790 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94790 | russell | 2007-12-26 11:06:26 -0600 (Wed, 26 Dec 2007) | 5 lines Don't store DTMF BEGIN frames while a channel is in autoservice. It's just going to make ast_read() do a lot of extra work when the channel comes back out of autoservice. (closes issue #11628, patched by me) ........ * channels/chan_iax2.c: Fix a bug in peer handling that caused multiple instances of a peer to end up in the peers container after a reload. Somehow, this bug doesn't exist in 1.4 ... (closes issue #11626) (reported by pnlarsson, additional info from mvanbaak, fixed by me) * utils: update svn:ignore for astcanary 2007-12-26 15:58 +0000 [r94782] Mark Michelson * configs/extconfig.conf.sample, main/logger.c, CHANGES: Adding support for storing the queue log entries in a realtime backend. (closes issue #11625, reported and patched by sergee) Thank you very much to sergee for adding this new feature! 2007-12-26 10:14 +0000 [r94774] Luigi Rizzo * channels/console_gui.c (added), channels/vcodecs.c (added), channels/console_video.c: Split console_video.c so that video codecs and gui functions are in separate files (still #include'd because of tangling in the data structures, but this is going to be cleaned up). The video grabbing functions still need to be moved to a separate file. 2007-12-25 04:10 +0000 [r94771-94773] Tilghman Lesher * apps/app_pickupchan.c (added): Add pickup by channel (Closes issue #11161) * channels/chan_zap.c, configs/zapata.conf.sample: Change the abbreviated TON from 'A' to 'V', since 'A' is a legitimate DTMF character. Also, fix the documentation to match the code. * res/res_agi.c: Add channel thread ID to the information passed to AGI. Reported by: dror99 Patch by: tilghman (Closes issue #11162) 2007-12-24 19:43 +0000 [r94764-94768] Tilghman Lesher * main/channel.c, /: Merged revisions 94767 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94767 | tilghman | 2007-12-24 13:36:59 -0600 (Mon, 24 Dec 2007) | 5 lines Race: we need to wait to queue a NewChannel event until after the channel is inserted into the channel list. The reason is because some manager users immediately queue requests from the channel when they see that event and are confused when Asterisk reports no such channel. (Closes issue #11632) ........ * /: Merged revisions 94763 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94763 | tilghman | 2007-12-24 09:39:56 -0600 (Mon, 24 Dec 2007) | 5 lines Another bit of bad logic in realtime_peer Reported by: dimas Patch by: dimas (Closes issue #11631) ........ 2007-12-23 14:51 +0000 [r94713-94741] Luigi Rizzo * channels/console_video.c, channels/console_video.h: support sdl_videodriver to send output to x11/aalib/console * channels/console_video.c: move reading info from the keypad to a separate function. Remove an unused keypad field and some debugging messages. Adjust formatting on config file parsing * channels/console_video.c: make sure the minimum surface depth is 16bpp so we can create YUVoverlays. With this change we can do setenv SDL_VIDEODRIVER aalib and output to an ascii window (which is still in an X11 window). If you also do unsetenv DISPLAY then the output goes into the main asterisk window, unfortunately it interferes with the normal output so you don't see much. In any case, i don't think we are very far away from having a working xterm videophone! * channels/Makefile: avoid rebuilding dependent files if the generated busy.h and ringtone.h do not change. Ths masks (but does not solve) a but that i am seeing in doing a 'gmake install' without donig a 'gmake all' first. 2007-12-23 01:38 +0000 [r94662] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 94660 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94660 | tilghman | 2007-12-22 19:21:03 -0600 (Sat, 22 Dec 2007) | 2 lines Argh... I suppose third time's the charm. ........ 2007-12-22 22:44 +0000 [r94615-94638] Luigi Rizzo * configs/oss.conf.sample, channels/console_video.c: Change the name of config file entries for keypad regions from 'keypad_entry' to 'region'. Fix the example file accordingly. Also make some fixes in the code do reset entries on reload of the keypad. The recently committed kpad2.jpg has the correct names. * images/kpad2.jpg (added): add a sample keypad (with annotations) for console video * channels/console_video.c, channels/Makefile, channels/chan_oss.c, channels/console_video.h (added): Build console_video support by linking in, as opposed to including, console_video.c This will ease the task of splitting console_video.c into its components (V4L and X11 grabbers, various video codecs and packetizers, SDL), as well as ease future extensions (e.g. additional video sources, codecs and rendering engines). For the time being nothing changes for users: video support is off by default, and requires -DHAVE_VIDEO_CONSOLE on the command line to be included (if SDL and FFMPEG are available). 2007-12-21 21:19 +0000 [r94593] Mark Michelson * apps/app_voicemail.c: Something I've been itching to do for a while now. A minor optimization in app_voicemail. Since the dtable in base_encode always gets populated with the same values every time and never changes, make it static and const and only initialize it once. Also, there's no reason to define BASEMAXINLINE twice, so remove the redundant #define. 2007-12-21 20:50 +0000 [r94549-94551] Matthew Fredrickson * channels/chan_zap.c: We should only clear this value if we have to * channels/chan_zap.c: Commit non TCP transport part of #11506. Includes numerous additional parameters, as well as RLT support for DMS type switches 2007-12-21 20:38 +0000 [r94542-94548] Mark Michelson * res/res_config_sqlite.c: Store dates using local time instead of UTC (closes issue #11610, reported and patched by rbraun_performatique) * apps/app_queue.c: Fix a memory leak when reloading queue rules. * CHANGES: The one documentation source I forgot to update after the merge of the queue-penalty branch was the CHANGES file. No longer! * apps/app_voicemail.c: Lots of coding guidelines cleanup. * /, apps/app_voicemail.c: Merged revisions 94540 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94540 | mmichelson | 2007-12-21 14:11:34 -0600 (Fri, 21 Dec 2007) | 8 lines Better quota support for using IMAP storage voicemail (closes issue #11415, reported by jaroth) (closes issue #11152, reported by selsky) Patch provided by jaroth ........ 2007-12-21 20:12 +0000 [r94541] Jason Parker * codecs/codec_lpc10.c, codecs/codec_g722.c, codecs/codec_adpcm.c, codecs/codec_alaw.c, codecs/codec_speex.c, codecs/codec_g726.c, codecs/codec_gsm.c, codecs/codec_ulaw.c, codecs/codec_zap.c: codecs.conf really shouldn't be mandatory.. it never had been before, so let's go back to being optional. A big "thank you" to pnlarsson on IRC for allowing me access to his system to debug this. Closes issue #11584. 2007-12-21 20:01 +0000 [r94477-94539] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 94538 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94538 | mmichelson | 2007-12-21 13:59:45 -0600 (Fri, 21 Dec 2007) | 5 lines The mail_copy c-client function does not expect a full imap mailbox string, just the name of the mailbox. (closes issue #11419, reported and patched by jaroth, with additional patchwork from me) ........ * main/dial.c: AST_LIST_REMOVE_CURRENT only takes one argument in trunk * main/dial.c, /: Merged revisions 94468 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94468 | mmichelson | 2007-12-21 10:49:35 -0600 (Fri, 21 Dec 2007) | 6 lines Since we are freeing list elements within a list traversal, we need to use the safe traversal and remove the item from the list before freeing it. (closes issue 11612, reported by dtyoo) ........ 2007-12-21 16:12 +0000 [r94463-94465] Mark Michelson * /, apps/app_queue.c: Merged revisions 94464 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94464 | mmichelson | 2007-12-21 10:11:44 -0600 (Fri, 21 Dec 2007) | 3 lines Removing a debug message I accidentally just committed ........ * /, main/say.c, apps/app_queue.c: Merged revisions 94420 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94420 | mmichelson | 2007-12-21 09:45:14 -0600 (Fri, 21 Dec 2007) | 5 lines Fixing Portuguese syntax for saying dates and times. Also some coding guidelines cleanup. (closes issue #11599, reported and patched by caio1982, coding guidelines cleanup by me) ........ 2007-12-21 15:14 +0000 [r94419] Tilghman Lesher * /, main/asterisk.c: Merged revisions 94418 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94418 | tilghman | 2007-12-21 09:07:42 -0600 (Fri, 21 Dec 2007) | 2 lines Fix for restart-as-user problem reported via the -dev list ........ 2007-12-21 01:14 +0000 [r94345-94396] Mark Michelson * apps/app_queue.c: Moved the update of the queue_ent's rule list to just before we try to call queue members. This allows for the change in penalty levels to be executed at the most logical time frame. * configs/queues.conf.sample, doc/tex/channelvariables.tex, apps/app_queue.c, configs/queuerules.conf.sample (added): Merging the queue-penalty branch. In short, this allows one to dynamically adjust the QUEUE_MAX_PENALTY and the newly introduced QUEUE_MIN_PENALTY during a call depending on the amount of time passed. The purpose is to allow the call to open up to more (or maybe just different) members without the caller's losing his place in the queue. See configs/queuerules.conf.sample for an example of how to set up queue rules and configs/queues.conf.sample for how to associate a rule with a queue. Along with the functional changes, new CLI and manager commands exist to show the rules defined and there is an additional CLI command to reload the queue rules. Future enhancements that may be made: support for realtime queue rules and support for dynamically adding a rule through the manager or CLI. Also a manager command to reload the queue rules (I'll probably write this myself very soon). * apps/app_voicemail.c: The changes to header inclusion in trunk broke compilation of app_voicemail when using IMAP storage. The reason is that c-client has its own definitions for LOG_WARNING and LOG_DEBUG, so we need to be sure to include asterisk's definitions last so that we use the proper values in app_voicemail. (closes issue #11437, reported by blitzrage, patch suggested by blitzrage) 2007-12-20 22:39 +0000 [r94320] Russell Bryant * configs/zapata.conf.sample: Add a bit more to the description of the "mwimonitor" option. 2007-12-20 22:28 +0000 [r94319] Steve Murphy * build_tools/make_buildopts_h: closes issue #11287; thanks to snuffy for this fix, which will surely make all solaris owners shout praises to his name. 2007-12-20 20:25 +0000 [r94252-94257] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 94256 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r94256 | russell | 2007-12-20 14:22:22 -0600 (Thu, 20 Dec 2007) | 13 lines Merged revisions 94255 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r94255 | russell | 2007-12-20 14:21:41 -0600 (Thu, 20 Dec 2007) | 5 lines Fix another potential seg fault ... (closes issue #11606) Reported by: dimas ........ ................ * channels/chan_zap.c, /: Merged revisions 94251 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94251 | russell | 2007-12-20 14:08:42 -0600 (Thu, 20 Dec 2007) | 10 lines Fix a deadlock in d-channel handling in chan_zap. This deadlock was introduced by the fix to ensure that channels are properly locked when handling channel variables. There were sections of this code where the channel pvt was locked before the channel lock, when in fact it _must_ be the other way around. (closes issue #11582) Reported by: bugi ........ 2007-12-20 12:56 +0000 [r94168-94191] Luigi Rizzo * channels/chan_usbradio.c, include/asterisk/config.h, channels/console_video.c, channels/chan_oss.c: add some macros to simplify parsing the config file, see description in config.h . They are a variant of the set of macros i used in chan_oss.c, structured in a way to be more robust to the presence of spurious ';' - basically, they define wrappers for 'do {' and '} while (0)', plus some helper functions to deal with simple cases such as ast_copy_string, ast_malloc, strtoul, ast_true ... The prefix (CV_ as 'Config Variable') tries to be easy to remember and has been chosen to not conflict with other existing macros in the tree. For the time being, I have only updated the three source files in the tree that used the old M_* macros. Hopefully, more files will be converted. NOTE: I understand that inventing my own dialect of C is generally wrong; however, the lack of adequate support in the language encourages lazy programming practices (such as ignoring errors, bounds, etc.) and this increases the chance of vulnerability in the code, especially because we are parsing user input here. Hopefully, these macros and the use of ast_parse_arg (in config.h) should encourage the programmer to write more robust code. * include/asterisk/paths.h, res/snmp/agent.c, utils/ael_main.c, utils/extconf.c, main/asterisk.c, utils/conf2ael.c: modify http://svn.digium.com/view/asterisk?view=rev&rev=93603 so that paths and filename are writable by asterisk.c without causing segfaults. This involves defining the variables as const char *, and having them point to as static, writable buffer defined in asterisk.c On passing, fix some errors in using these variables in some files in utils/ , and in res/snmp/agent.c which was redefining a variable without using paths.h (not applicable to 1.4) 2007-12-19 23:17 +0000 [r94123-94124] Mark Michelson * apps/app_queue.c: 1. Unify the check for a penalty < 0 into the set_member_penalty code. 2. Fix an error when checking the CLI command for setting a member's penalty. 3. Fix a logging error if the incorrect parameter was the queue name or interface. (closes issue #11544, reported and patched by Laureano) * /, res/res_monitor.c: Merged revisions 94122 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94122 | mmichelson | 2007-12-19 17:02:22 -0600 (Wed, 19 Dec 2007) | 6 lines Sox versions 13.0.0 and newer do not have "soxmix" and instead use sox -m. res_monitor needs to use this if the user does not have soxmix. (closes issue #11589, reported by amessina, patch inspired by amessina but with a flourish from me) ........ 2007-12-19 22:51 +0000 [r94085] Russell Bryant * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 94077 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94077 | russell | 2007-12-19 16:48:48 -0600 (Wed, 19 Dec 2007) | 4 lines Check for the existence of the soxmix application on the target platform and have the result available in autoconfig.h. (part of issue #11589) ........ 2007-12-19 20:20 +0000 [r94052-94053] Tilghman Lesher * apps/app_voicemail.c: Add 'voicemail reload' command. Reported by: eliel Patch by: eliel (Closes issue #11365) * apps/app_waituntil.c (added): Add contributed WaitUntil app. Original code by pprindeville, updated for trunk by tilghman. (Closes issue #11487) 2007-12-19 19:29 +0000 [r94029] Russell Bryant * include/asterisk/time.h: Add a couple of new time API calls - ast_tvdiff_sec and ast_tvdiff_usec (closes issue #11270) Reported by: dimas Patches: tvdiff_us-4.patch uploaded by dimas (license 88) 2007-12-19 17:58 +0000 [r94002] Luigi Rizzo * channels/console_video.c: Add instructions on how to generate your own font. 2007-12-19 17:31 +0000 [r93956] Joshua Colp * /: Merged revisions 93955 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r93955 | file | 2007-12-19 13:29:20 -0400 (Wed, 19 Dec 2007) | 2 lines Make the 1.4 builders happy, ensure var is NULL. ........ 2007-12-19 17:13 +0000 [r93952] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 93949 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r93949 | tilghman | 2007-12-19 11:04:13 -0600 (Wed, 19 Dec 2007) | 3 lines Avoid segfault in chan_iax when peer isn't defined (Closes issue #11602) ........ 2007-12-19 17:09 +0000 [r93925-93950] Luigi Rizzo * main/utils.c, include/asterisk/strings.h: Add a new API function, written at least twice in app_voicemail.c and likely in other places too. This is quite useful when placing mail/html stuff in config files. /*! \brief Convert some C escape sequences (\b\f\n\r\t) into the equivalent characters. \brief s The string to be converted (will be modified). \return The converted string. */ char *ast_unescape_c(char *s); * include/asterisk/config.h, main/config.c: add support for PARSE_DOUBLE, and remove identifiers for types not supported (INT16 and UINT16) 2007-12-19 09:20 +0000 [r93899] Olle Johansson * CHANGES: Reorganize CHANGES a bit. The "misc" section grew too large... 2007-12-19 08:57 +0000 [r93898] Luigi Rizzo * configure, include/asterisk/autoconfig.h.in, configure.ac, acinclude.m4, makeopts.in: Properly document AST_EXT_TOOL_CHECK() and use it to check for NETSMP and GTK (GTK is not used thoug). AST_EXT_TOOL_CHECK() could be used for checking curl status as well, perhaps with a small addition because we currently seem to require a curl version greater than X.Y.Z Add a NETSMP_INCLUDE entry in makeopts.in We don't have yet any macros for using pkg-config to check for a specific package (right now there is only gtk2+ in the category). 2007-12-19 08:57 +0000 [r93897] Olle Johansson * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Adding the ability to specify the To: header in an outbound INVITE by adding an exclamation mark to the dial string. This patch also exists for 1.4 in the fixtoheader-1.4 branch and has been in production for quite some time. 2007-12-19 08:12 +0000 [r93875] Luigi Rizzo * res/snmp/agent.c: make netsmp build under AST_DEVMODE. Description, included in the source, is below. I should note that the PACKAGE_* macros that asterisk defines in autoconfig.h are not used anywhere in the tree so they should just be removed. /* * There is some collision collision between netsmp and asterisk names, * causing build under AST_DEVMODE to fail. * * The following PACKAGE_* macros are one place. * Also netsnmp has an improper check for HAVE_DMALLOC_H, using * #if HAVE_DMALLOC_H instead of #ifdef HAVE_DMALLOC_H * As a countermeasure we define it to 0, however this will fail * when the proper check is implemented. */ No 2007-12-19 07:01 +0000 [r93854] Olle Johansson * CHANGES, main/asterisk.c, doc/asterisk.sgml: Add option for starting remote Asterisk by naming the actual runtime socket instead of pointing to configuration file with -C Reported by: sobomax Patches: asterisk.c.diff.trunk uploaded by sobomax (license 359) doc changes by committer (closes issue #11598) 2007-12-19 00:09 +0000 [r93827] Dwayne M. Hubbard * apps/app_osplookup.c: add missing header file 2007-12-18 23:38 +0000 [r93804-93805] Tilghman Lesher * main/asterisk.c: Making the canary error message a little more obvious. * utils/Makefile, utils/astcanary.c (added), main/asterisk.c: Add a canary process, for high priority mode (asterisk -p) to ensure that if Asterisk goes into a busy loop, the machine will be recoverable. We'd still need to do a restart to put Asterisk back into high priority mode, but at least a reboot won't be required. (Closes issue #11559) 2007-12-18 21:13 +0000 [r93741] Olle Johansson * channels/chan_sip.c: Move some warnings away to debug since some devices send a packet with a silly string as a NAT keepalive packet. 2007-12-18 18:39 +0000 [r93672] Tilghman Lesher * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions 93668 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r93668 | tilghman | 2007-12-18 12:29:39 -0600 (Tue, 18 Dec 2007) | 10 lines Merged revisions 93667 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r93667 | tilghman | 2007-12-18 12:23:06 -0600 (Tue, 18 Dec 2007) | 2 lines Fixing AST-2007-027 (Closes issue #11119) ........ ................ 2007-12-18 18:20 +0000 [r93666] Luigi Rizzo * include/asterisk/paths.h: remove a leftover line with only a '#' (wonder why the compiler does not complain!) and variables that are only used in asterisk.c 2007-12-18 17:05 +0000 [r93626] Mark Michelson * main/channel.c, /: Merged revisions 93625 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r93625 | mmichelson | 2007-12-18 11:02:48 -0600 (Tue, 18 Dec 2007) | 6 lines Rework deadlock avoidance used in ast_write, since it meant that agent channels which were being monitored had one audio file recorded and one empty audio file saved. (closes issue #11529, reported by atis patched by me) ........ 2007-12-18 10:24 +0000 [r93558-93603] Luigi Rizzo * include/asterisk/paths.h, channels/chan_sip.c, res/res_crypto.c, utils/ael_main.c, utils/extconf.c, main/asterisk.c, res/res_monitor.c, utils/conf2ael.c: make configuration variable const so they are not accidentally modified. This requires casting the strings in asterisk.c when writing to them, so we do it through a macro to do it consistently. * channels/chan_unistim.c, res/res_crypto.c, main/astmm.c, apps/app_ices.c, utils/extconf.c, channels/chan_iax2.c, main/asterisk.c, main/config.c, main/db.c, apps/app_adsiprog.c, cdr/cdr_csv.c: remove unnecessary (char *) casts for ast_config_AST_* variables. There are some left in the .flex files, left to the maintainer... * build_tools/make_defaults_h, main/asterisk.c: Rename the macros in defaults.h - they are not meant to be globally visible. Document the fact that DEFAULT_TMP_DIR cannot be overridden from the default configuration (this needs to be fixed, as you could have a totally different spooldir configured at runtime, and yet DEFAULT_TMP_DIR keeps the compile-time default). Remove two unused entries for sounds and images. * Makefile.moddir_rules: make the code match documentation - now you can specify multiple words in MODULE_PREFIX. * CREDITS: Name the people responsible for some recent contributions to the tree. * Makefile: Two small changes: + document the difference between "A=foo make ..." and "make A=foo ..." and suggest using COPTS/LDOPTS if you need to use the second form to pass compiler and loader flags; + define only in one place the environment used to build stuff in menuselect/ 2007-12-18 07:56 +0000 [r93557] Olle Johansson * doc/CODING-GUIDELINES: A minor update, caused by a recent bug report ;-) 2007-12-18 07:22 +0000 [r93536] Luigi Rizzo * doc/CODING-GUIDELINES: small documentation update (nothing important). 2007-12-18 02:57 +0000 [r93514] Joshua Colp * channels/chan_unistim.c: You... will... build! I say so and therefore you will. 2007-12-18 02:42 +0000 [r93493] Kevin P. Fleming * channels/chan_unistim.c, include/asterisk/threadstorage.h: minor cleanups 2007-12-17 23:10 +0000 [r93464] Luigi Rizzo * channels/chan_unistim.c: fix building under cygwin. At this point WINARCH should go away. 2007-12-17 22:54 +0000 [r93405] Luigi Rizzo * channels/chan_unistim.c: remove some unnecessary includes 2007-12-17 22:50 +0000 [r93390] Jason Parker * /, main/translate.c: Merged revisions 93381 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r93381 | qwell | 2007-12-17 16:45:57 -0600 (Mon, 17 Dec 2007) | 4 lines What was I thinking when I wrote this masterpiece? -1 + 1 = 0.. who woulda thunk it?. ........ 2007-12-17 22:38 +0000 [r93380] Luigi Rizzo * channels/chan_oss.c: surprising as it may be, chan_oss compiles correctly under cygwin as well, provided you look for soundcard.h in the right place... 2007-12-17 22:29 +0000 [r93378] Joshua Colp * /, main/utils.c: Merged revisions 93377 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r93377 | file | 2007-12-17 18:28:09 -0400 (Mon, 17 Dec 2007) | 7 lines Do not try to access information about a lock when printing out a trylock attempt. It is possible for the lock that it references to no longer be valid. This would have caused segfaults or deadlocks. (issue #BE-263) (closes issue #11080) Reported by: callguy (closes issue #11100) Reported by: callguy ........ 2007-12-17 21:14 +0000 [r93337] Tilghman Lesher * /, include/asterisk/time.h: Merged revisions 93336 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r93336 | tilghman | 2007-12-17 15:12:42 -0600 (Mon, 17 Dec 2007) | 6 lines Today is tomorrow's yesterday, and yesterday's tomorrow is today, and tomorrow's tomorrow is the day after tomorrow, so who cares if you recycle anyway? If this confuses you, that's nothing compared to what this fixes. ;-) ........ 2007-12-17 21:12 +0000 [r93335] Olle Johansson * channels/chan_zap.c, /, channels/chan_sip.c, apps/app_queue.c, channels/chan_iax2.c, channels/chan_mgcp.c: Merged revisions 93182 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r93182 | oej | 2007-12-17 08:15:13 +0100 (MÃ¥n, 17 Dec 2007) | 8 lines Issue 11574: Add dependencies on res_monitor and res_features. I wonder if Asterisk can run at all without res_features. My guess is that there's propably a lot of more modules and the core that depends on it. Reported by: caio1982 (closes issue #11574) ........ 2007-12-17 20:42 +0000 [r93293-93297] Mark Michelson * apps/app_queue.c: Removing some leftover debug messages from a while back. * /, apps/app_voicemail.c: Merged revisions 93291 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r93291 | mmichelson | 2007-12-17 13:53:48 -0600 (Mon, 17 Dec 2007) | 6 lines We need to create the directory for a voicemail user even if they are using IMAP storage since greetings are stored in the filesystem. (closes issue #11388, reported by spditner, patch by me inspired by a patch by spditner) ........ 2007-12-17 18:07 +0000 [r93252] Joshua Colp * channels/chan_zap.c, /: Merged revisions 93250 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r93250 | file | 2007-12-17 14:05:55 -0400 (Mon, 17 Dec 2007) | 6 lines If a call is received with a called number IE containing nothing go to the 's' extension. (closes issue #9099) Reported by: kb1_kanobe2 Patches: 20070906__9099.diff.txt uploaded by Corydon76 (license 14) ........ 2007-12-17 17:16 +0000 [r93191-93224] Kevin P. Fleming * utils: all created files need to be listed in the ignore property * channels/chan_unistim.c, build_tools/menuselect-deps.in, configure, configure.ac, channels/Makefile, channels/chan_oss.c: make the configure script detect that it is running on a Windows platform, and report that information so that menuselect can use it (all information that is used to decide whether to build modules or not must be fed to menuselect so the user knows what will be built and why... don't make module build decisions in the makefiles, please) * Makefile: make using PRINT_DIR a little easier 2007-12-17 15:18 +0000 [r93187-93190] Joshua Colp * channels/chan_sip.c: Fix usage of rtptimeout. It can be used without rtpkeepalive, and the value can not be accessed directly in the SIP pvt structure. All RTP related timeouts have to be retrieved using the ast_rtp_* function calls. (closes issue #11562) Reported by: ibc * channels/chan_unistim.c: If no timezone is available use the default message. (closes issue #11576) Reported by: junky * channels/chan_unistim.c: Make chan_unistim actually be able to unload. When creating a thread that you want to pthread_join you have to explicitly create it as joinable, and also if using pthread_cancel you have to have a pthread_testcancel to see if it has been called. 2007-12-17 07:27 +0000 [r93184-93185] Kevin P. Fleming * codecs, /, build_tools/make_version, include/asterisk/autoconfig.h.in, configure.ac, apps, Makefile.moddir_rules, res/Makefile, pbx/Makefile, build_tools/prep_moduledeps (removed), channels/Makefile, cdr, formats, Makefile, codecs/Makefile, funcs, apps/Makefile, configure, build_tools/embed_modules.xml, cdr/Makefile, build_tools/prep_tarball, makeopts.in, formats/Makefile, res, pbx, channels, funcs/Makefile: Merged revisions 93180 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r93180 | kpfleming | 2007-12-16 22:44:51 -0800 (Sun, 16 Dec 2007) | 23 lines In http://lists.digium.com/pipermail/asterisk-dev/2007-December/031145.html, rizzo brought up some issues related to the way that the metadata required for menuselect and the rest of the build system is extracted from the source files. Since I had a few hours to kill on an airplane today, I decided to improve this situation... so now the system caches the extracted metadata and uses it to build the menuselect 'tree' as much as it can. The result of this is that when a single source file is changed, only the metadata for that file needs to be extracted again, and the rest is used from the cache files. I also reduced the number of forked processes required to do the metadata extraction; it was actually possible to do most of what we needed in the Makefiles themselves without using any shell scripts at all! On my laptop, these changes resulted in an 80% decrease in the time required for the 'menuselect.makeopts' automatic check to occur after editing a single source file. While doing this work I also cleaned up a few minor things in the Makefiles, adding a check for 'awk' to the configure script and changed all remaining places we use 'grep' or 'awk' to use the ones found by the configure script, and changed the 'prep_tarball' script to build the menuselect metadata so that tarballs of Asterisk will include it and won't require the user to wait while it is extracted after unpacking. ........ 2007-12-16 19:06 +0000 [r93173] Luigi Rizzo * Makefile: menuselect.makeopts is not a .PHONY target 2007-12-16 13:38 +0000 [r93163-93167] Olle Johansson * pbx/pbx_dundi.c: Convert from LOG_DEBUG etc to ast_debug. Thanks, dimas! (closes issue #11572) Reported by: dimas Patches: dundilog-trunk.patch uploaded by dimas (license 88) * main/manager.c, CHANGES: Adding a new CLI command for "manager reload", which is important now that you need to reload after changes. Thanks YS. Reported by: ys Patches: trunk93163_manager_reload.c.diff uploaded by ys (license 281) (related to issue #11414) * main/manager.c, CHANGES: Change manager so that registered accounts are stored in memory. This opens for a manager realtime implementation. If you change accounts in manager.conf, you now need to reload to activate the changes (deletions, additions). This was not the case with 1.4. Reported by: ys Patches: trunk93163_manager_reload.c.diff uploaded by ys (license 281) (closes issue #11414) * CHANGES: Adding console_video to CHANGES. It's important that we keep this file up to date, even with experimental stuff. * channels/chan_unistim.c, main/udptl.c, configs/dundi.conf.sample, channels/chan_sip.c, include/asterisk/rtp.h, include/asterisk/netsock.h, channels/iax2-provision.c, UPGRADE.txt, doc/tex/qos.tex, configs/skinny.conf.sample, CHANGES, channels/chan_iax2.c, main/rtp.c, main/netsock.c, configs/h323.conf.sample, configs/iax.conf.sample, channels/chan_skinny.c, configs/mgcp.conf.sample, configs/unistim.conf.sample, channels/chan_h323.c, configs/iaxprov.conf.sample, pbx/pbx_dundi.c, configs/sip.conf.sample, channels/chan_mgcp.c: HUGE improvements to QoS/CoS handling by IgorG - Refer to the proper documentation - Implement separate signalling/media QoS/CoS in many channels using RTP - Improve warnings and verbose messages - Deprecate some old settings Minor modifications by me, a big effort from IgorG. Thanks! Reported by: IgorG Patches: qoscleanup-89394-4-trunk.patch uploaded by IgorG (license 20) Tested by: IgorG (closes issue #11145) 2007-12-16 10:34 +0000 [r93162] Luigi Rizzo * Makefile: use a simpler idiom for 'cmp -s ...' 2007-12-16 09:37 +0000 [r93152-93161] Olle Johansson * main/asterisk.c: Don't drop the first character randomly in long listings in the CLI. Reported by: slavon Patches: asterisk-consolerefresh2.diff.txt uploaded by jamesgolovich (license 176) Tested by: eliel (closes issue #9325) * configs/sip.conf.sample, CHANGES: Update documentation * channels/chan_sip.c, configs/sip.conf.sample: Make more timers settable in SIP so that we can force timeout earlier on non-responsive SIP servers. Thanks, jcmoore, for the patch! Reported by: jcmoore Patches: peer_t1_timerb_trunk_v3.patch.txt uploaded by jcmoore (license 9) (closes issue #9771) * include/asterisk/file.h: Typo fixed earlier, that wasn't a typo after all. Didn't a clever guy once say "Compile before you commit" ? :-) 2007-12-15 08:10 +0000 [r93151] Russell Bryant * include/asterisk/file.h: fix a typo from revision 93138 2007-12-15 00:44 +0000 [r93138-93145] Luigi Rizzo * configs/oss.conf.sample: configuration options related to video support. * channels/console_video.c (added): Bring in video console support for chan_oss (and later chan_alsa too). This is disabled in the default build, you need to explicitly enable it compiling with make COPTS=-DHAVE_VIDEO_CONSOLE In return, you will be able to do a video call with chan_oss, using the webcam (or X11 grabbing) as local source, and rendering the incoming stream on your screen. Currently supported formats are h261, h263, h263+, h264, mpeg4 (all through the avcodec lib, part of ffmpeg). Incoming video is on the left, outgoing video is on the right, while the center displays a keypad (if configured so). Right clicking on the video windows increases the size, center clicking reduces the size. Dragging the mouse (with the left key) on the right window while the X11 grabber is active moves the grab area. This is the result of work by Sergio Fadda, Marta Carbone and myself, all properly disclaimed to digium. Note, there is a lot of work left to do in this module, including adding support for Video4LinuxV2 (I have patches from Matteo Brancaleoni which should be integrated), and making the GUI a lot more friendly than it is now (e.g. supporting merging or switching among multiple sources, a text window, and more). * channels/chan_oss.c: remove some redundant headers * include/asterisk/file.h: include mmap header if detected by configure 2007-12-14 22:02 +0000 [r93094-93115] Mark Michelson * apps/app_voicemail.c: Resolve a compiler warning * apps/app_voicemail.c: Change places where the name "INBOX" was hardcoded to use the imapfolder setting from voicemail.conf instead. This commit will help to get issue #11415 moving towards commitment. 2007-12-14 21:09 +0000 [r93090] Tilghman Lesher * Makefile, channels/chan_unistim.c, codecs/ilbc/iLBC_define.h: Solaris compat fixes Reported by: snuffy Patch by: snuffy,tilghman (Closes issue #11315) 2007-12-14 19:31 +0000 [r93067] Russell Bryant * pbx/pbx_dundi.c: make something static 2007-12-14 19:27 +0000 [r93066] Tilghman Lesher * apps/app_privacy.c, UPGRADE.txt, CHANGES, configs/privacy.conf.sample (removed): Remove use of privacy.conf by the Privacy app. Reported by: eliel Patch by: eliel (Closes issue #11344) 2007-12-14 19:19 +0000 [r93042-93065] Mark Michelson * main/pbx.c, main/manager.c, funcs/func_timeout.c: I needed to increment the numbers used on the VERBOSITY_ATLEAST calls by 1. Thanks to kpfleming for pointing this out. * include/asterisk/logger.h, main/pbx.c, main/manager.c, funcs/func_timeout.c: Changed VERBOSITY_LEVEL to VERBOSITY_ATLEAST to be more accurate. * include/asterisk/logger.h, main/pbx.c, main/manager.c, funcs/func_timeout.c, main/logger.c: After reading Russell's e-mail to the dev list stating that checking option_verbose is not equivalent to the check done by ast_verb, I wrote a macro, VERBOSITY_LEVEL, which does this check. I did a quick look in the source and used this macro in some places where option_verbose was used. I also converted some verbose messages in logger.c to use ast_verb instead of ast_verbose. 2007-12-14 18:24 +0000 [r93041] Tilghman Lesher * apps/app_meetme.c: gcc 4.1.3 wants a union used here. 2007-12-14 17:49 +0000 [r93001-93004] Russell Bryant * main/config.c: Print an error message if a #included file does not exist 2007-12-14 17:29 +0000 [r92999] Tilghman Lesher * res/res_agi.c: Publish the AGI events to manager. Reported by: moy Patch by: moy,tilghman (Closes issue #11337) 2007-12-14 15:59 +0000 [r92976] Mark Michelson * funcs/func_timeout.c: Reintroduce an optimization that was lost when converting trunk to use ast_verb. 2007-12-14 15:49 +0000 [r92939] Tilghman Lesher * main/editline/sys.h: If malloc.h is included in a Solaris build, the compilation breaks. Reported by: snuffy Patch by: snuffy (Closes issue #11313) 2007-12-14 15:18 +0000 [r92938] Joshua Colp * /, channels/chan_sip.c: Merged revisions 92937 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92937 | file | 2007-12-14 11:16:15 -0400 (Fri, 14 Dec 2007) | 4 lines Up the length of the format on the SIP channel since it can now be rather long. (closes issue #11552) Reported by: francesco_r ........ 2007-12-14 15:14 +0000 [r92936] Tilghman Lesher * /, res/res_agi.c: Merged revisions 92933 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92933 | tilghman | 2007-12-14 09:01:10 -0600 (Fri, 14 Dec 2007) | 5 lines Change help documentation to match actual behavior (FAILURE vs FAILED). Reported by: angeloxx-sir Patch by: tilghman (Closes issue #11548) ........ 2007-12-14 15:08 +0000 [r92935] Christian Richter * channels/chan_misdn.c, /: Merged revisions 92934 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92934 | crichter | 2007-12-14 16:05:28 +0100 (Fr, 14 Dez 2007) | 1 line fixed the sequencing of WAITING_4DIGS state setting and overlap_task thread starting. ........ 2007-12-14 14:48 +0000 [r92913] Tilghman Lesher * apps/app_dial.c, main/pbx.c, main/srv.c, channels/chan_skinny.c, res/res_features.c, apps/app_minivm.c, apps/app_amd.c, res/snmp/agent.c, apps/app_chanspy.c, apps/app_mixmonitor.c, main/asterisk.c, main/netsock.c, apps/app_voicemail.c: Convert ast_verbose to ast_verb. Reported by: snuffy Patch by: snuffy (Closes issue #11547) 2007-12-14 01:25 +0000 [r92876] Mark Michelson * /, include/asterisk/lock.h: Merged revisions 92875 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92875 | mmichelson | 2007-12-13 19:24:06 -0600 (Thu, 13 Dec 2007) | 7 lines When compiling with DETECT_DEADLOCKS, don't spam the CLI with messages about possible deadlocks. Instead just print the intended single message every five seconds. (closes issue 11537, reported and patched by dimas) ........ 2007-12-13 23:10 +0000 [r92816-92855] Tilghman Lesher * apps/app_meetme.c: When working with dates, use numeric form whenever possible, as it's faster. Also, a bunch of coding guidelines fixes. * channels/chan_zap.c, /: Merged revisions 92815 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92815 | tilghman | 2007-12-13 15:28:39 -0600 (Thu, 13 Dec 2007) | 5 lines Properly initialize polarity statuses, so that they are detected properly. Reported by: julianjm Patch by: julianjm (Closes issue #10238) ........ 2007-12-13 20:23 +0000 [r92811] Joshua Colp * include/asterisk/app.h, include/asterisk/module.h, res/res_agi.c, apps/app_rpt.c: Move usage of the old LOCAL_USER_* macros to the new ast_module_user_* functions in a few documentation places. (closes issue #11533) Reported by: IgorG Patches: oldmacroclean.v1.diff uploaded by IgorG (license 20) 2007-12-13 20:14 +0000 [r92810] Jason Parker * main/pbx.c, /: Merged revisions 92809 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92809 | qwell | 2007-12-13 14:13:48 -0600 (Thu, 13 Dec 2007) | 1 line Make application help text a little more clear about the use of extensions in a filename. ........ 2007-12-13 20:12 +0000 [r92806-92808] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 92807 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92807 | mmichelson | 2007-12-13 14:03:20 -0600 (Thu, 13 Dec 2007) | 3 lines Prevent another potential fd leak ........ * /, apps/app_voicemail.c: Merged revisions 92803 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92803 | mmichelson | 2007-12-13 13:49:55 -0600 (Thu, 13 Dec 2007) | 3 lines Prevent a possible fd leak. ........ 2007-12-13 17:46 +0000 [r92779] Tilghman Lesher * cdr/cdr_adaptive_odbc.c: Don't use backslash as an escape character, unless it really is an escape character. 2007-12-13 16:23 +0000 [r92758] Jason Parker * channels/chan_sip.c: Remove remnants of a poorly merged commit. (92697) 2007-12-13 15:40 +0000 [r92737] Doug Bailey * apps/app_voicemail.c: Tag voicemails with UTC time as opposed to local time zone 2007-12-13 00:18 +0000 [r92697] Jason Parker * /, channels/chan_sip.c, channels/chan_h323.c, main/config.c: Merged revisions 92696 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #10690) ........ r92696 | qwell | 2007-12-12 18:11:09 -0600 (Wed, 12 Dec 2007) | 7 lines If a typo is found in a config file, we previous continued on with what was already loaded. We do not want to do this (see bug below for details). This makes it so that if a [ is found without a ], the entire config will fail, and nothing in it will be loaded. Issue 10690. ........ 2007-12-12 23:44 +0000 [r92676] Russell Bryant * channels/chan_iax2.c: Revert an "optimization" that I added in revision 89887, as the user who reported issue #11449 has demonstrated that it actually was a performance hit on his machine. I think that it is possible that it could still be a benefit on systems under higher load, especially SMP systems, but I don't have enough time or interest to find out at the moment. (closes issue #11449) 2007-12-12 21:22 +0000 [r92618] Jason Parker * /, apps/app_meetme.c, channels/ringtone.h: Merged revisions 92617 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11048) ........ r92617 | qwell | 2007-12-12 15:15:45 -0600 (Wed, 12 Dec 2007) | 4 lines Don't increment user count until after name has been recorded (if enabled). Issue 11048, tested by pep. ........ 2007-12-12 20:05 +0000 [r92594] Tilghman Lesher * apps/app_dial.c, main/logger.c, main/utils.c, apps/app_mixmonitor.c: Conversions of free to ast_free, where applicable, and several other formatting fixes. Reported by: eliel Patch by: eliel,tilghman (Closes issue #11209) 2007-12-12 19:50 +0000 [r92562] Russell Bryant * res/res_features.c: Merged revisions 92556 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92556 | russell | 2007-12-12 13:40:02 -0600 (Wed, 12 Dec 2007) | 1 line resolve compiler warning ........ 2007-12-12 17:51 +0000 [r92511-92526] Mark Michelson * res/res_features.c: Same change to trunk as revision 92510. I'm not sure why I merged this way, but I did. 2007-12-12 17:15 +0000 [r92476-92507] Tilghman Lesher * main/asterisk.c: Correctly handle possible memory allocation failure Reported by: eliel Patch by: eliel (Closes issue #11512) * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 92463 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92463 | tilghman | 2007-12-12 10:52:56 -0600 (Wed, 12 Dec 2007) | 4 lines Test directly for the API that fixed AST-2007-026, to ensure that older versions of PostgreSQL are no longer acceptable. (Closes issue #11526) ........ 2007-12-12 16:11 +0000 [r92444] Mark Michelson * /, apps/app_queue.c: Merged revisions 92443 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92443 | mmichelson | 2007-12-12 10:08:55 -0600 (Wed, 12 Dec 2007) | 3 lines Removing an unused variable. ........ 2007-12-11 22:20 +0000 [r92423] Olle Johansson * include/asterisk/term.h, channels/misdn/isdn_msg_parser.c, channels/ringtone.h, include/asterisk/ulaw.h, include/jitterbuf.h, include/asterisk/manager.h, include/asterisk/transcap.h, channels/misdn/isdn_lib.c, channels/gentone.c, include/asterisk/zapata.h, channels/misdn/isdn_lib.h, include/asterisk/doxyref.h, channels/DialTone.h, channels/misdn/ie.c, channels/misdn/chan_misdn_config.h, channels/iax2.h, channels/misdn/portinfo.c, include/asterisk/udptl.h, main/cygload.c, include/asterisk/translate.h: Doxygen updates, formatting. misdn stuff needs a lot of doxygenification (Hello, Qwell :-) ) 2007-12-11 22:10 +0000 [r92422] Mark Michelson * channels/chan_zap.c, configure, include/asterisk/autoconfig.h.in, configure.ac: Trunk build would fail due to the nonexistence of zaptel hwgain structures missing. Patched configure to check for this stuff and put a #ifdef around the offending code in chan_zap. Thanks to file for overseeing this. 2007-12-11 21:58 +0000 [r92421] Jason Parker * channels/chan_sip.c: We need to set the address we want to match against before we actually do the match.. Closes issue #11518. 2007-12-11 21:46 +0000 [r92402] Mark Michelson * res/res_musiconhold.c: Removing a pointless memset. The memory was just calloc'd, so the memory is already zeroed out 2007-12-11 21:17 +0000 [r92401] Jason Parker * apps/app_controlplayback.c: Add variable to show which key was pressed to stop playback. Issue #11377, initial patch by johan. 2007-12-11 20:06 +0000 [r92364-92365] Joshua Colp * res/res_monitor.c: Only look to see if options are set if some have been provided. (closes issue #11505) Reported by: Mike Anikienko * main/global_datastores.c, /: Merged revisions 92363 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92363 | file | 2007-12-11 15:51:40 -0400 (Tue, 11 Dec 2007) | 6 lines Fix potential memory leak with the dialed interfaces list if another memory allocation fails. (closes issue #11507) Reported by: eliel Patches: global_datastores.c.patch uploaded by eliel (license 64) ........ 2007-12-11 17:44 +0000 [r92324] Mark Michelson * /, apps/app_queue.c: Merged revisions 92323 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92323 | mmichelson | 2007-12-11 11:42:25 -0600 (Tue, 11 Dec 2007) | 10 lines Fixing autofill to be more accurate. Specifically, if calls ahead of the current caller were ringing members (but not yet bridged) there could be available members and waiting callers who would not get matched up. The member availability checker was correctly determining the number of available members in this scenario, but the queue itself did not parallelly reflect this status on the pending calls. This commit corrects the issue. (closes issue #11459, reported by equissoftware, patched by me) ........ 2007-12-11 16:29 +0000 [r92305] Russell Bryant * include/asterisk/unaligned.h, main/event.c: * In unaligned.h, remove some unnecessary casts and mark the arg of the get_unaligned functions as const * In event.c, use get_unaligned_uint32() in a couple of places to fix issues on architectures that don't allow unaligned access 2007-12-11 14:17 +0000 [r92267-92285] Olle Johansson * include/asterisk/devicestate.h, include/asterisk/agi.h, include/asterisk/astobj2.h, include/asterisk/extconf.h, include/asterisk/io.h, include/asterisk/cdr.h, include/asterisk/aes.h, include/asterisk/_private.h, include/asterisk/localtime.h, include/asterisk/hashtab.h, include/asterisk/callerid.h, include/asterisk/logger.h, include/asterisk/doxyref.h, include/asterisk/app.h, include/asterisk/adsi.h, include/asterisk/event.h, include/asterisk/causes.h, include/asterisk/alaw.h, include/asterisk/ast_expr.h, include/asterisk/dsp.h, include/asterisk/mod_format.h, include/asterisk/ael_structs.h, include/asterisk/astdb.h: A lot of doxygen updates * include/asterisk/frame.h: Doxygen updates 2007-12-10 20:18 +0000 [r92243] Doug Bailey * channels/chan_zap.c: Add CLI commands to dynamically set hw and sw gains 2007-12-10 16:48 +0000 [r92205-92206] Joshua Colp * utils/check_expr.c: Add ast_atomic_fetchadd_int_slow to check_expr for platforms that need it. (closes issue #11484) Reported by: snuffy * /, main/rtp.c: Merged revisions 92204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92204 | file | 2007-12-10 12:36:15 -0400 (Mon, 10 Dec 2007) | 6 lines Add G729A as another possible payload name for G729. Some devices use this instead of G729, which is perfectly normal since the payload number itself is defined and can't be used by anything else so the name doesn't matter that much. (closes issue #11483) Reported by: revolution Patches: rtp.diff uploaded by revolution (license 346) ........ 2007-12-10 16:30 +0000 [r92203] Mark Michelson * /, apps/app_queue.c: Merged revisions 92202 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92202 | mmichelson | 2007-12-10 10:29:44 -0600 (Mon, 10 Dec 2007) | 7 lines If there are no members in a queue, then the loop where the datastore for detecting duplicate dialed numbers will be skipped, meaning the datastore isn't created. This means that when we try to free it, there's a crash. This stops that crash from occurring. (closes issue #11499, reported by slavon, patched by eliel) ........ 2007-12-10 16:15 +0000 [r92199-92201] Joshua Colp * res/res_agi.c: Only send a SIGHUP if the pid is greater than -1, otherwise all PIDs greater than -1 will get the SIGHUP... and that is bad. (closes issue #11453) Reported by: alanmcmillan 2007-12-10 14:18 +0000 [r92140-92160] Olle Johansson * channels/chan_sip.c: Removing some LOG_DEBUG items * /, channels/chan_sip.c: Merged revisions 92158 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92158 | oej | 2007-12-10 15:04:44 +0100 (MÃ¥n, 10 Dec 2007) | 16 lines Avoid reinvite race situations with two Asterisks trying to reinvite each other in 1.4 and trunk. This patch implements support for the 491 error code that Asterisk 1.4 generates on situations where we get an incoming INVITE and already has one in progress. Thanks to mavetju for reporting and to Raj Jain for an excellent explanation of the problem. Patch by myself. Tested with 8 Asterisk servers connected to each other in a training network. Closes issue #10481 ........ * doc/manager_1_1.txt, apps/app_voicemail.c: Add a few extra headers in the voicemail users listing in manager 1.1. Update documentation too. (closes issue #11495) Reported by: caio1982 Patches: extra_vm_manager_info1.diff uploaded by caio1982 (license 22) 2007-12-10 09:00 +0000 [r91929-92122] Luigi Rizzo * build_tools/make_version, build_tools/make_version_h: simplify/cleanup the scripts * utils/Makefile: remove relative paths and use ASTTOPDIR instead. Give a default value to ASTTOPDIR if unset so we can at least do a 'make clean' without too much trouble. The proper fix, however, is to partition the top level Makefile in a 'setup' and a 'main' part, in a way that the 'setup' part can be included from subdirs' Makefiles and allow targets to be built without going through the top level Makefile. * utils/clicompat.c: simplify this file * doc/CODING-GUIDELINES: add a bit of info on the build infrastructure * Makefile: Fix the detection of modules installed from this build. You can now add the path of local module subdirs from the command line with make LOCAL_MOD_SUBDIRS= .... * codecs/Makefile, apps/Makefile, Makefile.moddir_rules, cdr/Makefile, pbx/Makefile, res/Makefile, channels/Makefile, formats/Makefile, funcs/Makefile: Put into Makefile.moddir_rules the common instructions used to generate loadable and embedded module lists. Individual Makefiles now are a lot simpler, possibly as simple as this: -include $(ASTTOPDIR)/menuselect.makeopts $(ASTTOPDIR)/menuselect.makedeps MODULE_PREFIX=cdr_ all: _all include $(ASTTOPDIR)/Makefile.moddir_rules and also more flexible because in a single directory we can combine various types of modules (app_, cdr_, func_, ... ) by simply listing them in the MODULE_PREFIX variable. The individual Makefiles can also create list of modules to be excluded by listing them in the variablel MODULE_EXCLUDE (see an example in channels/Makefile). With this change it becomes trivial to integrate a directory with locally created/modified sources into the main build. * Makefile, Makefile.moddir_rules: make the install target a bit less noisy * Makefile: document usage of several exported variables * utils/Makefile: add hashtab.c to the list of files deleted * Makefile.moddir_rules: another place where ../ should have been ASTTOPDIR * codecs/Makefile, utils/Makefile, apps/Makefile, cdr/Makefile, pbx/Makefile, res/Makefile, channels/Makefile, formats/Makefile, funcs/Makefile: normalize subdirs' Makefile by using ASTTOPDIR and not .. to reference the top level directory. * Makefile: Implement the outcome of a discussion on the -dev list re. the use of DESTDIR and INSTALL_PATH - many thanks to Tzafrir Cohen and Simon Perreault for extremely useful feedback: 1. comment out the [directories] section the created asterisk.conf ; you can set the correct defaults at build time using INSTALL_PATH, so the repetition here is redundant and often wrong. (The next step now is move asterisk.conf outside the Makefile to asterisk.conf.sample, because there is little if anything here that needs to be constructed at build/install time). 2. use DESTDIR?=$(INSTALL_PATH) so you only need to specify a path once if the two coincide. This should have no ill side effects, because if you don't specify DESTDIR, you really need INSTALL_PATH="" to set the correct defaults, and if you specify DESTDIR the value is not overridden. The second part required moving the 'export DESTDIR' right after the assignment to prevent DESTDIR getting set by the export (this is documented in the Makefile).o hopefully avoid the mistake)$ With this change you can now do something like this from your source tree: make INSTALL_PATH=/some/place install samples and then main/asterisk -vdc which will pick up the correct config files and libraries from /some/place - i.e. great for developers! * main/config.c: remove unused code, and simplify the logic for #include/#exec (still a lot of cleanup needed here). * main/config.c: Implement comment_buffer and lline_buffer in terms of the ast_str_*() API. I don't know if comment_buffers etc are actually used at all... * main/config.c: unify some common code * main/config.c: normalize formatting * main/config.c: document a nice technique to exit from a block in case of errors. * main/config.c: a little bit of documentation on how lines are parsed. * utils/ael_main.c: normalize header order, and add a comment on the need to clean up this file. * include/asterisk/network.h: some platforms (e.g. FreeBSD4) need netinet/in.h to be included before arpa/inet.h 2007-12-07 23:32 +0000 [r91832-91891] Jason Parker * /, main/dsp.c: Merged revisions 91890 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11273) ........ r91890 | qwell | 2007-12-07 17:29:01 -0600 (Fri, 07 Dec 2007) | 4 lines We need to make sure we free the input frame if we return a different frame in ast_dsp_process. Issue 11273, pointed out by dimas, with a patch by eliel. ........ * pbx/pbx_lua.c, configs/extensions.lua.sample: Update documentation for pbx_lua. Closes issue #11492, patch by mnicholson. 2007-12-07 21:25 +0000 [r91784-91831] Russell Bryant * /, main/utils.c: Merged revisions 91830 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91830 | russell | 2007-12-07 15:24:33 -0600 (Fri, 07 Dec 2007) | 5 lines Make the lock protecting each thread's list of locks it currently holds recursive. I think that this will fix the situation where some people have said that "core show locks" locks up the CLI. (related to issue #11080) ........ * /, include/asterisk/lock.h: Merged revisions 91828 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91828 | russell | 2007-12-07 15:17:24 -0600 (Fri, 07 Dec 2007) | 6 lines Fix another bug in the DEBUG_THREADS code. The ast_mutex_init() function had the mutex attribute object marked as static. This means that multiple threads initializing locks at the same time could step on each other and end up with improperly initialized locks. (found when tracking down locking issues related to issue #11080) ........ * /, include/asterisk/lock.h: Merged revisions 91826 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91826 | russell | 2007-12-07 15:11:08 -0600 (Fri, 07 Dec 2007) | 6 lines I love fixing lock related errors in the lock debugging code. That's about as ironic as it gets in Asterisk programming land. Anyway, I spotted this bug while trying to track down why systems are locking up and acting weird in issue #11080. The mutex attribute object was marked as static in this function when it should not have been. ........ * apps/app_dial.c, /: Merged revisions 91783 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91783 | russell | 2007-12-07 10:38:48 -0600 (Fri, 07 Dec 2007) | 6 lines * Add channel locking around datastore operations that expect the channel to be locked. * Document why we don't record Local channels in the dialed interfaces list. * Remove the dialed variable as it isn't needed. * Restructure some code for clarity and coding guidelines stuff ........ 2007-12-07 16:37 +0000 [r91782] Jason Parker * channels/chan_sip.c: Fix a small typo in a comment. Closes issue #11490 2007-12-07 16:28 +0000 [r91781] Russell Bryant * /, apps/app_queue.c: Merged revisions 91780 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91780 | russell | 2007-12-07 10:25:25 -0600 (Fri, 07 Dec 2007) | 7 lines * Add channel locking around datastore operations that expect the channel to be locked. * Document why we don't record Local channels in the dialed interfaces list. * Handle memory allocation failure. * Remove the dialed variable, as it wasn't actually needed. * Tweak some formatting to conform to coding guidelines. ........ 2007-12-07 16:11 +0000 [r91779] Jason Parker * doc/asterisk-mib.txt, main/pbx.c, res/snmp/agent.c, include/asterisk/pbx.h, main/cli.c: Add count of total number of calls processed by asterisk during it's lifetime. Add number of total calls and current calls to SNMP. Closes issue #10057, patch by jcmoore. 2007-12-07 16:11 +0000 [r91778] Russell Bryant * main/autoservice.c, /: Merged revisions 91777 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91777 | russell | 2007-12-07 10:08:35 -0600 (Fri, 07 Dec 2007) | 6 lines * Add a bit more of a verbose comment as to why a hangup frame needs to be queued up if autoservice gets a NULL return from ast_read(). * Make the process of queueing the hangup frame more efficient by putting the frame where it is going to end up and avoiding some locking and extra memory allocations and freeing. ........ 2007-12-07 15:40 +0000 [r91738] Mark Michelson * main/autoservice.c, /: Merged revisions 91737 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91737 | mmichelson | 2007-12-07 09:39:58 -0600 (Fri, 07 Dec 2007) | 7 lines Hangups that happen during autoservice were not processed appropriately. This is because a hangup actually causes a NULL frame to be received, not a hangup frame. Queueing a hangup if we receive a NULL frame during autoservice corrects this problem (closes issue #11467, reported by jmls, patched by me) ........ 2007-12-07 02:52 +0000 [r91676-91700] Russell Bryant * apps/app_dial.c, /: Merged revisions 91693 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91693 | russell | 2007-12-06 20:51:22 -0600 (Thu, 06 Dec 2007) | 2 lines Don't unlock the dialed_interfaces list until we're done messing with the iterator. ........ * apps/app_dial.c, /, apps/app_queue.c: Merged revisions 91677 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91677 | russell | 2007-12-06 20:38:40 -0600 (Thu, 06 Dec 2007) | 4 lines Allow dialing local channels from Queue() and Dial() again. There was a slight flaw in the code to prevent call forwards from looping that caused this problem. (related to issue #11486) ........ * /, apps/app_queue.c: Merged revisions 91675 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91675 | russell | 2007-12-06 20:19:45 -0600 (Thu, 06 Dec 2007) | 7 lines Fix in an issue in the call forwarding handling code that was causing crashes on every call into a queue. I'm not entirely sure about the logic in this part of the code, so I want to look at it some more tomorrow. However, this makes it safe and keeps it from crashing. (closes issue #11486, reported by adamg, patched by me) ........ 2007-12-07 00:58 +0000 [r91617-91638] Tilghman Lesher * /, main/rtp.c: Merged revisions 91637 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91637 | tilghman | 2007-12-06 18:52:17 -0600 (Thu, 06 Dec 2007) | 5 lines At the end of a call, when we're reporting, RTCP may already be partially torn down, so check for NULL dereference Reported by: blitzrage Patch by: tilghman (Closes issue #11450) ........ * channels/chan_zap.c: Add a manager event for PRI events: this will help manager users detect when a D-channel goes down * main/cdr.c: If duration or billsec are not yet calculated, calculate them on demand. 2007-12-06 21:57 +0000 [r91598] Jason Parker * cdr/cdr_sqlite3_custom.c: Fix a problem with quoting in sqlite3 cdr module.. Closes issue #11070, patch by seanbright. 2007-12-06 21:03 +0000 [r91579] Mark Michelson * apps/app_voicemail.c: Handle allocation failure of the heard and deleted arrays of the vm_state. (closes issue #11408, reported and patched by jaroth) 2007-12-06 20:52 +0000 [r91561] Tilghman Lesher * /, cdr/cdr_pgsql.c: Merged revisions 90166,90736,90753 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90166 | tilghman | 2007-11-29 13:48:10 -0600 (Thu, 29 Nov 2007) | 3 lines Properly escape cdr->src and cdr->dst and ensure we use thread-safe escaping (Fixes AST-2007-026) ........ r90736 | tilghman | 2007-12-03 17:23:55 -0600 (Mon, 03 Dec 2007) | 5 lines If both dbhost and dbsock were not set, a NULL deref could result Reported by: xrg Patch by: tilghman (Closes issue #11387) ........ r90753 | tilghman | 2007-12-03 17:50:51 -0600 (Mon, 03 Dec 2007) | 5 lines Solaris requires the inclusion of sys/loadavg.h for getloadavg(). Reported by: snuffy Patch by: snuffy,tilghman (Closes issue #11430) ........ 2007-12-06 16:54 +0000 [r91472] Matthew Fredrickson * channels/chan_zap.c: Make sure we clear these flags when libpri is not installed 2007-12-06 16:51 +0000 [r91440-91458] Joshua Colp * main/udptl.c, /: Merged revisions 91450 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91450 | file | 2007-12-06 12:49:42 -0400 (Thu, 06 Dec 2007) | 6 lines Fix various in the udptl implementation. It could return empty modem frames, have an incorrect sequence number on packets, and display the wrong sequence number in the debug messages. (closes issue #11228) Reported by: Cache Patches: udptl-4.patch uploaded by dimas (license 88) ........ * /, channels/chan_sip.c: Merged revisions 91439 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91439 | file | 2007-12-06 12:14:26 -0400 (Thu, 06 Dec 2007) | 4 lines Add support for accepting and sending T.38 in the initial INVITE. (closes issue #9402) Reported by: thdei ........ 2007-12-06 15:56 +0000 [r91347-91438] Olle Johansson * doc/manager_1_1.txt (added), UPGRADE.txt: Adding documentation for the massive manager changes to manager version 1.1 - hopefully a more consistent manager interface. * main/manager.c: - The Ping Action - Now use Response: success - New header "Ping: pong" :-) - The Events action - Now use Response: Success - The new status is reported as "Events: On" or "Events: Off" - Report if manager is enabled in the reload event Small cleanups... From moremanager * main/channel.c: Changes to manager events in channel.c - Newstate event - Now has "CalleridNum" for numeric caller id, like Newchannel - The event does not send "" for unknown caller IDs just an empty field - Newstate and Newchannel events - these have changed headers "State" -> ChannelStateDesc Text based channel state -> ChannelState Numeric channel state - The events does not send "" for unknown caller IDs just an empty field - Newstate event - Now has "CalleridNum" for numeric caller id, like Newchannel - The event does not send "" for unknown caller IDs just an empty field - Link and Unlink events - The "Link" and "Unlink" bridge events in channel.c are now renamed to "Bridge" - The link state is in the bridgestate: header as "Link" or "Unlink" - For channel.c bridges, "Bridgetype: core" is added. This opens up for bridge events in rtp.c and channel drivers - The "Rename" manager event has a renamed header, to use the same terminology for the current channel as other events - Oldname -> Channel (Moremanager) * main/cdr.c: New manager event when a channel changes account code. Maybe belongs in the new cdr category? ---moremanager--- Event: NewAccountCode Modules: cdr.c Purpose: To report a change in account code for a live channel Example: Event: NewAccountCode Privilege: call,all Channel: SIP/olle-01844600 Uniqueid: 1177530895.2 AccountCode: Stinas account 1234848484 OldAccountCode: Olles Account 12345 * apps/app_dial.c: - Dial event - Event Dial has new headers, to comply with other events - Source -> Channel Channel name (caller) - SrcUniqueID -> UniqueID Uniqueid (new) -> Dialstring Dialstring in app data (moremanager) * apps/app_meetme.c: Adding small missing but important comma... * apps/app_meetme.c: A big oops... * apps/app_meetme.c: The MeetmeJoin now has caller ID name and Caller ID number fields (like MeetMeLeave) (Moremanager) * channels/chan_zap.c: Update ZapShowChannels so that you can specify one channel. Action ZapShowChannels Header changes - Channel: -> ZapChannel For active channels, the Channel: and Uniqueid: headers are added You can now add a "ZapChannel: " argument to zapshowchannels actions to only get information about one channel. From the moremanager branch * main/logger.c: Doxygen updates * include/asterisk/logger.h, /, main/logger.c, main/loader.c: Merged revisions 91366 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91366 | oej | 2007-12-06 13:54:11 +0100 (Tor, 06 Dec 2007) | 4 lines Make sure logger is reloaded at general reload in the cli. (Discovered during Asterisk training in Portugal) ........ * main/manager.c: Change description of new manager command * main/manager.c, CHANGES: Add manager command for showing all current channels. Thanks, eliel, for writing the original patch. Modified by me to follow other manager events and the new "moremanager" style. (closes issue #11478) Reported by: eliel Patches: manager.c.patch uploaded by eliel (license 64) 2007-12-06 04:37 +0000 [r91328] Joshua Colp * main/channel.c: Instead of iterating through the entire epoll events array just look at the ones that will actually contain data. (props to eliel on IRC for this) 2007-12-05 22:57 +0000 [r91291-91293] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 91292 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91292 | mmichelson | 2007-12-05 16:57:13 -0600 (Wed, 05 Dec 2007) | 3 lines Reverting extra stuff I didn't mean to commit ........ * apps/app_dial.c, /, apps/app_voicemail.c: Merged revisions 91273 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91273 | mmichelson | 2007-12-05 16:35:52 -0600 (Wed, 05 Dec 2007) | 11 lines The 'G' option for Dial() did not properly handle the case where only a label was provided. This was due to the fact that the answering channel did not have an extension set, so ast_parseable_goto would fail. This fix eliminates the call to ast_parseable_goto on the answering channel since it is a wasteful call. The answering channel and the calling channel are both directed to the same extension and context, just different priorities, so we can just copy the values from the calling channel to the answering channel and increment the answering channel's priority. (closes issue #11382, reported by jon, patch by me with correction by jon) ........ 2007-12-05 21:46 +0000 [r91238] Tilghman Lesher * /, sounds/Makefile: Merged revisions 91237 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91237 | tilghman | 2007-12-05 15:38:13 -0600 (Wed, 05 Dec 2007) | 2 lines Upgrade to the latest version of extra sounds ........ 2007-12-05 17:49 +0000 [r91193-91197] Russell Bryant * /, main/threadstorage.c: Merged revisions 91192 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91192 | russell | 2007-12-05 11:31:42 -0600 (Wed, 05 Dec 2007) | 10 lines Make the lock in the threadstorage debugging code untracked to avoid a deadlock on thread destruction. (closes issue #11207) Reported by: ys Patches: threadstorage.c.diff uploaded by ys (license 281) Also fixes an open bug report: (closes issue #11446) ........ * apps/app_directory.c: Resolve compiler warnings. 2007-12-05 16:46 +0000 [r91172-91173] Tilghman Lesher * main/manager.c, UPGRADE.txt, configs/manager.conf.sample, CHANGES, include/asterisk/manager.h, cdr/cdr_manager.c: Change cdr_manager to use a "CDR" level, rather than the (overcrowded) "call" level. (Closes issue #11015) * CHANGES, apps/app_directory.c: Added multiple name listing. (Closes issue #10413) 2007-12-05 16:14 +0000 [r91171] Joshua Colp * configs/http.conf.sample: Remove second prefix line. Only need it documented once in the same file. (closes issue #11472) Reported by: eserra Patches: http.conf.sample.diff uploaded by eserra (license 45) 2007-12-05 13:09 +0000 [r91151-91152] Olle Johansson * channels/chan_sip.c, UPGRADE.txt, configs/sip.conf.sample: Rename "username" to "defaultuser" to match with "defaultip". "Username" still works, but is deprecated. * channels/chan_sip.c: Remove the cseqs from "sip show channel" and make more place for the call ID. 2007-12-05 03:48 +0000 [r91133] Kevin P. Fleming * channels/chan_zap.c: revert part of my changes from earlier today since this code is no longer dependent on libpri.h 2007-12-05 03:34 +0000 [r91029-91131] Russell Bryant * res/res_odbc.c: Use ast_free() instead of free(). (closes issue #11309) Reported by: Laureano Patches: res_odbc.c.patch uploaded by Laureano (license 265) * /, include/asterisk/lock.h: Merged revisions 91070 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91070 | russell | 2007-12-04 18:35:31 -0600 (Tue, 04 Dec 2007) | 11 lines Fix some crashes in chan_iax2 that were reported as happening on Mac systems. It turns out that the problem was the Mac version of the ast_atomic_fetchadd_int() function. The Mac atomic add function returns the _new_ value, while this function is supposed to return the old value. So, the crashes happened on unreferencing objects. If the reference count was decreased to 1, ao2_ref() thought that it had been decreased to zero, and called the destructor. However, there was still an outstanding reference around. (closes issue #11176) (closes issue #11289) ........ * /, main/utils.c: Merged revisions 91074 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91074 | russell | 2007-12-04 18:48:47 -0600 (Tue, 04 Dec 2007) | 4 lines When DEBUG_THREADS is enabled, we only have the details about who is holding a lock that we are waiting on for a mutex, not rwlocks. This should fix the problem where people have reported "core show locks" crashing sometimes. ........ * channels/chan_zap.c: Fix mwimonitornotify on reload ... again. This option was only read at startup so a reload would erase it and not reset it. (pointed out by tzafrir) * utils/astman.c: Fix the build of astman. Any file that includes any asterisk sub-headers needs to first include asterisk.h. (closes issue #11394) 2007-12-04 22:44 +0000 [r91012] Matthew Fredrickson * channels/chan_zap.c: Don't error when we don't have libpri installed with libss7 support. Also, print the debug message anyway if we can't find the right PRI 2007-12-04 22:07 +0000 [r91010-91011] Russell Bryant * main/pbx.c, /: Merged revisions 90967 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90967 | russell | 2007-12-04 13:57:39 -0600 (Tue, 04 Dec 2007) | 7 lines Make some changes to some additions I made recently for doing channel autoservice when looking up extensions. This code was added to handle the case where a dialplan switch was in use that could block for a long time. However, the way that I added it, it did this for all extension lookups. However, lookups in the in-memory tree of extensions should _not_ take long enough to matter. So, move the autoservice stuff to be only around executing a switch. ........ * channels/chan_zap.c: Fix resetting mwimonitornotify on reload. I guess I only added this line in my head. (thanks to tzafrir for pointing it out) 2007-12-04 21:46 +0000 [r90993] Tilghman Lesher * channels/chan_usbradio.c: Coding guidelines fixups (Closes issue #11412) 2007-12-04 21:23 +0000 [r90991] Jason Parker * channels/chan_sip.c, CHANGES: Add manager action 'sipshowregistry'. Closes issue #11464, patch by eliel. 2007-12-04 19:08 +0000 [r90949] Russell Bryant * include/asterisk/callerid.h, channels/chan_zap.c, main/callerid.c, CHANGES, configs/zapata.conf.sample: Add support for monitoring MWI on FXO lines. This introduces two new options for zapata.conf: mwimonitor and mwimonitornotify. The mwimonitor option enables MWI monitoring. When the MWI state on a line changes, then the script specified by mwimonitornotify will be executed for custom handling of the state change, similar to the externnotify option of voicemail.conf. Also, when the MWI state on an FXO line changes, an internal Asterisk event is generated to indicate the new state of the associated mailbox. That may, any module that cares about MWI information will get notified and can handle it just as if app_voicemail had sent this notification. (BE-253, original patch from markster, with some minor modifications by me to add comments, documentation, and internal event support) 2007-12-04 18:29 +0000 [r90930] Mark Michelson * apps/app_voicemail.c: Kevin suggested doing the reverse of my last commit, since imap_retrieve_file does not modify the contents of the "mailbox" string. In other words, I'm changing the imap_retrieve_file function to take a const char* as the third argument so that I don't need to cast const char*'s as char*'s to suppress compiler warnings. 2007-12-04 18:15 +0000 [r90929] Jason Parker * Makefile: Add Makefile alias target 'pdf' which does the same thing as asterisk.pdf. Issue 11452, reported by blitzrage. 2007-12-04 18:14 +0000 [r90928] Mark Michelson * apps/app_voicemail.c: Suppress a compiler warning due to discarding a "const" qualifier 2007-12-04 18:09 +0000 [r90927] Jason Parker * main/global_datastores.c: Fix build, that some people aren't seeing for some reason. 2007-12-04 17:51 +0000 [r90899] Mark Michelson * apps/app_queue.c: Wrong locking style got merged from 1.4 to trunk. My mistake. 2007-12-04 17:40 +0000 [r90880] Kevin P. Fleming * channels/chan_zap.c: fix build of this module when libpri and/or libss7 are or are not present 2007-12-04 17:38 +0000 [r90879] Jason Parker * main/channel.c, /: Merged revisions 90876 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11454) ........ r90876 | qwell | 2007-12-04 11:28:08 -0600 (Tue, 04 Dec 2007) | 4 lines If we fail to create a channel after allocating a timing fd, we need to make sure to close it. Issue 11454, patch by eliel. ........ 2007-12-04 17:36 +0000 [r90878] Russell Bryant * main/Makefile: Fix a silly little typo :) 2007-12-04 17:35 +0000 [r90877] Jason Parker * apps/app_dial.c: Fix build in trunk. This was fixed in 1.4, but blocked in trunk since this hadn't been merged yet. 2007-12-04 17:08 +0000 [r90873] Mark Michelson * apps/app_dial.c, main/global_datastores.c (added), channels/chan_local.c, /, main/Makefile, include/asterisk/channel.h, include/asterisk/global_datastores.h (added), apps/app_queue.c: Merged revisions 90735 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03 Dec 2007) | 22 lines A big one... This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop. This is accomplished by creating a datastore on the calling channel which has a linked list of all devices dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore is detached from the channel and destroyed. This change also introduces some side effects to the code which I shall enumerate here: 1. Datastore inheritance has been backported from trunk into 1.4 2. A large chunk of code has been removed from app_dial. This chunk is the section of code which handles the call forward case after the channel has been requested but before it has been called. This was removed because call-forwarding still works fine without it, it makes the code less error-prone should it need changing, and it made this set of changes much less painful to just have the forwarding handled in one place in each module. 3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore which is attached to the channel may be created and attached in either app_dial or app_queue, so they need a common place to find the datastore info. This approach was taken in case similar datastores are needed in the future, there will be a common place to add them. ........ 2007-12-04 15:16 +0000 [r90852-90854] Olle Johansson * apps/app_queue.c: (closes issue #11431) Reported by: Laureano Patches: app_queue.c.patch uploaded by Laureano (license 265) * main/pbx.c, CHANGES: (closes issue #11422) Reported by: eliel Patches: core.show.hint.patch uploaded by eliel (license 64) * CHANGES: (closes issue #11462) Reported by: eliel Patches: CHANGES.patch uploaded by eliel (license 64) 2007-12-04 15:01 +0000 [r90851] Tilghman Lesher * res/res_agi.c: Pass the Asterisk version to AGI scripts as part of the initial dump of info Reported by: acunningham Patch by: acunningham (Closes issue #11398) 2007-12-04 11:50 +0000 [r90834] Luigi Rizzo * res/Makefile: fix build on cygwin 2007-12-03 23:52 +0000 [r90760] Tilghman Lesher * /, include/asterisk/compat.h: Merged revisions 90753 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90753 | tilghman | 2007-12-03 17:50:51 -0600 (Mon, 03 Dec 2007) | 5 lines Solaris requires the inclusion of sys/loadavg.h for getloadavg(). Reported by: snuffy Patch by: snuffy,tilghman (Closes issue #11430) ........ 2007-12-03 23:49 +0000 [r90746] Steve Murphy * main/hashtab.c: A small fix from snuffy 2007-12-03 23:48 +0000 [r90738] Jason Parker * res/res_monitor.c: Add manager events for when a monitor is started or stopped. Closes issue #10191, patch by dgradecak. 2007-12-03 23:29 +0000 [r90737] Tilghman Lesher * res/res_config_pgsql.c, /: Merged revisions 90736 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90736 | tilghman | 2007-12-03 17:23:55 -0600 (Mon, 03 Dec 2007) | 5 lines If both dbhost and dbsock were not set, a NULL deref could result Reported by: xrg Patch by: tilghman (Closes issue #11387) ........ 2007-12-03 22:07 +0000 [r90697] Jason Parker * /, apps/app_meetme.c: Merged revisions 90696 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes issue #11383) ........ r90696 | qwell | 2007-12-03 16:06:36 -0600 (Mon, 03 Dec 2007) | 4 lines Make sure we always close the conference fd if we have an open one. Issue 11383, reported by markmhy, patch by eliel. ........ 2007-12-03 21:24 +0000 [r90670] Mark Michelson * apps/app_voicemail.c: Replacing some calls to free() with ast_free(). (closes issue #11448, reported and patched by jaroth) 2007-12-03 21:03 +0000 [r90656] Joshua Colp * include/asterisk/agi.h, res/res_agi.c, CHANGES: Add AGI commands for speech recognition. These mirror the dialplan applications mostly but present the information in a nicer fashion. The SPEECH RECOGNIZE command for example will return the results instead of having to query the dialplan functions. 2007-12-03 21:00 +0000 [r90644] Mark Michelson * /, channels/chan_mgcp.c: Merged revisions 90639 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90639 | mmichelson | 2007-12-03 14:59:51 -0600 (Mon, 03 Dec 2007) | 5 lines Changing some bad logic when calculating the interdigit timeout. (closes issue #11402, reported and patched by eferro) ........ 2007-12-03 20:58 +0000 [r90631] Jason Parker * /, res/res_features.c: Merged revisions 90607 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes issue #11436) ........ r90607 | qwell | 2007-12-03 14:51:17 -0600 (Mon, 03 Dec 2007) | 4 lines Fix crash in ParkAndAnnounce application. Issue #11436, reported by lytledd, patch by eliel. ........ 2007-12-03 20:30 +0000 [r90591] Tilghman Lesher * main/channel.c: Avoid an additional function call. Reported by: eliel Patch by: eliel (Closes issue #11438) 2007-12-03 20:07 +0000 [r90550-90589] Joshua Colp * /, main/rtp.c: Merged revisions 90588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90588 | file | 2007-12-03 16:05:42 -0400 (Mon, 03 Dec 2007) | 2 lines Do not create a smoother for G723.1 frames, they need to be left alone to their native 20/24 byte size. ........ * main/channel.c, /, include/asterisk/channel.h, .cleancount: Merged revisions 90548 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90548 | file | 2007-12-03 14:40:56 -0400 (Mon, 03 Dec 2007) | 2 lines Preserve the indication currently playing on a channel when a masquerade operation happens. (issue #BE-88) ........ 2007-12-03 16:46 +0000 [r90528] Mark Michelson * configs/queues.conf.sample: Updating sample queues.conf file to show how multiple periodic announcements may be specified since this was not documented previously (closes issue #11432, reported and patched by Laureano) 2007-12-03 14:14 +0000 [r90508] Joshua Colp * apps/app_dial.c: Remove the file descriptors from the main poll channel when the channel is hung up during the dialing attempt, and make sure a channel exists before trying to remove it at the end. (closes issue #11441) Reported by: blitzrage 2007-12-02 18:20 +0000 [r90471] Russell Bryant * /, apps/app_queue.c: Merged revisions 90470 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90470 | russell | 2007-12-02 12:18:52 -0600 (Sun, 02 Dec 2007) | 6 lines The other day when I went through making changes as a result of the ao2_link() change, I added some code to set pointers to NULL after they were unreferenced. This pointed out that in this place, the object was unreferenced before the code was done using it. So, move the unref down a little bit. (crash reported by jmls on IRC) ........ 2007-12-02 09:42 +0000 [r90433] Tilghman Lesher * main/autoservice.c, /: Merged revisions 90432 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90432 | tilghman | 2007-12-02 03:34:23 -0600 (Sun, 02 Dec 2007) | 7 lines Clarify the return value on autoservice. Specifically, if you started autoservice and autoservice was already on, it would erroneously return an error. Reported by: adiemus Patch by: dimas (Closes issue #11433) ........ 2007-12-01 01:37 +0000 [r90410] Jason Parker * res/res_adsi.c: Only reload if the config file has changed. Closes issue #11281, patch by eliel. 2007-11-30 21:19 +0000 [r90388] Mark Michelson * apps/app_dial.c, include/asterisk/app.h, include/asterisk/audiohook.h, res/res_features.c, include/asterisk/channel.h, main/audiohook.c, apps/app_queue.c, configs/features.conf.sample: Adding support for the "automixmonitor" dial and queue options. This works in much the same way as the automonitor, except that instead of using the monitor app, it uses the mixmonitor app. By providing an 'x' or 'X' as a dial or queue option, a DTMF sequence may be entered (as defined in features.conf) to start the one-touch mixmonitor. This patch also introduces some new API calls to the audiohooks code for searching for an audiohook by type and for searching for a running audiohook by type. Big thanks to joetester for writing the initial patch, testing it and patiently waiting for it to be committed. (closes issue #10185, reported and patched by xmarksthespot) 2007-11-30 19:34 +0000 [r90311-90351] Russell Bryant * main/manager.c, /, include/asterisk/astobj2.h, apps/app_queue.c, channels/chan_iax2.c, main/astobj2.c, main/config.c: Merged revisions 90348 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90348 | russell | 2007-11-30 13:26:04 -0600 (Fri, 30 Nov 2007) | 8 lines Change the behavior of ao2_link(). Previously, in inherited a reference. Now, it automatically increases the reference count to reflect the reference that is now held by the container. This was done to be more consistent with ao2_unlink(), which automatically releases the reference held by the container. It also makes it so it is no longer possible for a pointer to be invalid after ao2_link() returns. ........ * /, include/asterisk/astobj2.h: Merged revisions 90310 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90310 | russell | 2007-11-30 12:46:46 -0600 (Fri, 30 Nov 2007) | 2 lines Add some notes on the behavior of ao2_unlink() after a discussion with Tilghman ........ 2007-11-30 14:45 +0000 [r90270] Joshua Colp * /, channels/chan_sip.c: Merged revisions 90269 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90269 | file | 2007-11-30 10:43:15 -0400 (Fri, 30 Nov 2007) | 6 lines Fix locking issues under one legged replaces scenarios. (closes issue #11420) Reported by: irroot Patches: chan_sip_oneleg.patch uploaded by irroot (license 52) ........ 2007-11-30 00:16 +0000 [r90164-90232] Mark Michelson * /, channels/chan_mgcp.c: Merged revisions 90231 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90231 | mmichelson | 2007-11-29 18:16:04 -0600 (Thu, 29 Nov 2007) | 5 lines Clear the DTMF buffer if the call times out. (closes issue #11418, reported and patched by eferro) ........ * /, apps/app_queue.c: Merged revisions 90163 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90163 | mmichelson | 2007-11-29 13:38:39 -0600 (Thu, 29 Nov 2007) | 6 lines This patch handles the case where a queue member with a negative penalty is added via the manager. If a negative value is submitted for a member penalty, we set it to 0. (closes issue #11411, reported and patched by Laureano) ........ 2007-11-29 19:35 +0000 [r90156-90162] Tilghman Lesher * res/res_config_pgsql.c, /: Merged revisions 90160 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90160 | tilghman | 2007-11-29 13:24:11 -0600 (Thu, 29 Nov 2007) | 2 lines Properly escape input buffers (Fixes AST-2007-025) ........ * /, formats/format_wav.c, formats/format_pcm.c, formats/format_ogg_vorbis.c, main/file.c, include/asterisk/mod_format.h, formats/format_h263.c, formats/format_h264.c, formats/format_wav_gsm.c, formats/format_g726.c: Merged revisions 90155 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90155 | tilghman | 2007-11-29 11:29:59 -0600 (Thu, 29 Nov 2007) | 5 lines Use of "private" as a field name in a header file messes with C++ projects Reported by: chewbacca Patch by: casper (Closes issue #11401) ........ * include/asterisk/lock.h: Fix build of trunk * /, sounds/Makefile: Merged revisions 90154 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90154 | tilghman | 2007-11-29 11:18:09 -0600 (Thu, 29 Nov 2007) | 2 lines Upgrade the core sounds release version ........ 2007-11-29 13:38 +0000 [r90149-90150] Kevin P. Fleming * utils/Makefile, utils/hashtest.c: let's try this again... *all* compilation and linking in Asterisk should be done using the standard compilation rules, not manually created ones. changing hashtest.c to use these rules caused the compiler to notice a large number of coding guidelines violations, so those are fixed too. * main/manager.c: restore behavior from the 1.4 branch... manager users created via users.conf should default to *all* permissions, not none 2007-11-29 00:37 +0000 [r90139-90148] Russell Bryant * main/channel.c, /, include/asterisk/channel.h, funcs/func_callerid.c: Merged revisions 90145 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90145 | russell | 2007-11-28 18:20:34 -0600 (Wed, 28 Nov 2007) | 5 lines This set of changes is to make some callerID handling thread-safe. The ast_set_callerid() function needed to lock the channel. Also, the handlers for the CALLERID() dialplan function needed to lock the channel when reading or writing callerid values directly on the channel structure. ........ * include/asterisk/file.h, /, main/file.c: Merged revisions 90142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90142 | russell | 2007-11-28 18:06:08 -0600 (Wed, 28 Nov 2007) | 4 lines Merge a change from team/russell/chan_refcount ... This makes ast_stopstream() thread-safe. ........ * include/asterisk/audiohook.h: Merge another small doxygen change from team/russell/chan_refcount to indicate that a channel doesn't need to be locked before calling a certain function. * include/asterisk/channel.h: Merge some channel.h doxygen updates from team/russell/chan_refcount This was mostly to note whether a channel needed to be locked or not before calling these functions. However, I added some other things, too. 2007-11-28 23:03 +0000 [r90102] Joshua Colp * /, res/res_musiconhold.c, apps/app_queue.c: Merged revisions 90101 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90101 | file | 2007-11-28 18:59:28 -0400 (Wed, 28 Nov 2007) | 6 lines Fix a few memory leaks. (closes issue #11405) Reported by: eliel Patches: load_realtime.patch uploaded by eliel (license 64) ........ 2007-11-28 22:44 +0000 [r90100] Kevin P. Fleming * configs/users.conf.sample, main/manager.c, /: Merged revisions 90098 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90098 | kpfleming | 2007-11-28 16:30:46 -0600 (Wed, 28 Nov 2007) | 2 lines it is impossible to set permissions for manager accounts created by users.conf (reported internally, patched by me) ........ 2007-11-28 22:32 +0000 [r90099] Joshua Colp * main/cli.c: file says... compile before you commit! 2007-11-28 22:17 +0000 [r90060-90061] Mark Michelson * main/pbx.c: Removing a pointless check of option_debug * main/pbx.c, /: Merged revisions 90059 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90059 | mmichelson | 2007-11-28 16:08:50 -0600 (Wed, 28 Nov 2007) | 13 lines Removing some seemingly pointless code. This sets a channel variable for every priority executed in the dialplan if you have debug set to anything non-zero. This seems pointless due to the fact that these channel variables are not referenced anywhere else in the code and their names are esoteric enough that they would not be practical to reference in the dialplan. Plus the fact that this behavior isn't documented anywhere means that the change is not likely to cause any disruption. If anything, this may actually cause a slight performance increase if running with debug on. The motivating influence for this code change is the eventwhencalled option for queues. If set to vars, all channel variables will be output to the manager. These unnecessary channel variables make the output a lot more difficult to deal with. ........ 2007-11-28 20:33 +0000 [r90039] Steve Murphy * main/ast_expr2f.c, main/ast_expr2.fl: Made expr2 parser 8-bit transparent 2007-11-28 20:27 +0000 [r90038] Jason Parker * main/pbx.c, res/res_crypto.c, include/asterisk/cli.h, main/cli.c: Remove "old"-style CLI handler, since nothing uses it anymore. Closes issue #11403, patch by eliel. This also completes the janitor project. 2007-11-28 15:48 +0000 [r89981-89982] Joshua Colp * main/cli.c: Hide CLI commands starting with _ from tab completion as was done previously. (closes issue #11395) Reported by: eliel Patches: cli.c.patch uploaded by eliel (license 64) * main/abstract_jb.c, res/res_agi.c: Fix a few log messages. (closes issue #11396) Reported by: IgorG Patches: spell.v1.diff uploaded by IgorG (license 20) 2007-11-28 00:49 +0000 [r89947] Russell Bryant * apps/app_voicemail.c: Merge some little changes from team/russell/chan_refcount to help reduce the diff to trunk. This just removes some checks on the return value of alloca(), as behavior is undefined if it runs out of stack space, and we don't check it anywhere else. 2007-11-28 00:47 +0000 [r89946] Mark Michelson * configs/musiconhold.conf.sample, configs/extconfig.conf.sample, res/res_musiconhold.c, CHANGES: Adding support for realtime music on hold. The following are the main points: 1. When moh is started, we search first in memory to find the class. If we do not find it in memory, we search realtime instead. 2. When moh is restarted (as in, it had been started on this particular channel, stopped, and now we're starting it again), if using the "files" mode, then realtime will always be rechecked. If you are using other modes, however, we will simply reattach to the external running process which was playing moh earlier in the call. This is a necessary compromise so that we don't end up with too many background processes. 3. musiconhold.conf has a general section now. It has one option: cachertclasses. If set to yes, then moh classes found in realtime will be added to the in-memory list. This has the advantage of not requiring database lookups each time moh is started, but it has the disadvantage of not truly being realtime. I have tested this for functionality, and it passes. I also tested this under valgrind and there are no memory problems reported under typical use. Special thanks to Sergee for implementing this feature and enduring my complaints on the bugtracker! (closes issue #11196, reported and patched by sergee) 2007-11-28 00:24 +0000 [r89840-89915] Russell Bryant * main/pbx.c, /, include/asterisk/pbx.h: Merged revisions 89893 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89893 | russell | 2007-11-27 18:20:13 -0600 (Tue, 27 Nov 2007) | 4 lines - update documentation for some of the goto functions to note that they handle locking the channel as needed - update ast_explicit_goto() to lock the channel as needed ........ * include/asterisk/channel.h: Document that the channel is not locked when the send_digit_begin and end callbacks get called. * main/autoservice.c, /: Merged revisions 89886 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89886 | russell | 2007-11-27 17:47:28 -0600 (Tue, 27 Nov 2007) | 2 lines Don't do frame processing if ast_read() returned NULL. ........ * channels/chan_iax2.c: Merge changes from team/russell/iax2_frame_queue This patch is an optimization for chan_iax2. This module is now heavily multi-threaded. However, there is still a good number of globally shared resources that prevent things from happen asynchronously. One of those things was the global IAX frame queue. This queue was used to hold frames that have been deferred for transmitting by another thread, and frames that may need to get retransmitted. I changed the frame queue to be per-call, since almost all of the frame queue handling only cares about frames specific to a call number. * /, apps/app_queue.c: Merged revisions 89844 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89844 | russell | 2007-11-27 17:21:13 -0600 (Tue, 27 Nov 2007) | 3 lines Instead of depending on the return value of ast_true(), explicitly set the eventwhencalled variable to 1. ........ * main/pbx.c, /: Merged revisions 89839 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89839 | russell | 2007-11-27 17:16:00 -0600 (Tue, 27 Nov 2007) | 2 lines Don't start/stop autoservice in pbx_extension_helper() unless a channel exists ........ 2007-11-27 23:11 +0000 [r89838] Mark Michelson * /, apps/app_queue.c: Merged revisions 89837 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89837 | mmichelson | 2007-11-27 17:10:05 -0600 (Tue, 27 Nov 2007) | 12 lines Two changes with regards to the 'eventwhencalled' option of queues.conf 1) Due to some signed vs. unsigned silliness, setting 'eventwhencalled' to 'vars' or 'yes' did exactly the same thing. Thus the sign change of the ast_true call. 2) The vars2manager function overwrote a \n for every channel variable it parsed, resulting in bizarre output for the channel variables. This patch remedies this. (related to issue #11385, however I'm not sure if this will actually be enough to close it) ........ 2007-11-27 22:42 +0000 [r89835] Russell Bryant * channels/chan_misdn.c: Bring in a small change from team/russell/chan_refcount This replaces tab completion code with the use of a public function that does the same thing 2007-11-27 22:14 +0000 [r89792] Steve Murphy * main/pbx.c, pbx/pbx_config.c: closes issue #11294; missed the conditional unlock of the contexts when the hash table is used instead; also, used the ast_free_ptr as advised. 2007-11-27 22:05 +0000 [r89791] Russell Bryant * main/autoservice.c, main/pbx.c, /: Merged revisions 89790 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89790 | russell | 2007-11-27 15:45:51 -0600 (Tue, 27 Nov 2007) | 41 lines Merge changes from team/russell/autoservice_1.4 This set of changes fixes an issue that was reported to me on IRC yesterday. The user, d1mas, was using chan_zap for incoming calls and was having DTMF recognition issues in some situations. Specifically, he noticed that the problem occurred when using DISA or WaitExten. He also noticed that when using Read, the problem did not occur. His system also used DUNDi for dialplan lookups. So, he theorized that if the DUNDi lookups blocked for some period of time, that audio from the zap channel could get lost. If the audio got lost, then it wouldn't be run through the DTMF detector, and digits could get lost. He was correct, and the following set of changes fixes the problem. However, the changes go a little bit further than what was necessary to fix this exact problem. 1) I updated pbx_extension_helper() to autoservice the associated channel to handle cases where extension lookups may take a long time. This would normally be a dialplan switch that does some lookup over the network, such as the DUNDi or IAX2 switches. This ensures that even while a DUNDi lookup is blocking, the channel will be continuously serviced. 2) I made a change to the autoservice code. This is actually something that has bothered me for a long time. When a channel is in autoservice, _all_ frames get thrown away. However, some frames really shouldn't be thrown away. The most notable examples are signalling (CONTROL) frames, and DTMF. So, this patch queues up important frames while a channel is in autoservice. When autoservice is stopped on the channel, the queued up frames get stuck back on the channel so that they can get processed instead of thrown away. 3) I made another change to the autoservice code to handle the case where autoservice is started on channels recursively. Previously, you could call ast_autoservice_start() multiple times on a channel, and it would stop the first time ast_autoservice_stop() gets called. Now, it will ensure that autoservice doesn't actually stop until the final call to ast_autoservice_stop(). ........ 2007-11-27 21:10 +0000 [r89769-89772] Olle Johansson * main/dnsmgr.c, res/res_jabber.c, main/enum.c, main/asterisk.c: A few more "moremanager" fixes * include/asterisk.h, main/asterisk.c, main/loader.c: More "moremanager" fixes. Manager commands to check module status. * include/asterisk/manager.h: More "moremanager" changes - doxygen docs and changing manager version (finally) before making more dramatic changes. * channels/chan_iax2.c: More additions from the "moremanager" branch, this time for IAX2. 2007-11-27 20:21 +0000 [r89721] Kevin P. Fleming * /, main/app.c: Merged revisions 89709 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89709 | kpfleming | 2007-11-27 14:16:56 -0600 (Tue, 27 Nov 2007) | 2 lines on second thought... revert all the other changes i've made in app options parsing leaving only one: if an empty argument is supplied for an option, set that argument pointer to point to an empty string rather than NULL, so that the application can do normal checks on it without worrying about it being NULL ........ 2007-11-27 20:17 +0000 [r89710] Russell Bryant * channels/chan_sip.c: remove a duplicate manager event 2007-11-27 19:50 +0000 [r89706] Olle Johansson * channels/chan_gtalk.c: Manager events from the "moremanager" branch 2007-11-27 19:47 +0000 [r89704] Kevin P. Fleming * /, main/app.c: Merged revisions 89701 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89701 | kpfleming | 2007-11-27 13:36:55 -0600 (Tue, 27 Nov 2007) | 2 lines generate a warning when an application option that requires an argument is ignored due to lack of an argument ........ 2007-11-27 19:45 +0000 [r89698-89702] Olle Johansson * channels/chan_sip.c: Starting to merge changes from the "moremanager" branch. Documentation will follow. * /, channels/chan_sip.c, include/asterisk/rtp.h, main/rtp.c: The following patch with updates for trunk. Works much better in trunk. Also by accident fixed a bad typo by a previous committer, which actually made video calls not work fully... Merged revisions 89630 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89630 | oej | 2007-11-27 16:23:17 +0100 (Tis, 27 Nov 2007) | 12 lines If we get a codec offer using a well-known payload type, but using it for another codec that we don't know, Asterisk did not remove that codec from the list. With this patch, we remove the codec from audio and video rtp objects and deny it ever existed. Thanks to lasse for testing. (closes issue #11376) Reported by: lasse Patches: bug11376.txt uploaded by oej (license 306) Tested by: lasse ........ 2007-11-27 19:12 +0000 [r89683] Jason Parker * include/asterisk/strings.h: Add an S_COR macro, which is similar to the existing S_OR macro, except with an additional boolean arg. A hack such as: foo ? S_OR(bar, "baz") : "baz" becomes: S_COR(foo, bar, "baz") 2007-11-27 18:50 +0000 [r89682] Steve Murphy * res/ael/ael.y, pbx/ael/ael-test/ref.ael-test11, pbx/ael/ael-test/ref.ael-test20, pbx/ael/ael-test/ref.ael-test14, pbx/ael/ael-test/ref.ael-test15, pbx/ael/ael-test/ref.ael-ntest9, pbx/ael/ael-test/ref.ael-test16, pbx/ael/ael-test/ref.ael-test18, pbx/ael/ael-test/ref.ael-test19, pbx/ael/ael-test/ref.ael-ntest10, res/ael/ael.tab.c, pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-ntest12, pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-ntest22, res/ael/ael_lex.c, pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test5, pbx/ael/ael-test/ref.ael-test6, res/ael/ael.flex, pbx/ael/ael-test/ref.ael-test7, pbx/ael/ael-test/ref.ael-test8: made AEL 8-bit transparent; mainly the lexer was tossing chars with the hi-order bit set. Not nice. Also, allow @ in extension names, and a backslash, also. 2007-11-27 17:01 +0000 [r89637] Joshua Colp * main/utils.c: Ensure the value returned from ast_random is between 0 and RAND_MAX on 64-bit platforms. (closes issue #11348) Reported by: sperreault 2007-11-27 16:13 +0000 [r89635] Russell Bryant * /, configs/voicemail.conf.sample: Merged revisions 89634 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89634 | russell | 2007-11-27 10:12:33 -0600 (Tue, 27 Nov 2007) | 3 lines Add a note to the sample voicemail config noting that when using IMAP storage, only the first format specified will be attached to the message. ........ 2007-11-27 15:41 +0000 [r89632] Tilghman Lesher * /, funcs/func_env.c: Merged revisions 89631 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89631 | tilghman | 2007-11-27 09:38:03 -0600 (Tue, 27 Nov 2007) | 3 lines Default result of STAT should be "0" not "". Reported via the -users mailing list, fixed by me. ........ 2007-11-27 07:36 +0000 [r89625] Olle Johansson * /, configs/sip.conf.sample: Merged revisions 89624 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89624 | oej | 2007-11-27 08:34:19 +0100 (Tis, 27 Nov 2007) | 6 lines Clarify limitonpeers=yes (closes issue #11304) Reported by: pj ........ 2007-11-27 06:47 +0000 [r89623] Steve Murphy * apps/app_dial.c, main/cdr.c, /, configs/cdr.conf.sample, include/asterisk/cdr.h: Merged revisions 89622 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89622 | murf | 2007-11-26 23:24:02 -0700 (Mon, 26 Nov 2007) | 1 line closes issue #11379; OK, this is an attempt to make both sides happy. To the cdr.conf file, I added the option 'unanswered', which defaults to 'no'. In this mode, you will see a cdr for a call, whether it was answered or not. The disposition will be NO ANSWER or ANSWERED, as appropriate. The src is as you'd expect, the destination channel will be one of the channels from the Dial() call, usually the last in the list if more than one chan was specified. With unanswered set to 'yes', you will still see this cdr entry in both cases. But in the case where the dial timed out, you will also see a cdr for each line attempted, marked NO ANSWER, with no destination channel name. The new option defaults to 'no', so you don't see the pesky extra cdr's by default, and you will not see the irritating 'not posted' messages. ........ 2007-11-26 23:15 +0000 [r89617-89621] Mark Michelson * pbx/ael/ael-test/ael-test19/extensions.ael, pbx/ael/ael-test/ael-vtest13/extensions.ael, doc/osp.txt, pbx/ael/ael-test/ael-test3/extensions.ael, pbx/ael/ael-test/ref.ael-vtest13, pbx/ael/ael-test/ael-test7/extensions.ael: Change all instances of "CALLERID(number)" to "CALLERID(num)" for consistency's sake (closes issue #11381, reported and patched by jon) * /, apps/app_playback.c: Merged revisions 89618 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89618 | mmichelson | 2007-11-26 17:10:49 -0600 (Mon, 26 Nov 2007) | 7 lines After issuing a "say load new", if a caller hangs up during the middle of playback of a number, app_playback will continue to try to play the remaining files. With this change, no more files will be played back upon hangup. (closes issue #11345, reported and patched by IgorG) ........ 2007-11-26 22:52 +0000 [r89615] Russell Bryant * configure, configure.ac: Update the configure script check for libpri to check for the newest function that was just added. Cresl1n, please keep this in mind when making these changes to libpri or libss7. 2007-11-26 21:23 +0000 [r89613] Olle Johansson * channels/chan_sip.c, configs/sip.conf.sample: Rename "limitonpeer" to "counteronpeer" since the call-limit is deprecated. Both still works in this version. 2007-11-26 21:14 +0000 [r89612] Joshua Colp * main/dial.c, /: Merged revisions 89610 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89610 | file | 2007-11-26 17:10:29 -0400 (Mon, 26 Nov 2007) | 2 lines Fix issues with async dialing with an application executing. The application has to be terminated and control returned to the thread before hanging things up. (issue #BE-252) ........ 2007-11-26 21:12 +0000 [r89606-89611] Olle Johansson * channels/chan_sip.c: Formatting, doxygenification * channels/chan_sip.c: Formatting changes, cleaning up some code * include/asterisk/doxyref.h, channels/chan_sip.c: Start using Doxygen groupings to group variables and defines. * apps/app_meetme.c, UPGRADE.txt, CHANGES, main/cli.c: - Mark "concise" as deprecated - Restructure other changes to UPGRADE.txt and CHANGES We're still looking for scripts that replace asterisk -rx "show shannels concise" by using the manager interface, but still produces the same output. Anyone? 2007-11-26 18:11 +0000 [r89600-89602] Joshua Colp * res/res_features.c, apps/app_queue.c: Perform some module use counting audits. This is now done outside the scope of the application/dialplan function so they do not need to worry about it. * /, res/res_features.c: Merged revisions 89599 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89599 | file | 2007-11-26 14:02:56 -0400 (Mon, 26 Nov 2007) | 6 lines Add module counting removal for error conditions. (closes issue #11333) Reported by: Laureano Patches: res_features_v2.c.patch uploaded by Laureano (license 265) ........ 2007-11-26 17:49 +0000 [r89596] Russell Bryant * main/pbx.c, /: Merged revisions 89594 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89594 | russell | 2007-11-26 11:41:04 -0600 (Mon, 26 Nov 2007) | 3 lines Add channel locking to a function that needed to be doing it. This is just a little something I noticed while working on a completely unrelated issue. ........ 2007-11-26 17:46 +0000 [r89595] Steve Murphy * utils/ael_main.c, utils/conf2ael.c, utils/check_expr.c: closes issue #11341; made changes to make utils again right with the MTX_PROFILE world. 2007-11-26 17:38 +0000 [r89593] Joshua Colp * /, pbx/pbx_config.c: Merged revisions 89592 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89592 | file | 2007-11-26 13:36:45 -0400 (Mon, 26 Nov 2007) | 6 lines Use ast_free to free memory, or else we shall implode if MALLOC_DEBUG is enabled. (closes issue #11347) Reported by: ys Patches: pbx.pbx_config.c.diff uploaded by ys (license 281) ........ 2007-11-26 17:26 +0000 [r89591] Steve Murphy * main/hashtab.c: closes issue #11356; Many thanks to snuffy for his code review and changes to cut down duplication. I tested this against hashtest, and it passes. I reviewed the changes, and they look reasonable. I had to remove a few const decls to make things compile on my workstation, 2007-11-26 17:25 +0000 [r89590] Russell Bryant * Makefile: make sure we check to see if the configure script has been executed on a new checkout or after a distclean 2007-11-26 17:23 +0000 [r89589] Joshua Colp * /, apps/app_mixmonitor.c: Merged revisions 89587 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89587 | file | 2007-11-26 13:20:58 -0400 (Mon, 26 Nov 2007) | 6 lines Close the audio file before sending it to the post processing application. (closes issue #11357) Reported by: reformed Patches: mixmonitor.patch uploaded by reformed (license 330) ........ 2007-11-26 17:21 +0000 [r89588] Kevin P. Fleming * /, main/app.c, apps/app_controlplayback.c: Merged revisions 89586 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89586 | kpfleming | 2007-11-26 11:20:36 -0600 (Mon, 26 Nov 2007) | 2 lines when parsing application options that take arguments, don't indicate that the option was supplied unless a non-zero-length argument was found for it ........ 2007-11-26 16:24 +0000 [r89583] Steve Murphy * main/pbx.c, CHANGES, configs/extensions.conf.sample: Thanks to pnlarsson for noting the spelling error in the cli commands. Also, added some verbage about the new algorithm to CHANGES. 2007-11-26 16:20 +0000 [r89582] Joshua Colp * main/utils.c: Revert change for 11348 until it can be looked at even more. 2007-11-26 15:50 +0000 [r89581] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 89580 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89580 | mmichelson | 2007-11-26 09:48:06 -0600 (Mon, 26 Nov 2007) | 6 lines Revert vmu->email back to an empty string if it was empty when imap_store_file was called. This prevents sending a duplicate e-mail. (closes issue #11204, reported by spditner, patched by me) ........ 2007-11-26 15:36 +0000 [r89570-89578] Joshua Colp * main/channel.c, /: Merged revisions 89577 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89577 | file | 2007-11-26 11:34:38 -0400 (Mon, 26 Nov 2007) | 6 lines If channel allocation fails because the alert pipe could not be created also free the scheduler context. (closes issue #11355) Reported by: eliel Patches: main.channel.c.patch uploaded by eliel (license 64) ........ * main/utils.c: Make the behavior of using /dev/urandom for random numbers the same as random(). (closes issue #11348) Reported by: sperreault Patches: ast_random2.diff uploaded by sperreault (license 252) * channels/chan_sip.c: Instead of printing out one codec in sip show channels print out all of the native ones (this is for video). (closes issue #11366) Reported by: ovi * /, apps/app_meetme.c: Merged revisions 89571 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89571 | file | 2007-11-26 10:41:03 -0400 (Mon, 26 Nov 2007) | 4 lines When unloading app_meetme destroy any auto created contexts created by SLA. (closes issue #11367) Reported by: eliel ........ * apps/app_controlplayback.c: Don't crash if the 'o' option of ControlPlayback is used without any value. (closes issue #11375) Reported by: johan 2007-11-25 21:12 +0000 [r89564-89566] Olle Johansson * channels/chan_usbradio.c: Formatting changes * main/channel.c, include/asterisk/channel.h: Try to get channel.h and channel.c aligned in regards to ast_set_callerid as well as change name of variables to follow the rest of the naming. 2007-11-25 17:50 +0000 [r89560-89561] Tilghman Lesher * include/asterisk/res_odbc.h, res/res_config_odbc.c, /, res/res_odbc.c, configs/res_odbc.conf.sample: Merged revisions 89559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89559 | tilghman | 2007-11-25 11:17:10 -0600 (Sun, 25 Nov 2007) | 14 lines We previously attempted to use the ESCAPE clause to set the escape delimiter to a backslash. Unfortunately, this does not universally work on all databases, since on databases which natively use the backslash as a delimiter, the backslash itself needs to be delimited, but on other databases that have no delimiter, backslashing the backslash causes an error. So the only solution that I can come up with is to create an option in res_odbc that explicitly specifies whether or not backslash is a native delimiter. If it is, we use it natively; if not, we use the ESCAPE clause to make it one. Reported by: elguero Patch by: tilghman (Closes issue #11364) ........ * channels/chan_sip.c: Typo (someone needs to test compile before committing his changes) 2007-11-25 12:18 +0000 [r89551-89557] Olle Johansson * channels/chan_sip.c: More doxygen changes * channels/chan_sip.c: Housekeeping * channels/chan_sip.c: Formatting, doxygen updates * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: - Deprecate "call-limit" in chan_sip. No other channel driver enforces call-limits and we now have the groupcount system to implement call-limits in the dialplan. You can use the "setvar" option in realtime/sip.conf to set limits per device. - Implement "callcounter" as a new option to enable the call counting we need to report device status to queue, manager and SIP subscriptions. The call counter setting is now enabled in the code by setting the device call-limit to 999. When we remove the call limit, we can simply enable this with a boolean setting. * channels/chan_sip.c, include/asterisk/channel.h: Housekeeping... - Fix typo in chan_sip - Remove changes to caller ID structure, moving it to branch (russellb) 2007-11-24 21:00 +0000 [r89547] Steve Murphy * main/pbx.c, include/asterisk/pbx.h, pbx/pbx_config.c, configs/extensions.conf.sample: closes issue #11363; where the pattern _20x. buried in an included context, didn't match 2012; There were a small set of problems to fix: 1. I needed NOT to score patterns unless you are at the end of the data string. 2. Capital N,X,Z and small n,x,z are OK in patterns. I canonicalize the patterns in the trie to caps. 3. When a pattern ends with dot or exclamation, CANMATCH/MATCHMORE should always report this pattern, no matter the length. With this commit, I also supplied the wish of Luigi, where the user can select which pattern matching algorithm to use, the old (legacy) pattern matcher, or the new, trie based matcher. The OLD matcher is the default. A new [general] section variable, extenpatternmatchnew, is added to the extensions.conf, and the example config has it set to false. If true, the new matcher is used. In all other respects, the context/exten structs are the same; the tries and hashtabs are formed, but in the new mode the tries are not used. A new CLI command 'dialplan set extenpatternmatch true/false' is provided to allow switching at run time. I beg users that are forced to return to the old matcher to please report the reason in the bug tracker. Measured the speed benefit of the new matcher against an impossibly large context with 10,000 extensions: the new matcher is 374 times faster. 2007-11-24 17:07 +0000 [r89546] Tilghman Lesher * /, res/res_adsi.c: Merged revisions 89545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89545 | tilghman | 2007-11-24 10:59:59 -0600 (Sat, 24 Nov 2007) | 5 lines Free some frames that would otherwise leak on error. Reported by: Laureano Patch by: Laureano,tilghman (Closes issue #11351) ........ 2007-11-24 16:53 +0000 [r89544] Steve Murphy * main/app.c: Added include to allow trunk to compile. Hope this doesn't louse thing up. 2007-11-24 13:57 +0000 [r89542-89543] Luigi Rizzo * channels/chan_h323.c: remove a DEBUG_THREADS message that accesses private lock fields. If needed, the code to extract this information should be implemented in some generic header or library and the function called here. (closed bug #11362) * main/acl.c, main/http.c, main/app.c: remove some unnecessary includes 2007-11-24 06:24 +0000 [r89535-89541] Tilghman Lesher * /, main/app.c, apps/app_voicemail.c: Merged revisions 89540 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89540 | tilghman | 2007-11-24 00:19:23 -0600 (Sat, 24 Nov 2007) | 9 lines Currently, zero-length voicemail messages cause a hangup in VoicemailMain. This change fixes the problem, with a multi-faceted approach. First, we do our best to avoid these messages from being created in the first place, and second, if that fails, we detect when the voicemail message is zero-length and avoid exiting at that point. Reported by: dtyoo Patch by: gkloepfer,tilghman (Closes issue #11083) ........ * main/manager.c, /: Merged revisions 89536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89536 | tilghman | 2007-11-23 11:18:26 -0600 (Fri, 23 Nov 2007) | 10 lines Up until this point, the XML output of the manager has been technically invalid, due to the repetition of certain parameters in a single event. This caused various issues for XML parsers, some of which refused to parse at all, given the invalidity of the rendered XML. So this commit fixes the XML output, ensuring that each entity parameter has a unique name, thus ensuring valid XML. Reported by: msetim Patch by: tilghman (Closes issue #10220) ........ * res/res_config_odbc.c, /: Merged revisions 89534 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89534 | tilghman | 2007-11-23 11:05:10 -0600 (Fri, 23 Nov 2007) | 5 lines Use ESCAPE clause for the first parameter, not just 2nd-Nth parameters. Reported by: apsaras Patch by: tilghman (Closes issue #11353) ........ 2007-11-23 15:54 +0000 [r89532-89533] Luigi Rizzo * channels/chan_oss.c: put in the necessary hooks for video support in the console. This is a NOP as far as the current code is concerned, but there is already support in ./configure and the Makefiles for the various libraries used by console_video.c (not yet in the tree) so addition is trivial. * channels/chan_sip.c: set rtpmap video info according to what is read from SDP; make the format explicit in a debug message; print the audio instead of aggregated peer capability in a debugging msg. 2007-11-23 09:40 +0000 [r89531] Olle Johansson * include/asterisk/channel.h: Let's start with implementing the base architecture for UTF8 caller ID's so we can handle multiple formats properly. This is not carved in stone, but a proposal to start with. We need to add support for transliterations as well as UTF8 handling, propably with libiconv. Murf is looking into that for the dialplan. 2007-11-23 09:03 +0000 [r89530] Luigi Rizzo * include/asterisk/image.h, formats/format_jpeg.c: formatting cleanup on the header, normalization of the assignment of descriptor fields. 2007-11-23 02:37 +0000 [r89529] Russell Bryant * configs/agents.conf.sample, /: Merged revisions 89527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89527 | russell | 2007-11-22 12:29:41 -0500 (Thu, 22 Nov 2007) | 3 lines mvanbaak pointed out a spelling error in this sample configuration file. While I was at it, I went ahead and tweaked it a little bit more. ........ 2007-11-22 07:10 +0000 [r89514-89526] Luigi Rizzo * doc/CODING-GUIDELINES: new info on the management of headers * apps/app_echo.c, apps/app_sendtext.c, apps/app_verbose.c, apps/app_milliwatt.c: more header removal * include/asterisk/channel.h: formatting cleanup * include/asterisk.h, apps/app_read.c, apps/app_record.c, apps/app_echo.c, apps/app_readexten.c, include/asterisk/channel.h, apps/app_system.c, apps/app_transfer.c, res/ael/pval.c, include/asterisk/app.h, apps/app_dumpchan.c, include/asterisk/module.h, apps/app_url.c, include/asterisk/pbx.h, apps/app_senddtmf.c, pbx/pbx_config.c, apps/app_mixmonitor.c, apps/app_stack.c, apps/app_verbose.c, apps/app_milliwatt.c, apps/app_cdr.c, apps/app_while.c: shuffle a little bit the content of header files to reduce dependencies. In this commit: - move the ast_register/unregister_app functions to module.h to avoid the need to include pbx.h for the simpler apps; - move the ast_group structure to channel.h to remove the dependency of app.h on linkedlists.h Note, this is a long process that I am doing in small steps. The main difficulty is that now for each subsystem we have a single header (e.g. channel.h) included by the subsystem provider (usually one file, e.g. channel.c) and by its clients (dozens of them, e.g. we have some 70+ apps and 30+ functions). This requires the clients to include all the extra headers required by the provider (eg. lock.h, linkedlists.h, definitions of substructures...) even though many of the clients would be just happy with opaque struct declarations and function prototypes. The long term plan is to eventually rectify this structure so that the compilation can become faster, and also APIs are more stable. * funcs/func_md5.c, funcs/func_module.c, funcs/func_blacklist.c, apps/app_url.c, funcs/func_sha1.c, funcs/func_global.c: remove some useless includes * include/asterisk/audiohook.h, apps/app_dictate.c, apps/app_readexten.c, apps/app_directory.c, apps/app_senddtmf.c, apps/app_mixmonitor.c, apps/app_stack.c, apps/app_controlplayback.c: more removal of redundant headers * apps/app_read.c, apps/app_echo.c, apps/app_record.c, apps/app_userevent.c, apps/app_image.c, apps/app_system.c, apps/app_verbose.c, apps/app_milliwatt.c, apps/app_playback.c, apps/app_while.c: remove redundant headers * main/file.c, main/netsock.c: more removal of fcntl.h and other system headers * codecs/codec_lpc10.c, codecs/codec_g722.c, codecs/codec_a_mu.c, codecs/codec_speex.c, codecs/codec_alaw.c, codecs/codec_adpcm.c, res/res_crypto.c, codecs/codec_g726.c, apps/app_test.c, formats/format_ogg_vorbis.c, codecs/codec_gsm.c, res/res_agi.c, apps/app_mp3.c, main/app.c, codecs/codec_ulaw.c, codecs/codec_ilbc.c: remove a number of #include which are either useless or done elsewhere * formats/format_sln.c, formats/format_wav.c, formats/format_ogg_vorbis.c, include/asterisk/_private.h, formats/format_wav_gsm.c, formats/format_ilbc.c, include/asterisk/file.h, formats/format_vox.c, formats/format_pcm.c, main/file.c, formats/format_h263.c, formats/format_g723.c, formats/format_h264.c, include/asterisk/frame.h, formats/format_jpeg.c, formats/format_g726.c, formats/format_gsm.c, formats/format_g729.c: implement the split of file.h and mod_format.h * include/asterisk/mod_format.h (added): Add a specific header for providers of file and format handling routines, moving here structs and function declarations formerly in file.h 2007-11-21 23:54 +0000 [r89513] Steve Murphy * apps/app_dial.c, channels/chan_sip.c, channels/chan_skinny.c, res/res_features.c, apps/app_queue.c, channels/chan_iax2.c: closes issue #11285, where an unload of a module that creates a dialplan context, causes a crash when you do a 'dialplan show' of that context. This is because the registrar string is defined in the module, and the stale pointer is traversed. The reporter offered a patch that would always strdup the registrar string, which is practical, but I preferred to destroy the created contexts in each module where one is created. That seemed more symmetric. There were only 6 place in asterisk where this is done: chan_sip, chan_iax2, chan_skinny, res_features, app_dial, and app_queue. The two apps destroyed the context, but left the contexts. All is fixed now and unloads should be dialplan friendly. 2007-11-21 23:24 +0000 [r89511-89512] Luigi Rizzo * funcs/func_rand.c, cdr/cdr_sqlite3_custom.c, apps/app_readfile.c, channels/chan_local.c, apps/app_record.c, funcs/func_strings.c, apps/app_sayunixtime.c, apps/app_test.c, apps/app_alarmreceiver.c, cdr/cdr_adaptive_odbc.c, apps/app_image.c, apps/app_chanisavail.c, apps/app_ices.c, channels/chan_iax2.c, apps/app_exec.c, pbx/pbx_loopback.c, pbx/pbx_spool.c, channels/chan_skinny.c, apps/app_dumpchan.c, apps/app_zapscan.c, apps/app_zapras.c, pbx/pbx_realtime.c, channels/chan_alsa.c, apps/app_amd.c, apps/app_url.c, apps/app_externalivr.c, cdr/cdr_odbc.c, apps/app_dial.c, funcs/func_timeout.c, apps/app_page.c, apps/app_privacy.c, channels/chan_agent.c, apps/app_disa.c, apps/app_morsecode.c, channels/iax2-provision.c, funcs/func_cut.c, apps/app_talkdetect.c, apps/app_transfer.c, apps/app_db.c, apps/app_playback.c, funcs/func_curl.c, channels/chan_misdn.c, apps/app_zapbarge.c, apps/app_waitforring.c, apps/app_sendtext.c, channels/chan_features.c, apps/app_macro.c, funcs/func_iconv.c, pbx/pbx_config.c, apps/app_mixmonitor.c, apps/app_chanspy.c, apps/app_voicemail.c, channels/chan_unistim.c, channels/chan_vpb.cc, apps/app_meetme.c, apps/app_authenticate.c, apps/app_readexten.c, funcs/func_vmcount.c, channels/chan_gtalk.c, cdr/cdr_pgsql.c, apps/app_followme.c, cdr/cdr_radius.c, apps/app_controlplayback.c, cdr/cdr_csv.c, channels/chan_phone.c, funcs/func_enum.c, apps/app_osplookup.c, funcs/func_odbc.c, apps/app_mp3.c, apps/app_minivm.c, apps/app_rpt.c, channels/chan_mgcp.c, apps/app_parkandannounce.c, apps/app_while.c, apps/app_adsiprog.c, apps/app_nbscat.c, funcs/func_version.c, funcs/func_db.c, channels/chan_zap.c, apps/app_read.c, channels/chan_sip.c, apps/app_festival.c, apps/app_waitforsilence.c, funcs/func_lock.c, pbx/pbx_lua.c, apps/app_system.c, apps/app_getcpeid.c, apps/app_queue.c, channels/chan_oss.c, cdr/cdr_tds.c, funcs/func_realtime.c, channels/chan_jingle.c, channels/chan_usbradio.c, apps/app_channelredirect.c, apps/app_flash.c, apps/app_directed_pickup.c, funcs/func_blacklist.c, channels/chan_h323.c, pbx/pbx_dundi.c, apps/app_sms.c, channels/chan_nbs.c, apps/app_senddtmf.c, funcs/func_callerid.c, apps/app_verbose.c, apps/app_stack.c, pbx/pbx_gtkconsole.c: remove another set of redundant #include "asterisk/options.h" * main/udptl.c, main/autoservice.c, main/frame.c, res/res_snmp.c, main/say.c, res/res_features.c, main/devicestate.c, main/utils.c, res/res_musiconhold.c, res/res_jabber.c, main/indications.c, main/enum.c, res/res_config_sqlite.c, main/config.c, main/loader.c, main/term.c, main/cli.c, main/io.c, main/channel.c, main/cdr.c, main/dial.c, res/res_smdi.c, res/res_config_odbc.c, main/manager.c, res/res_agi.c, main/http.c, main/logger.c, res/res_realtime.c, main/app.c, main/image.c, main/dns.c, main/db.c, res/res_speech.c, main/sched.c, main/pbx.c, res/res_config_pgsql.c, main/dnsmgr.c, main/translate.c, res/res_crypto.c, res/res_adsi.c, main/jitterbuf.c, main/acl.c, formats/format_ogg_vorbis.c, res/res_ael_share.c, res/res_monitor.c, main/rtp.c, main/netsock.c, main/srv.c, main/hashtab.c, main/privacy.c, main/adsistub.c, main/abstract_jb.c, main/file.c, main/callerid.c, main/astmm.c, main/audiohook.c, formats/format_g726.c, main/asterisk.c, res/res_odbc.c, main/dsp.c: remove a bunch of useless #include "options.h" 2007-11-21 22:37 +0000 [r89509-89510] Matthew Fredrickson * channels/chan_zap.c: Remove unneccessary explicit case for BRI * channels/chan_zap.c: Take some debug code out :-) 2007-11-21 22:20 +0000 [r89508] Luigi Rizzo * main/cygload.c: add a missing return 2007-11-21 22:07 +0000 [r89507] Matthew Fredrickson * channels/chan_zap.c: Add BRI support to chan_zap 2007-11-21 21:30 +0000 [r89506] Luigi Rizzo * utils/Makefile, configure, configure.ac: enable support for stack backtrace for stuff built in utils/ (this was present in the main tree but forgotten here). 2007-11-21 20:38 +0000 [r89505] Steve Murphy * main/pbx.c: closes issue #11290; the proposed patch was a good guess, and would solve the bug to some extent, but was really masking the real issue, that there were bad entries in the table. This fix removes the condition that the hashtab updates be done on exten removal only when the pattern_tree was present, which is silly. The operations that apply to the pattern tree are instead made conditional. Also, threw back in routines that kpfleming deleted because of probs in the 64-bit world. Tested on both 32 and 64-bit machines (compile). Tested the reload problem with over 20 reloads, and no problems. If you find more problems, please reopen 11290. 2007-11-21 20:22 +0000 [r89504] Terry Wilson * res/res_features.c: Simplify comparison in parking fix 2007-11-21 19:28 +0000 [r89494-89496] Mark Michelson * /, apps/app_queue.c: Merged revisions 89495 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89495 | mmichelson | 2007-11-21 13:27:51 -0600 (Wed, 21 Nov 2007) | 3 lines Fix a small error I made in my previous commit ........ * /, apps/app_queue.c: Merged revisions 89493 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89493 | mmichelson | 2007-11-21 13:24:22 -0600 (Wed, 21 Nov 2007) | 5 lines Changing an inaccurate debug message to be less inaccurate. Under the circumstances, this message would always report that there were 0 members available, even though that may not be true. ........ 2007-11-21 19:20 +0000 [r89492] Terry Wilson * /, res/res_features.c: Merged revisions 89491 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89491 | twilson | 2007-11-21 12:59:27 -0600 (Wed, 21 Nov 2007) | 4 lines If a channel gets masqueraded in the middle of a park, don't play the announcement to the masqueraded channel, and dial back to the original channel on timeout. ........ 2007-11-21 18:52 +0000 [r89490] Russell Bryant * main/dsp.c: Remove obsolete OLD_DSP_ROUTINES code. Also, remove the FAX_DETECT define and only do the calculations if fax detection is enabled on the dsp. (closes issue #11331) Reported by: dimas Patches: dsp.patch uploaded by dimas (license 88) 2007-11-21 18:38 +0000 [r89489] Tilghman Lesher * apps/app_read.c, UPGRADE.txt, CHANGES: Change Read to set READSTATUS as an indication of the result Also, some cleanup to CHANGES. Reported by: michael-fig Patch by: michael-fig,tilghman (Closes issue #11004) 2007-11-21 18:24 +0000 [r89488] Russell Bryant * channels/chan_iax2.c: fix a small gramatical error in a comment 2007-11-21 18:19 +0000 [r89487] Mark Michelson * main/utils.c: There existed about a 1 in 4 billion chance that reading from /dev/urandom would return LONG_MIN (1 in 9 quintillion if using 64-bit longs). Since there is no positive equivalent of LONG_MIN, the result of labs() in this case is unpredictable. This fixes that situation. (closes issue #11336, reported and patched by sperreault) 2007-11-21 16:24 +0000 [r89484] Russell Bryant * channels/chan_unistim.c: Fix some code that was supposed to ensure that a buffer was terminated, but was writing to the wrong byte. Also, remove some non-thread safe test code. (closes issue #11317) Reported by: IgorG Patches: unistim-2.patch uploaded by IgorG (license 20) - additional changes by me 2007-11-21 16:08 +0000 [r89483] Mark Michelson * main/pbx.c: I introduced a deadlock avoidance into 1.4, which I attempted to port to trunk as well. Unfortunately, since trunk uses read/write locks for the context lock, it means that I have actually *introduced* a deadlock condition since they are not recursive. Removing this change for now and will look into introducing a different one. 2007-11-21 16:07 +0000 [r89480-89482] Kevin P. Fleming * include/asterisk.h, include/asterisk/compat.h, utils/ael_main.c, utils/conf2ael.c: move these forward declarations back to asterisk.h where they belong... even though asterisk.h includes compat.h, these declarations have nothing to do with the being platform-compatible and are directly related to being part of Asterisk * channels/chan_usbradio.c: get this to actually compile... * main/pbx.c: remove some debugging code that doesn't compile on 64-bit platforms 2007-11-21 15:17 +0000 [r89478-89479] Steve Murphy * res/res_features.c: OOOps! All the debug stuff I inserted was accidentally committed. I hereby revert it. * main/hashtab.c, res/res_features.c: closes issue #11265; Thanks to snuffy for his work on neatening up the code and removing duplicated code. 2007-11-21 08:28 +0000 [r89475-89477] Luigi Rizzo * channels/gentone-ulaw.c (removed): remove this file, it is not used anywhere. * main/astmm.c: add missing paths.h * configure, include/asterisk/autoconfig.h.in, configure.ac: add check for video4linux 2007-11-21 01:09 +0000 [r89474] Steve Murphy * main/pbx.c: A free in add_pri was ultimately the source of the grief we were having with parking. This set of changes fixes that problem, and introduces some more error messages, and puts debugs into ifdefs for what could be short-term usage. Txs to Terry W. for his help, guidance, and especially patience. 2007-11-21 00:23 +0000 [r89472-89473] Luigi Rizzo * main/sha1.c, agi/eagi-test.c, utils/smsq.c, utils/hashtest2.c, main/minimime/mm.h, utils/check_expr.c: more header removal/normalization * configure, include/asterisk/autoconfig.h.in, configure.ac: X11 checks (at least some - for other platforms with unusual X11 locations you might need to add more directories) 2007-11-21 00:21 +0000 [r89470] Russell Bryant * apps/app_meetme.c, CHANGES: Merge changes from team/russell/sla_trunk_moh ... * Added the ability to specify the music on hold class used to play into the conference when there is only one member and the M option is used. * Added the ability to specify a music on hold class to play instead of ringing for the SLATrunk application. (patched by me, and tested internally) 2007-11-21 00:20 +0000 [r89469] Luigi Rizzo * makeopts.in: complete support for X11 2007-11-20 23:29 +0000 [r89467-89468] Tilghman Lesher * apps/app_meetme.c, cdr/cdr_sqlite.c, pbx/pbx_lua.c: Make trunk build again * main/say.c: Add support for new recorded character sounds Closes issue #5208 2007-11-20 23:16 +0000 [r89465-89466] Luigi Rizzo * channels/chan_unistim.c, cdr/cdr_sqlite3_custom.c, apps/app_dictate.c, apps/app_test.c, apps/app_ices.c, apps/app_followme.c, channels/chan_iax2.c, main/config.c, main/loader.c, main/cli.c, cdr/cdr_csv.c, main/channel.c, main/manager.c, pbx/pbx_spool.c, include/asterisk/compat.h, res/res_agi.c, apps/app_minivm.c, main/logger.c, main/http.c, main/app.c, main/image.c, apps/app_directory.c, main/db.c, cdr/cdr_custom.c, apps/app_adsiprog.c, apps/app_dial.c, include/asterisk/utils.h, include/asterisk.h, main/pbx.c, channels/chan_sip.c, res/res_crypto.c, include/asterisk/channel.h, res/res_monitor.c, include/asterisk/paths.h, main/file.c, apps/app_sms.c, include/asterisk/ael_structs.h, pbx/pbx_config.c, apps/app_mixmonitor.c, apps/app_chanspy.c, apps/app_voicemail.c: move asterisk/paths.h outside asterisk.h and into those files who really need it. * main/pbx.c, include/asterisk.h, main/frame.c, main/dnsmgr.c, main/threadstorage.c, main/devicestate.c, include/asterisk/_private.h (added), main/astobj2.c, main/loader.c, main/term.c, main/cli.c, main/channel.c, main/manager.c, main/logger.c, build_tools/strip_nonapi, main/event.c, main/asterisk.c, main/db.c: move internal function declarations to include/asterisk/_private.h 2007-11-20 19:29 +0000 [r89464] Russell Bryant * configure, configure.ac: i got a little carried away with commas ... 2007-11-20 19:28 +0000 [r89463] Kevin P. Fleming * include/asterisk/module.h, build_tools/make_buildopts_h, main/loader.c: switch compile-time option checking to string storage mode in this branch too 2007-11-20 19:11 +0000 [r89460] Russell Bryant * configure, configure.ac: fix the zaptel configure script check 2007-11-20 18:20 +0000 [r89459] Luigi Rizzo * acinclude.m4: the 'version' is now $7 not $6 (wait a bit before regenerating configure, i have more changes) 2007-11-20 17:59 +0000 [r89458] Mark Michelson * main/pbx.c, /: Merged revisions 89457 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89457 | mmichelson | 2007-11-20 11:50:31 -0600 (Tue, 20 Nov 2007) | 9 lines According to comments in main/pbx.c, it is essential that if we are going to lock the conlock as well as the hints lock, it must be locked in that respective order. In order to prevent a potential deadlock, we need to lock the conlock prior to locking the hints lock in ast_hint_state_changed (see the call stack example on issue #11323 for how this can happen). (closes issue #11323, reported by eelcob, suggestion for patch by eelcob, patch by me) ........ 2007-11-20 17:11 +0000 [r89454-89455] Luigi Rizzo * makeopts.in: prepare to support console_video * apps/Makefile, Makefile.moddir_rules, pbx/Makefile, res/Makefile, channels/Makefile: Fix building of modules under cygwin. After this commit we can actually load modules under windows, and we can start debugging more interesting problems related to the load order and functionality of modules. 2007-11-20 16:11 +0000 [r89453] Mark Michelson * configs/sip.conf.sample: Changed occurrences of "busy-level" to "busylevel" in sip.conf.sample in light of commit 89441. Thanks to pj for pointing out the need for this (closes issue #11307, reported by pj) 2007-11-20 15:39 +0000 [r89452] Luigi Rizzo * configure, configure.ac, acinclude.m4: add an argument for extra headers to AC_EXT_LIB_CHECK, and on passing simplify the code. Too bad that every time we need to regenerate configure... 2007-11-20 15:30 +0000 [r89451] Steve Murphy * /, doc/tex/queues-with-callback-members.tex: Merged revisions 89450 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89450 | murf | 2007-11-20 08:22:08 -0700 (Tue, 20 Nov 2007) | 1 line closes issue #11324; break statements missing in switch cases. ........ 2007-11-20 15:00 +0000 [r89449] Joshua Colp * main/translate.c: Minor documentation tweak and if an incorrect parameter is given to core show translation return the usage information. (closes issue #11316) Reported by: eliel Patches: translate.c.patch uploaded by eliel (license 64) 2007-11-20 14:54 +0000 [r89448] Luigi Rizzo * configure, acinclude.m4: comment a bit the code in acinclude.m4 There is still a lot of code to clean up there, but hopefully this should clarify what goes on in there. 2007-11-20 14:49 +0000 [r89447] Joshua Colp * channels/h323/ast_h323.cxx: Include the compatibility header file in ast_h323.cxx for compatibility reasons. (closes issue #11311) Reported by: falves11 2007-11-20 14:44 +0000 [r89444-89446] Olle Johansson * channels/chan_sip.c: Fix sip show history. Closes issue #11312 * channels/chan_sip.c: Change terminology a bit for CLI commands handling SIP channels/calls/dialogs/whatever. Closes issue #11312 2007-11-20 07:42 +0000 [r89443] Luigi Rizzo * Makefile, main/Makefile, Makefile.moddir_rules: initial makefile changes to build loadable modules under cygwin (not complete yet - still need to sort out dependecies on res_*) 2007-11-20 00:17 +0000 [r89442] Steve Murphy * main/pbx.c: Get rid of some debug messages in pbx.c 2007-11-19 23:24 +0000 [r89441] Mark Michelson * channels/chan_sip.c, CHANGES: Changed the "busy-level" option in sip.conf to "busylevel" to be more parallel with the SIPPEER() argument of the same name. The deprecation procedure is not being used here since this is a trunk-only option. (closes issue #11307, reported by pj, patched by me) 2007-11-19 23:03 +0000 [r89439-89440] Russell Bryant * include/asterisk/module.h: Be a bit more pedantic about the type for holding the md5 sum for the build options. Also, doxygenify the comment. * funcs/func_sysinfo.c: Make the SYSINFO documentation reflect which options were compiled in 2007-11-19 22:55 +0000 [r89438] Steve Murphy * main/pbx.c: These changes were made in response to niklas@tese.se's letter of 11-17-2007, where he had 20 and 201 in two different contexts, included in the same context. In that particular case, we were behaving the same as 1.4, but after experimenting, I quickly found that if 20 and 201 were in the same extension, 1.4 would return 201, and this code returns 20. These changes now enable the current code to replicate the behavior of 1.4 in respect to MATCHMORE in cases like this. 2007-11-19 21:18 +0000 [r89430-89433] Luigi Rizzo * channels/chan_vpb.cc, channels/misdn_config.c, main/dsp.c: another few errno.h removals * pbx/pbx_loopback.c, apps/app_zapbarge.c, pbx/pbx_spool.c, apps/app_meetme.c, pbx/pbx_ael.c, pbx/pbx_lua.c, pbx/pbx_realtime.c, pbx/pbx_dundi.c, apps/app_externalivr.c, apps/app_directory.c, apps/app_system.c, pbx/pbx_config.c, apps/app_milliwatt.c: more errno.h removal * funcs/func_sysinfo.c: remove unnecessary headers * funcs/func_base64.c, funcs/func_volume.c: remove some unnecessary includes. 2007-11-19 20:13 +0000 [r89429] Tilghman Lesher * channels/chan_sip.c: Change delimiter of SIPPEER to be comma (instead of pipe) and further deprecate the old ':' delimiter Reported by: pj Patch by: tilghman Closes issue #11305 2007-11-19 19:51 +0000 [r89424-89428] Luigi Rizzo * codecs/codec_lpc10.c, codecs/codec_a_mu.c, codecs/codec_g722.c, codecs/codec_adpcm.c, codecs/codec_alaw.c, codecs/codec_speex.c, codecs/codec_g726.c, codecs/codec_gsm.c, codecs/codec_ulaw.c, codecs/codec_ilbc.c, codecs/codec_zap.c: remove some useless includes from codecs * formats/format_ilbc.c, formats/format_sln.c, formats/format_vox.c, formats/format_wav.c, formats/format_pcm.c, formats/format_ogg_vorbis.c, formats/format_g723.c, formats/format_h263.c, formats/format_h264.c, formats/format_wav_gsm.c, formats/format_g726.c, formats/format_jpeg.c, formats/format_gsm.c, formats/format_g729.c: format handlers don't need network, lock, channel and scheduler headers * include/asterisk.h, include/asterisk/compat.h, include/asterisk/lock.h, utils/extconf.c, include/asterisk/abstract_jb.h: move the declaration of struct ast_channel ast_frame and ast_module to compat.h so it is always available - hopefully this will let us reduce the number of inclusions of channel.h and frame.h * main/udptl.c, main/autoservice.c, funcs/func_rand.c, cdr/cdr_sqlite3_custom.c, main/frame.c, funcs/func_module.c, main/threadstorage.c, main/say.c, funcs/func_env.c, funcs/func_strings.c, main/devicestate.c, cdr/cdr_adaptive_odbc.c, main/indications.c, main/config.c, main/loader.c, main/term.c, main/cli.c, funcs/func_shell.c, main/http.c, cdr/cdr_odbc.c, main/db.c, cdr/cdr_manager.c, main/sched.c, main/pbx.c, funcs/func_timeout.c, funcs/func_math.c, funcs/func_cut.c, main/chanvars.c, main/netsock.c, funcs/func_curl.c, main/srv.c, main/privacy.c, funcs/func_cdr.c, funcs/func_channel.c, main/audiohook.c, funcs/func_iconv.c, main/alaw.c, main/asterisk.c, funcs/func_base64.c, funcs/func_md5.c, funcs/func_sysinfo.c, main/utils.c, funcs/func_sha1.c, cdr/cdr_pgsql.c, funcs/func_logic.c, cdr/cdr_radius.c, main/enum.c, funcs/func_uri.c, main/io.c, cdr/cdr_csv.c, main/ulaw.c, main/channel.c, main/cdr.c, funcs/func_enum.c, main/dial.c, funcs/func_groupcount.c, main/manager.c, main/tdd.c, funcs/func_odbc.c, cdr/cdr_sqlite.c, main/logger.c, main/app.c, main/image.c, main/dns.c, cdr/cdr_custom.c, funcs/func_version.c, funcs/func_db.c, main/dnsmgr.c, main/translate.c, main/slinfactory.c, funcs/func_lock.c, main/acl.c, main/rtp.c, cdr/cdr_tds.c, funcs/func_realtime.c, main/hashtab.c, funcs/func_blacklist.c, main/abstract_jb.c, main/cryptostub.c, main/adsistub.c, main/file.c, main/callerid.c, main/astmm.c, funcs/func_callerid.c, main/dsp.c: another bunch of include removals (errno.h and asterisk/logger.h) * channels/chan_local.c, apps/app_record.c, apps/app_alarmreceiver.c, apps/app_chanisavail.c, apps/app_ices.c, apps/app_exec.c, channels/chan_iax2.c, channels/chan_skinny.c, formats/format_pcm.c, apps/app_dumpchan.c, apps/app_zapras.c, formats/format_h263.c, codecs/codec_g722.c, formats/format_wav.c, apps/app_softhangup.c, codecs/codec_g726.c, formats/format_ogg_vorbis.c, apps/app_morsecode.c, apps/app_talkdetect.c, apps/app_db.c, apps/app_speech_utils.c, apps/app_sendtext.c, formats/format_g726.c, apps/app_mixmonitor.c, res/res_odbc.c, apps/app_voicemail.c, channels/chan_vpb.cc, formats/format_sln.c, res/res_snmp.c, apps/app_dictate.c, apps/app_authenticate.c, apps/app_readexten.c, codecs/codec_gsm.c, apps/app_userevent.c, channels/chan_gtalk.c, res/res_jabber.c, apps/app_setcallerid.c, res/res_config_odbc.c, apps/app_osplookup.c, apps/app_mp3.c, apps/app_minivm.c, res/res_realtime.c, formats/format_h264.c, apps/app_directory.c, apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c, codecs/codec_lpc10.c, res/res_config_pgsql.c, apps/app_read.c, channels/chan_sip.c, codecs/codec_alaw.c, res/res_adsi.c, res/res_crypto.c, channels/chan_jingle.c, apps/app_channelredirect.c, apps/app_forkcdr.c, formats/format_vox.c, apps/app_sms.c, formats/format_g723.c, apps/app_verbose.c, apps/app_stack.c, apps/app_readfile.c, res/res_features.c, codecs/codec_adpcm.c, apps/app_sayunixtime.c, apps/app_test.c, apps/app_image.c, formats/format_wav_gsm.c, res/res_smdi.c, include/asterisk/compat.h, apps/app_skel.c, apps/app_zapscan.c, channels/chan_alsa.c, apps/app_url.c, apps/app_externalivr.c, formats/format_jpeg.c, formats/format_gsm.c, apps/app_milliwatt.c, apps/app_dial.c, apps/app_page.c, apps/app_privacy.c, codecs/codec_speex.c, apps/app_echo.c, channels/chan_agent.c, apps/app_disa.c, channels/iax2-provision.c, res/res_ael_share.c, apps/app_transfer.c, res/res_monitor.c, apps/app_playback.c, channels/chan_misdn.c, apps/app_waitforring.c, apps/app_zapbarge.c, channels/chan_features.c, apps/app_macro.c, apps/app_zapateller.c, res/res_indications.c, codecs/codec_ilbc.c, apps/app_chanspy.c, channels/chan_unistim.c, apps/app_meetme.c, res/res_musiconhold.c, apps/app_followme.c, codecs/codec_zap.c, res/res_config_sqlite.c, channels/misdn_config.c, apps/app_controlplayback.c, formats/format_ilbc.c, channels/chan_phone.c, res/res_agi.c, main/logger.c, apps/app_ivrdemo.c, apps/app_parkandannounce.c, res/res_clioriginate.c, apps/app_while.c, include/asterisk.h, apps/app_nbscat.c, channels/chan_zap.c, codecs/codec_a_mu.c, res/res_limit.c, apps/app_festival.c, apps/app_waitforsilence.c, res/res_convert.c, apps/app_getcpeid.c, apps/app_system.c, apps/app_queue.c, channels/chan_oss.c, channels/chan_usbradio.c, apps/app_flash.c, apps/app_directed_pickup.c, channels/chan_h323.c, codecs/codec_ulaw.c, channels/chan_nbs.c, apps/app_senddtmf.c, formats/format_g729.c: include "logger.h" and errno.h from asterisk.h - usage shows that they were included almost everywhere. Remove some of the instances. 2007-11-19 17:18 +0000 [r89422] Steve Murphy * main/pbx.c: a correction to code involved in an extension removal 2007-11-19 16:29 +0000 [r89421] Mark Michelson * funcs/func_sysinfo.c (added), CHANGES: Adding SYSINFO() dialplan function for retrieval of system information 2007-11-19 15:55 +0000 [r89417-89420] Joshua Colp * /, res/res_features.c: Merged revisions 89419 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89419 | file | 2007-11-19 11:53:32 -0400 (Mon, 19 Nov 2007) | 6 lines Print out the correct filename (features.conf) in the log message when parkpos options are incorrect. (closes issue #11295) Reported by: Laureano Patches: res_features.c.patch uploaded by Laureano (license 265) ........ * /, doc/tex/localchannel.tex: Merged revisions 89416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89416 | file | 2007-11-19 11:24:12 -0400 (Mon, 19 Nov 2007) | 4 lines Clarify documentation a bit, include that a frame has to pass through the core in order for the Local channel optimization to happen. (closes issue #11246) Reported by: jon ........ 2007-11-19 14:36 +0000 [r89412] Luigi Rizzo * include/asterisk/logger.h: revert inclusion of options.h 2007-11-19 14:03 +0000 [r89410] Joshua Colp * apps/app_playback.c: Change warning messages (which are really debug messages) into debug messages. (closes issue #11288) Reported by: IgorG Patches: saydebug-89394-1-trunk.patch uploaded by IgorG (license 20) 2007-11-19 09:16 +0000 [r89404-89407] Olle Johansson * CHANGES: Update CHANGES * channels/chan_sip.c: Adding busy-level to the SIP_PEER() dialplan function. With this, you can control the peer in the dialplan, so you avoid placing outbound calls when the device has reached busy-level. Reported by pj. Closes bug #11180 * main/acl.c: Add some debugging to the routines that finds our local IP address. Related to bug #9225 * channels/chan_sip.c: Make some notes about a problem I found with the OPTIONs handler while working with the bug tracker. Since we don't authenticate devices (peers/users) on OPTIONS we don't have the proper context set for the user/peer. However, we might not want to process an authentication for every OPTIONS, so we could have a config option for this, "optionsforceok" to always answer 200 OK on the request and not check device or destination, nor add a SDP. If Asterisk sends the OPTIONs request, it doesn't care about the reply. Some devices use OPTIONs to discover capabilities, since we should answer like an INVITE from the device and we need to support that properly too, which we don't today. So much to do :-) 2007-11-18 21:50 +0000 [r89394-89399] Joshua Colp * build_tools/make_buildopts_h: Add OSX into the logic that uses md5 instead of md5sum. * include/asterisk/compat.h: Use the easy way that rizzo mentioned, only include malloc.h on the Windows platform. * include/asterisk/compat.h: Revert last commit, apparently buildbot lied to me. * include/asterisk/compat.h: Change how we handle alloca to conform with how it is suggested in the autoconf manual for AC_FUNC_ALLOCA. FreeBSD 6 now builds again and no other platforms should be broken by this. * configure, configure.ac: Change autoconf logic a bit so it says what it is looking for in two instances where it didn't. * configure, include/asterisk/autoconfig.h.in, configure.ac, include/asterisk/lock.h, include/asterisk/network.h: Use autoconf logic to determine the presence of PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP and PTHREAD_MUTEX_RECURSIVE_NP. Enclose error message from network.h in " 2007-11-17 21:47 +0000 [r89393] Matthew Fredrickson * channels/chan_zap.c: Add SS7 Generic address support (#11156) 2007-11-17 19:29 +0000 [r89389-89392] Luigi Rizzo * include/asterisk/compat.h: if alloca.h is not present, try malloc.h * agi/Makefile: temporarily disable this target in mingw * Makefile: will i ever get precedences for windows right ? in the meantime, use a variable to ease enabling/disabling print subdirectories. * Makefile: reformulate dependencies in a more correct way 2007-11-17 17:46 +0000 [r89388] Steve Murphy * main/pbx.c, pbx/pbx_dundi.c: a quick fix to pbx_dundi.c to make it so it will compile. Hope I did the right thing. And some additions to removal of extens to take care of hashtab pointers in all cases. 2007-11-17 17:27 +0000 [r89363-89387] Luigi Rizzo * Makefile.moddir_rules, Makefile.rules: as discussed some time ago on the -dev list, create embedde object with a .eo suffix even if they are coming from .cc sources. This simplifies the handling in the build scripts. * include/asterisk/network.h: prefer socket.h over other variants (winsock etc.) * channels/chan_local.c, main/translate.c, channels/chan_features.c, main/http.c, main/config.c: trim more redundant headers * main/acl.c: remove unnecessary includes * main/udptl.c, main/dnsmgr.c, channels/chan_sip.c, main/acl.c, main/dns.c, main/rtp.c, main/netsock.c: fix breakage induced by previous mistake * Makefile: wrong variable, wrong order -> broken build. * include/asterisk/acl.h, include/asterisk/utils.h, include/asterisk/autoconfig.h.in, include/asterisk/rtp.h, configure.ac, main/acl.c, include/asterisk/netsock.h, main/utils.c, include/asterisk/manager.h, main/netsock.c, main/manager.c, res/res_agi.c, pbx/pbx_dundi.c, include/asterisk/udptl.h, include/asterisk/dnsmgr.h, main/asterisk.c: start using asterisk/network.h for network related headers. Also remove some unnecessary includes. * include/asterisk/network.h (added): wrapper for all generic network headers that have different names and locations on the various systems. * main/cygload.c: main is called main not amain! * main/Makefile: conditional targets for building the windows version * Makefile: support cygwin targets * Makefile.moddir_rules: and this is the last one to have asterisk compile (not run yet) natively under cygwin. * apps/app_sms.c: another cygwin compatibility fix. This one must be handled in a better way in configure, also for other architectures * utils/Makefile, main/Makefile, utils/extconf.c: more cygwin/mingw32 compatibility fixes * include/asterisk/channel.h: use autoconf results to conditionally compile timersub * include/asterisk/lock.h: compatibility fixes for cygwin * include/asterisk/compat.h: some version of flex produce code that wants __STDC_VERSION__ defined, but the compiler does not always define it. * Makefile: these linker flags apply to both cygwin and mingw32 * utils/hashtest2.c: add a return NULL to a function that is expected to return a value so compilers that don't understand that this code is NOTREACHED will not complain (the fault is not much on the compiler but on the declaration of pthread_exit on certain platforms) s/certain platform/cygwin/ if you are really curious * main/loader.c: define RTLD_LOCAL for platforms that don't have it. This is only to complete the build, clearly the linker behaviour will be completely different and likely to cause trouble in those cases. * channels/Makefile: filter out modules that do not compile under windows (this should be handled with the dependencies generated by configure and menuselect, but will be fixed later) * main/utils.c: netdb.h is used for gethostbyname, and it was not included in some platforms. * main/cygload.c (added): Loader for cygwin where asterisk is really a big dll (something like this is already in 1.2) * configure, include/asterisk/autoconfig.h.in, configure.ac: timersub is a macro not a function, so write the check in a way that detects both formats. 2007-11-17 06:34 +0000 [r89359-89362] Russell Bryant * pbx/pbx_lua.c: fix the build of pbx_lua * configure, include/asterisk/autoconfig.h.in, include/asterisk/compat.h, configure.ac, include/asterisk/io.h, include/asterisk/channel.h: Update the configure script check for sys/poll.h to also provide the result in include/asterisk/autoconfig.h. Also, move the conditional include of sys/poll.h or asterisk/poll-compat.h into asterisk/config.h instead of the two headers it existed in before. * build_tools/make_buildopts_h: actually let this compile, oops :( * build_tools/make_buildopts_h: Use the fix suggested by Tilghman on the -dev to make cutting up the BUILDSUM friendly to non-bash shells. I think this should work for BSD/mingw as well, but did not yet remove the switch statement. 2007-11-17 04:19 +0000 [r89348-89358] Luigi Rizzo * Makefile: linker flags for mingw32 * configure, include/asterisk/autoconfig.h.in, configure.ac: add detection for timersub() and winsock.h/winsock2.h * include/asterisk/endian.h: provide definitions for __LITTLE_ENDIAN and __BIG_ENDIAN if not present. * main/Makefile, include/asterisk/io.h, include/asterisk/channel.h: use poll as detected by configure * configure, configure.ac, makeopts.in: use autoconf to check for the existence of sys/poll.h * build_tools/make_buildopts_h: this script is run on the build system, not on the host. * Makefile.moddir_rules: compatibility fix for mingw32 * configure, include/asterisk/autoconfig.h.in, configure.ac, acinclude.m4, makeopts.in: acinclude.m4: add a function to help checking sdl-config, gtk-config and the like (this could be used for gtk and gtk2 as well) Other files: add tests for sdl, sdl_image and avcodec and regenerate configure and autoconfig.h.in * include/asterisk/autoconfig.h.in, configure.ac: add check for the presence of glob * channels/chan_jingle.c, channels/chan_unistim.c, funcs/func_enum.c, channels/chan_local.c, channels/chan_misdn.c, channels/chan_skinny.c, funcs/func_odbc.c, channels/chan_h323.c, utils/ael_main.c, cdr/cdr_pgsql.c, channels/chan_gtalk.c, apps/app_db.c, channels/chan_mgcp.c: more removal of duplicate #include lines * main/udptl.c, funcs/func_module.c, res/res_features.c, funcs/func_lock.c, res/res_adsi.c, funcs/func_strings.c, channels/chan_agent.c, pbx/dundi-parser.c, main/rtp.c, pbx/pbx_loopback.c, funcs/func_blacklist.c, channels/chan_features.c, apps/app_dumpchan.c, res/res_agi.c, main/logger.c, pbx/pbx_realtime.c, pbx/pbx_dundi.c, apps/app_rpt.c, main/asterisk.c, apps/app_parkandannounce.c: remove a bunch of duplicate includes Reproduce with grep -r #include . | grep -v .svn | grep -v Binary | sort | uniq -c | sort -nr 2007-11-16 23:44 +0000 [r89347] Terry Wilson * res/res_features.c: Fix broken parking dial-back 2007-11-16 23:33 +0000 [r89346] Steve Murphy * main/pbx.c: My goodness, haven't handled an extension deletion. Add code to ast_context_remove_extension2() to remove an extension from the trie. Done by marking it deleted. The scoreboard won't update for it any more. Also, a couple of calls to insert hashtab had a spurious ->exten, which was removed. 2007-11-16 23:28 +0000 [r89341-89345] Luigi Rizzo * include/asterisk/paths.h, include/asterisk.h: paths are already in include/asterisk/paths.h so don't duplicate them in include/asterisk.h * include/asterisk/utils.h, include/asterisk/lock.h: whitespace only change - adjust indentation and add some comments on the content of these two files. utils.h (which is included in over 150 files) contains a lot of unrelated functions which require the inclusion of a large number of other headers. At some point we should partition its content in a better way. 2007-11-16 21:23 +0000 [r89333-89338] Luigi Rizzo * include/asterisk/logger.h: logger.h does not need options.h * include/asterisk/utils.h, channels/chan_sip.c, include/asterisk/astobj.h, include/asterisk/compat.h, include/asterisk/channel.h, include/asterisk/strings.h, utils/extconf.c, include/asterisk/frame.h, include/asterisk/stringfields.h, include/asterisk/endian.h: remove redundant #include "asterisk/compat.h", but make sure that asterisk/compiler.h is included everywhere * main/acl.c, main/asterisk.c: remove duplicate headers. Properly check for netdb.h (there is actually tens of places to fix) * Makefile.rules: put back default optimization to -O6 (previously changed by mistake) * main/frame.c, main/threadstorage.c, apps/app_alarmreceiver.c, apps/app_ices.c, channels/chan_iax2.c, apps/app_exec.c, channels/chan_skinny.c, main/strcompat.c, pbx/pbx_ael.c, apps/app_zapras.c, formats/format_h263.c, cdr/cdr_odbc.c, include/asterisk/sha1.h, main/db.c, cdr/cdr_manager.c, main/pbx.c, funcs/func_timeout.c, formats/format_wav.c, apps/app_softhangup.c, codecs/codec_g726.c, funcs/func_cut.c, apps/app_talkdetect.c, apps/app_db.c, funcs/func_channel.c, main/privacy.c, funcs/func_iconv.c, pbx/pbx_config.c, main/asterisk.c, res/res_odbc.c, include/asterisk/stringfields.h, apps/app_voicemail.c, formats/format_sln.c, apps/app_authenticate.c, apps/app_readexten.c, apps/app_userevent.c, codecs/codec_gsm.c, Makefile.rules, apps/app_setcallerid.c, include/asterisk/astmm.h, res/res_config_odbc.c, apps/app_osplookup.c, funcs/func_odbc.c, apps/app_mp3.c, formats/format_h264.c, apps/app_directory.c, main/md5.c, res/res_config_pgsql.c, main/dnsmgr.c, funcs/func_version.c, channels/chan_sip.c, funcs/func_lock.c, res/res_crypto.c, include/asterisk/cli.h, channels/chan_jingle.c, apps/app_forkcdr.c, funcs/func_blacklist.c, main/abstract_jb.c, main/file.c, apps/app_sms.c, formats/format_g723.c, main/astmm.c, apps/app_stack.c, apps/app_verbose.c, main/dsp.c, main/udptl.c, main/autoservice.c, funcs/func_module.c, codecs/codec_adpcm.c, cdr/cdr_adaptive_odbc.c, main/devicestate.c, apps/app_image.c, formats/format_wav_gsm.c, main/indications.c, pbx/pbx_loopback.c, funcs/func_shell.c, include/asterisk/compat.h, apps/app_skel.c, main/plc.c, channels/chan_alsa.c, apps/app_externalivr.c, formats/format_gsm.c, apps/app_milliwatt.c, res/res_speech.c, main/sched.c, apps/app_dial.c, apps/app_page.c, apps/app_disa.c, channels/iax2-provision.c, res/res_monitor.c, main/netsock.c, apps/app_waitforring.c, main/fixedjitterbuf.c, include/asterisk/lock.h, apps/app_chanspy.c, apps/app_cdr.c, channels/chan_unistim.c, funcs/func_base64.c, funcs/func_md5.c, apps/app_meetme.c, main/sha1.c, funcs/func_vmcount.c, res/res_musiconhold.c, cdr/cdr_radius.c, apps/app_followme.c, res/res_config_sqlite.c, main/fskmodem.c, channels/misdn_config.c, apps/app_controlplayback.c, cdr/cdr_csv.c, formats/format_ilbc.c, main/cdr.c, channels/chan_phone.c, funcs/func_enum.c, main/dial.c, main/manager.c, funcs/func_groupcount.c, cdr/cdr_sqlite.c, main/logger.c, main/image.c, apps/app_ivrdemo.c, res/res_clioriginate.c, apps/app_nbscat.c, codecs/codec_a_mu.c, channels/chan_zap.c, main/slinfactory.c, res/res_convert.c, pbx/pbx_lua.c, apps/app_queue.c, apps/app_system.c, channels/chan_oss.c, cdr/cdr_tds.c, funcs/func_realtime.c, channels/chan_usbradio.c, main/hashtab.c, apps/app_flash.c, include/asterisk/strings.h, apps/app_senddtmf.c, funcs/func_callerid.c, include/asterisk/time.h, channels/chan_local.c, funcs/func_dialgroup.c, funcs/func_env.c, apps/app_record.c, funcs/func_strings.c, apps/app_chanisavail.c, pbx/pbx_spool.c, apps/app_dumpchan.c, formats/format_pcm.c, main/http.c, main/stdtime/localtime.c, codecs/codec_g722.c, apps/app_morsecode.c, formats/format_ogg_vorbis.c, channels/iax2-parser.c, apps/app_speech_utils.c, include/asterisk/logger.h, main/srv.c, apps/app_sendtext.c, funcs/func_cdr.c, include/asterisk/md5.h, utils/hashtest2.c, utils/ael_main.c, main/audiohook.c, apps/app_mixmonitor.c, formats/format_g726.c, channels/chan_vpb.cc, apps/app_dictate.c, channels/chan_gtalk.c, funcs/func_logic.c, cdr/cdr_pgsql.c, res/res_jabber.c, funcs/func_uri.c, main/io.c, include/asterisk/abstract_jb.h, main/channel.c, apps/app_minivm.c, res/res_realtime.c, main/dns.c, apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c, codecs/codec_lpc10.c, apps/app_read.c, codecs/codec_alaw.c, res/res_adsi.c, include/asterisk/plc.h, apps/app_channelredirect.c, formats/format_vox.c, main/cryptostub.c, main/callerid.c, pbx/pbx_dundi.c, funcs/func_devstate.c, funcs/func_rand.c, apps/app_readfile.c, cdr/cdr_sqlite3_custom.c, main/say.c, res/res_features.c, apps/app_sayunixtime.c, apps/app_test.c, main/config.c, main/loader.c, main/term.c, main/cli.c, res/res_smdi.c, include/asterisk/astobj.h, apps/app_zapscan.c, apps/app_amd.c, pbx/pbx_realtime.c, apps/app_url.c, formats/format_jpeg.c, include/asterisk/utils.h, apps/app_privacy.c, codecs/codec_speex.c, apps/app_echo.c, channels/chan_agent.c, funcs/func_math.c, res/res_ael_share.c, pbx/dundi-parser.c, apps/app_transfer.c, include/asterisk/manager.h, apps/app_playback.c, main/chanvars.c, apps/app_zapbarge.c, channels/chan_misdn.c, funcs/func_curl.c, channels/chan_features.c, apps/app_macro.c, codecs/codec_ilbc.c, res/res_indications.c, apps/app_zapateller.c, main/dlfcn.c, include/asterisk/slinfactory.h, utils/hashtest.c, main/utils.c, funcs/func_sha1.c, codecs/codec_zap.c, main/enum.c, include/asterisk/file.h, main/tdd.c, funcs/func_volume.c, res/res_agi.c, main/app.c, apps/app_parkandannounce.c, cdr/cdr_custom.c, apps/app_while.c, funcs/func_db.c, res/res_limit.c, apps/app_festival.c, apps/app_waitforsilence.c, main/translate.c, include/asterisk/config.h, main/jitterbuf.c, main/acl.c, apps/app_getcpeid.c, funcs/func_global.c, main/rtp.c, funcs/func_extstate.c, apps/app_directed_pickup.c, main/adsistub.c, channels/chan_h323.c, codecs/codec_ulaw.c, main/event.c, channels/chan_nbs.c, pbx/pbx_gtkconsole.c, formats/format_g729.c: Start untangling header inclusion in a way that does not affect build times - tested, there is no measureable difference before and after this commit. In this change: use asterisk/compat.h to include a small set of system headers: inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h, stdlib.h, alloca.h, stdio.h Where available, the inclusion is conditional on HAVE_FOO_H as determined by autoconf. Normally, source files should not include any of the above system headers, and instead use either "asterisk.h" or "asterisk/compat.h" which does it better. For the time being I have left alone second-level directories (main/db1-ast, etc.). 2007-11-16 19:51 +0000 [r89331-89332] Mark Michelson * main/manager.c: Fixing a problem pointed out by Qwell * main/manager.c: Added some locks that should have been around astman_send_error, at least according to the comments. (closes issue #11258, reported and patched by eliel) 2007-11-16 19:26 +0000 [r89329-89330] Steve Murphy * main/pbx.c: This corrects a hashtab removal, given a bad argument * main/pbx.c, res/res_features.c: This fixes a problem with pattern ranges; and corrects a situation in res_features, where an extension would be created with the name Zap/51, as an example. THe / is bad because it would tend to mean that the 51 is to be cid matched. 2007-11-16 18:48 +0000 [r89328] Luigi Rizzo * build_tools/make_buildopts_h: both md5sum and variable substitutions such as ${BUILDSUM:0:8} are not available in FreeBSD. For the time being, put in a workaround so we can build the system, and wait for the result of the discussion on whether we can store the md5 as a string rather than 4 ints (if so, we won't need more complex tricks with awk or sed for splitting the md5). 1.4 will be fixed when we decide the issue. 2007-11-16 17:11 +0000 [r89327] Mark Michelson * apps/app_voicemail.c: Adding confirmation playback when forwarding voicemail messages. This will attempt to play the name(s) of the person(s) to whom you are forwarding the message prior to prompting for prepending. If no name is found, the extension is read back verbatim. (closes issue #9046, reported and patched by jaroth) 2007-11-16 16:56 +0000 [r89326] Kevin P. Fleming * /, include/asterisk/module.h, build_tools/make_buildopts_h, main/loader.c: Merged revisions 89325 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89325 | kpfleming | 2007-11-16 10:47:46 -0600 (Fri, 16 Nov 2007) | 4 lines To help combat problems where people build external modules (asterisk-addons or others) and then change the build options of the Asterisk build in a way that makes the incompatible without warning, this commit introduces an MD5 signature of the important build-time options and includes that signature into modules when they are built. When the loader loads one of these modules and notices the problem, it will emit a warning to console and refuse to initialize the module, as doing so could cause the system to be unstable or even crash. If you upgrade to this version of Asterisk, you must rebuild *all* of your modules that came from other sources before trying to run this version. If you are using Digium's G.729 binary codec module, you will need v33 or newer. ........ 2007-11-16 15:44 +0000 [r89324] Mark Michelson * /, apps/app_queue.c: Merged revisions 89323 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89323 | mmichelson | 2007-11-16 09:28:22 -0600 (Fri, 16 Nov 2007) | 5 lines Make realtime queues accessible from the QUEUE_MEMBER_COUNT function. (closes issue #11271, reported and patched by atis, with small modifications from me) ........ 2007-11-16 10:07 +0000 [r89322] Luigi Rizzo * include/asterisk/config.h, main/config.c: add a small new function to retrieve variables from a config once we have a pointer to the category. 2007-11-16 10:06 +0000 [r89321] Christian Richter * channels/chan_misdn.c: fixed #10631, about one way audio. thanks IgorG again. 2007-11-16 09:51 +0000 [r89320] Luigi Rizzo * channels/chan_oss.c: move the inner part of config file parsing to a separate function, so it can be reused in the implementation of cli commands when they have a similar syntax. 2007-11-16 08:54 +0000 [r89319] Christian Richter * channels/chan_misdn.c: fixed compilation of chan_misdn, #11269, thanks IgorG. 2007-11-15 23:50 +0000 [r89299-89312] Tilghman Lesher * main/utils.c, include/asterisk/stringfields.h: If we're going to be passing a negative value for the size of a stringfield, in order to indicate something, then using an UNSIGNED parameter is bad, mmmmmkay? * Makefile, /: Merged revisions 89302 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89302 | tilghman | 2007-11-15 12:37:38 -0600 (Thu, 15 Nov 2007) | 2 lines Start Asterisk in Debian at a more reasonable time (since zaptel is at level 20) ........ * /, channels/misdn/isdn_lib.c: Merged revisions 89301 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89301 | tilghman | 2007-11-15 12:23:14 -0600 (Thu, 15 Nov 2007) | 2 lines Fix an uninitialized memory read found by valgrind ........ * apps/app_zapscan.c: Fix trunk breakage due to chan->lock being renamed. * /, channels/chan_iax2.c: Merged revisions 89298 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89298 | tilghman | 2007-11-15 12:05:56 -0600 (Thu, 15 Nov 2007) | 5 lines Yet another memory corruption issue. Reported by: atis Patch by: tilghman Fixes issue #10923 ........ 2007-11-15 17:27 +0000 [r89297] Russell Bryant * /, apps/app_meetme.c: Merged revisions 89296 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89296 | russell | 2007-11-15 11:19:28 -0600 (Thu, 15 Nov 2007) | 8 lines Update the SLAStation application to account for the case where the SLA thread has a call out to the station, but the user has pressed a line button to answer the call instead of picking up the handset. If they do, the phone sends out a new INVITE. So, the SLAStation app must check to see if it is picking up a ringing trunk, and ensure that the other stations stop ringing. (reported internally, patched by me, tested by mogorman) ........ 2007-11-15 16:50 +0000 [r89294-89295] Steve Murphy * main/pbx.c: Get rid of a previously missed ast_log call for debug, no longer nec. * main/pbx.c: Perhaps I went overboard on initializing things. I can remove unnecc. stuff later. A few bug fixes. Killing small bugs on the way to killing bigger ones. Removed locking on hashtabs; there's plenty of locks already being taken. A small bug in the root_tree hashtab compare func. 2007-11-15 16:20 +0000 [r89293] Luigi Rizzo * main/channel.c, apps/app_channelredirect.c, main/manager.c, res/res_features.c, apps/app_softhangup.c, include/asterisk/channel.h, include/asterisk/lock.h, apps/app_senddtmf.c: access channel locks through ast_channel_lock/unlock/trylock and not through ast_mutex primitives. To detect all occurrences, I have renamed the lock field in struct ast_channel so it is clear that it shouldn't be used directly. There are some uses in res/res_features.c (see details of the diff) that are error prone as they try and lock two channels without caring about the order (or without explaining why it is safe). 2007-11-15 15:39 +0000 [r89290-89291] Joshua Colp * UPGRADE.txt: Fix typo in UPGRADE.txt. 'increase' should have been used, not 'increasing'. * channels/chan_sip.c, channels/chan_h323.c, channels/misdn_config.c: And file said... let trunk build again! Accomplished by some more constification, and marking a function in chan_sip as purposely unused until it is fixed up. 2007-11-15 14:58 +0000 [r89287-89289] Mark Michelson * main/manager.c, /: Merged revisions 89288 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89288 | mmichelson | 2007-11-15 08:57:28 -0600 (Thu, 15 Nov 2007) | 3 lines Undoing previous commit since I realize it was wrong ........ * main/manager.c, /: Merged revisions 89286 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89286 | mmichelson | 2007-11-15 08:54:10 -0600 (Thu, 15 Nov 2007) | 4 lines Adding a missing mutex unlock. (closes issue 11256, reported and patched by ys) ........ 2007-11-15 12:21 +0000 [r89278-89285] Olle Johansson * channels/chan_sip.c: Always relying on the responses when crossing NAT's are not a good solution, it breaks communication. Rizzo - you need to implement a configuration option for this code. It's good, but maybe should be off by default. * /, channels/chan_sip.c: Merged revisions 89281 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89281 | oej | 2007-11-15 12:26:22 +0100 (Tor, 15 Nov 2007) | 6 lines Don't send re-invites during pending INVITE transactions. Patch by one47 - thanks! Closes issue #9305 ........ * /, channels/chan_sip.c: Merged revisions 89280 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89280 | oej | 2007-11-15 12:15:09 +0100 (Tor, 15 Nov 2007) | 5 lines Improve support for multipart messages. Code by gasparz, changes by me (mostly formatting). Thanks, gasparz! Closes issue #10947 ........ * channels/chan_sip.c: Exit early instead of deciding to exit after processing the message. * channels/chan_sip.c, configs/sip.conf.sample: Add support for application/dtmf SIP INFO dtmf handling. Yep, another way of handling DTMF in SIP. Totally undocumented, but implemented in enough devices so we have to support it. Code by sergee, small changes by oej. Closes issue #11049 2007-11-15 01:42 +0000 [r89277] Steve Murphy * main/pbx.c: Had trouble playing with parking; spent a long time trying to reason out MATCHMORE mode. made these updates and xfers on zaptel lines seem to work ok now 2007-11-15 00:01 +0000 [r89273-89276] Tilghman Lesher * /, main/app.c: Merged revisions 89275 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89275 | tilghman | 2007-11-14 17:23:58 -0600 (Wed, 14 Nov 2007) | 5 lines When a recording ends with '#', we are improperly trimming an extra 200ms from the recording. Reported by: sim Patch by: tilghman Closes issue #11247 ........ * main/channel.c: Typo * main/channel.c: Add callerid to the Hangup manager event. Reported by: outtolunc Patch by: outtolunc Closes issue #11248 2007-11-14 18:05 +0000 [r89271-89272] Steve Murphy * main/pbx.c: Rescaled the weights of the patterns to give something more independent of pattern length; and make . less likely to win. Question: which should win for 14102241145-- _1xxxxxxx. or _XXXXXXXXXXX -- right now, the pure X pattern will win. * main/pbx.c: A further problem highlighted by 11233 has been resolved; a certain combination of patterns in a certain order, led to a malformed trie, due to a ptr not being initialized in the loop. Also, some tree printing prettifications. 2007-11-14 15:13 +0000 [r89269-89270] Tilghman Lesher * channels/chan_phone.c, channels/chan_zap.c, res/res_jabber.c, res/res_config_sqlite.c, main/config.c, res/res_odbc.c: One more typo in config.c; and missed conversions due to the constifying of ast_variable_new parameters * main/config.c: Typo 2007-11-14 13:18 +0000 [r89268] Luigi Rizzo * include/asterisk/acl.h, channels/chan_sip.c, include/asterisk/config.h, channels/chan_agent.c, res/res_adsi.c, main/acl.c, pbx/dundi-parser.c, apps/app_queue.c, channels/chan_iax2.c, main/enum.c, channels/chan_oss.c, apps/app_playback.c, main/config.c, pbx/dundi-parser.h, include/asterisk/abstract_jb.h, main/manager.c, channels/chan_skinny.c, apps/app_minivm.c, main/abstract_jb.c, main/logger.c, pbx/pbx_dundi.c, apps/app_directory.c, apps/app_voicemail.c: make the 'name' and 'value' fields in ast_variable const char * This prevents modifying the strings in the stored variables, and catched a few instances where this was actually done. Given the differences between trunk and 1.4 (and the fact that this is effectively an API change) it is better to fix 1.4 independently. These are chan_sip.c::sip_register() chan_skinny.c:: near line 2847 config.c:: near line 1774 logger.c::make_components() res_adsi.c:: near line 1049 I may have missed some instances for modules that do not build here. 2007-11-14 03:22 +0000 [r89263-89266] Russell Bryant * main/hashtab.c, include/asterisk/hashtab.h: Fix up various coding guidelines issues ... - handle memory allocation failures - add an ast_ prefix to a publicly exported function - put curly braces in the right places - add a bunch of spaces where they should be be used * res/res_clioriginate.c: - Use the ARRAY_LEN macro in a couple places - return errors from load_module / unload_module * apps/app_dial.c: Use BEGIN_OPTIONS / END_OPTIONS to make the syntax highlighting in my editor happy * apps/app_queue.c: Instead of reserving 800 bytes for periodic announcements, use an array of ast_str pointers and only alloate space for the strings as needed. 2007-11-14 01:16 +0000 [r89262] Joshua Colp * main/srv.c, /: Merged revisions 89260 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89260 | file | 2007-11-13 21:15:12 -0400 (Tue, 13 Nov 2007) | 4 lines Return the proper value when the srv_callback function executes properly. (closes issue #11240) Reported by: jtodd ........ 2007-11-14 01:15 +0000 [r89261] Russell Bryant * apps/app_queue.c: Convert most of the strings in the call_queue struct to use stringfields. 2007-11-14 00:54 +0000 [r89259] Kevin P. Fleming * main/channel.c, main/pbx.c: use simpler technique for removing known entries from lists 2007-11-14 00:33 +0000 [r89258] Russell Bryant * main/image.c: - Simplify removing an item from a list - move a verbose message to after the item is added to the list - make use of the ARRAY_LEN macro in one spot 2007-11-13 23:43 +0000 [r89256-89257] Steve Murphy * main/pbx.c: This hopefully will fix the re-opened 11233. Hadn't covered the case of a context with no patterns. (blush) * main/pbx.c: closes issue #11233 -- where some fine points in the algorithm to build the tree needed to be corrected. Many thanks for the test case, jtodd 2007-11-13 21:01 +0000 [r89250-89253] Russell Bryant * include/asterisk/lock.h: This fixes a build error on my mac. It also works on my linux box. Let me know if it breaks any other platform ... * res/res_features.c: Fix a typo pointed out by outtolunc, thanks :) * channels/chan_sip.c: - Convert initialization of a struct to C99 style instead of GNU style - Fix a minor spelling error in a comment * res/res_features.c, CHANGES: Update the ParkedCall application to grab the first available parked call if no parked extension is provided as an argument. (closes issue #10803) Reported by: outtolunc Patches: res_features-parkedcall-any.diff4 uploaded by outtolunc (license 237) - modified by me to work a bit differently ... 2007-11-13 19:48 +0000 [r89249] Jason Parker * /, res/res_features.c: Merged revisions 89248 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11237) ........ r89248 | qwell | 2007-11-13 13:47:45 -0600 (Tue, 13 Nov 2007) | 7 lines Revert change from revision 67064. It is documented behavior that if a parking extension already exists while using PARKINGEXTEN, dialplan execution will continue. If blind transferring to a Park with PARKINGEXTEN, you must keep this in mind, and handle the failure yourself. Issue 11237, reported by jon. ........ 2007-11-13 17:41 +0000 [r89247] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 89246 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89246 | tilghman | 2007-11-13 11:34:11 -0600 (Tue, 13 Nov 2007) | 2 lines If we set a value for qualify, we should actually pay attention to it, instead of overriding the value ........ 2007-11-13 16:03 +0000 [r89242] Mark Michelson * /, apps/app_mixmonitor.c: Merged revisions 89241 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89241 | mmichelson | 2007-11-13 10:02:02 -0600 (Tue, 13 Nov 2007) | 5 lines Reverting commit made in revision 89205 since it is unnecessary. Thanks to Kevin for pointing this out ........ 2007-11-13 14:03 +0000 [r89240] Tilghman Lesher * /, main/utils.c: Merged revisions 89239 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89239 | tilghman | 2007-11-13 07:51:53 -0600 (Tue, 13 Nov 2007) | 4 lines Debugging is running into the 16-lock limit. Increase to avoid. (This define is only effective when debugging is turned on, so there's no effect for most installations.) ........ 2007-11-13 01:19 +0000 [r89206-89207] Mark Michelson * apps/app_mixmonitor.c: There is the potential to copy uninitialized memory into the mixmonitor->post_process string. This fix prevents that. * /, apps/app_mixmonitor.c: Merged revisions 89205 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89205 | mmichelson | 2007-11-12 18:56:46 -0600 (Mon, 12 Nov 2007) | 5 lines Some sanity checking for MixMonitor. If only 1 argument is given, then the args.options and args.post_process strings are uninitialized and could contain garbage. This change handles this situation properly by only using arguments that we have parsed. ........ 2007-11-13 00:19 +0000 [r89202-89203] Jason Parker * Makefile: oops, somebody left out the directory here... * channels/chan_unistim.c, res/res_features.c, main/ast_expr2f.c, include/asterisk/config.h, res/res_convert.c, res/res_crypto.c, pbx/pbx_lua.c, include/asterisk/cli.h, include/asterisk/pbx.h, res/res_config_sqlite.c, res/res_monitor.c, include/asterisk/stringfields.h, res/res_clioriginate.c: Doxygen fixes. Also fix a common typo I kept seeing (arguement) in various files. Closes issue #11222, patch by snuffy (with arguement > argument by me). 2007-11-12 23:33 +0000 [r89196-89201] Steve Murphy * utils/hashtest.c: Don't forget the ASTERISK_VERSION for the sake of the mtx_prof stuff. * include/asterisk/hashtab.h: Thanks to snuffy for this doxygen update to hashtab.h; closes issue #11223 * main/hashtab.c, include/asterisk/hashtab.h: Thanks to snuff-work, who brought up that these fixes might need to be made. 2007-11-12 20:48 +0000 [r89195] Jason Parker * main/pbx.c, /: Merged revisions 89194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89194 | qwell | 2007-11-12 14:46:52 -0600 (Mon, 12 Nov 2007) | 1 line Fix a typo pointed out by De_Mon on #asterisk-dev ........ 2007-11-12 20:16 +0000 [r89190] Kevin P. Fleming * utils/Makefile, utils/hashtest.c: (closes issue #11221) Reported by: eliel Patches: utils.Makefile.patch uploaded by eliel (modified by me) (license 64) 2007-11-12 18:44 +0000 [r89186] Steve Murphy * main/pbx.c, pbx/pbx_realtime.c, pbx/pbx_dundi.c, funcs/func_logic.c, apps/app_exec.c, apps/app_queue.c, apps/app_mixmonitor.c, cdr/cdr_manager.c: Based on a note in asterisk-dev by Brian Capouch, I determined I too agressive in not initializing arrays passed to pbx_substitute_variables_xxxx; I reviewed the code (again) and hopefully found every possible spot where substitute_variables is called conditionally, and made sure the char array involved was set to a null string. 2007-11-12 17:44 +0000 [r89185] Tilghman Lesher * main/channel.c, /, channels/chan_sip.c: Merged revisions 89184 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89184 | tilghman | 2007-11-12 11:29:17 -0600 (Mon, 12 Nov 2007) | 5 lines Fix two cases of memory corruption caused by background threads. Reported by: atis Patch by: tilghman Fixes issue #10923 ........ 2007-11-12 13:36 +0000 [r89178-89179] Christian Richter * channels/chan_misdn.c, /, configs/misdn.conf.sample: Merged revisions 89173 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89173 | crichter | 2007-11-12 12:26:48 +0100 (Mo, 12 Nov 2007) | 1 line if we're NT and no number was dialed and overlapdial is set, we wait for the ISDN timeout instead of starting our own timer. added a comment for the misdn.conf.sample for the overlapdial config option. ........ * channels/misdn/isdn_lib_intern.h, channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c: Merged revisions 89172 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89172 | crichter | 2007-11-12 12:23:57 +0100 (Mo, 12 Nov 2007) | 1 line added restart all interfaces Restart_Indicator, to automatically send a RESTART after the L2 of a PTP Port comes up. Also fixed some places where we have send a RELEASE without need for it. ........ 2007-11-12 13:26 +0000 [r89177] Joshua Colp * channels/chan_unistim.c, utils/hashtest.c: Fix building on FreeBSD by including/not including some headers. (closes issue #11218) Reported by: ys Patches: trunk89169.diff uploaded by ys (license 281) 2007-11-12 13:22 +0000 [r89174-89176] Christian Richter * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged revisions 89171 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89171 | crichter | 2007-11-12 12:13:13 +0100 (Mo, 12 Nov 2007) | 1 line fixed a state/event issue with overlapdial=yes when no extension matched. removed the general sending of a RELEASE_COMPLETE when we receive a RELEASE, this is done by mISDNuser/mISDN. This makes it possible to use asterisk-1.4 with mISDN trunk, but requires users of mISDN/mISDNuser-1.1.X to upgrade to at least mISDNuser-1.1.6 (when using the NT mode at all) ........ * /, channels/misdn/isdn_lib.c: Merged revisions 89170 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89170 | crichter | 2007-11-12 10:57:23 +0100 (Mo, 12 Nov 2007) | 1 line fixed the support for CW and therefore for the reject_cause option. ........ * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample, channels/misdn/isdn_lib.c, channels/misdn_config.c: Merged revisions 89169 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89169 | crichter | 2007-11-12 10:45:36 +0100 (Mo, 12 Nov 2007) | 1 line aded ntkeepcalls option, to avoid droÃpping calls when the L2 goes down on a PTP link. There are some pbx which do turn off the L1 for a very short while and restart it immediately. normally T310 should be started and after 10 seconds or so the calls should be dropped, this is a simple fix wihtout this timer. ........ 2007-11-09 18:57 +0000 [r89130-89132] Jason Parker * configs/usbradio.conf.sample (added): Add usbradio.conf.sample from branches/1.4/configs - r84162. It was mistakenly deleted in 1.4 without ever being merged to trunk. Reported by eliel on #asterisk-dev. * cdr/cdr_sqlite3_custom.c, configs/cdr_sqlite3_custom.conf (removed), configs/cdr_sqlite3_custom.conf.sample (added): Fix a few potential deadlocks in cdr_sqlite3_custom. (also rename sample config to .sample) Closes issue #11208, patch by Laureano. 2007-11-09 16:00 +0000 [r89129] Steve Murphy * res/ael/pval.c, utils/Makefile, main/pbx.c, main/hashtab.c (added), main/Makefile, utils/hashtest.c (added), pbx/pbx_ael.c, include/asterisk/hashtab.h (added), main/config.c: This is the perhaps the biggest, boldest, most daring change I've ever committed to trunk. Forgive me in advance any disruption this may cause, and please, report any problems via the bugtracker. The upside is that this can speed up large dialplans by 20 times (or more). Context, extension, and priority matching are all fairly constant-time searches. I introduce here my hashtables (hashtabs), and a regression for them. I would have used the ast_obj2 tables, but mine are resizeable, and don't need the object destruction capability. The hashtab stuff is well tested and stable. I introduce a data structure, a trie, for extension pattern matching, in which knowledge of all patterns is accumulated, and all matches can be found via a single traversal of the tree. This is per-context. The trie is formed on the first lookup attempt, and stored in the context for future lookups. Destruction routines are in place for hashtabs and the pattern match trie. You can see the contents of the pattern match trie by using the 'dialplan show' cli command when 'core set debug' has been done to put it in debug mode. The pattern tree traversal only traverses those parts of the tree that are interesting. It uses a scoreboard sort of approach to find the best match. The speed of the traversal is more a function of the length of the pattern than the number of patterns in the tree. The tree also contains the CID matching patterns. See the source code comments for details on how everything works. I believe the approach general enough that any issues that might come up involving fine points in the pattern matching algorithm, can be solved by just tweaking things. We shall see. The current pattern matcher is fairly involved, and replicating every nuance of it is difficult. If you find and report problems, I will try to resolve than as quickly as I can. The trie and hashtabs are added to the existing context and exten structs, and none of the old machinery has been removed for the sake of the multitude of functions that use them. In the future, we can (maybe) weed out the linked lists and save some space. 2007-11-08 23:53 +0000 [r89124-89126] Jason Parker * /, main/say.c: Merged revisions 89125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11203) ........ r89125 | qwell | 2007-11-08 17:52:35 -0600 (Thu, 08 Nov 2007) | 4 lines Properly say the seconds here.. Issue 11203, fix described by vma. ........ * pbx/pbx_lua.c: Add check_hangup() method to pbx_lua, which can be used to check whether it is time to hangup a channel. Closes issue #11202, patch by mnicholson 2007-11-08 22:33 +0000 [r89122-89123] Mark Michelson * apps/app_voicemail.c: app_voicemail failed to build when compiling with IMAP_STORAGE Now it does not. * main/threadstorage.c: AST_LIST_REMOVE_CURRENT takes only one argument. Thanks to snuffy for pointing this out on IRC 2007-11-08 21:27 +0000 [r89121] Joshua Colp * funcs/func_env.c: Make func_env build again. 2007-11-08 21:01 +0000 [r89120] Mark Michelson * /, channels/chan_sip.c: Merged revisions 89119 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89119 | mmichelson | 2007-11-08 15:00:08 -0600 (Thu, 08 Nov 2007) | 7 lines Rework of the commit I made yesterday to use the already built-in ast_uri_decode function as opposed to my home-rolled one. Also added comments. Thanks to oej for pointing me in the right direction ........ 2007-11-08 20:39 +0000 [r89118] Kevin P. Fleming * channels/chan_features.c: convert this code to a more efficient idiom 2007-11-08 18:49 +0000 [r89116-89117] Jason Parker * res/res_smdi.c: Change a warning to a notice. Issue #11195, patch by eliel * /, configs/cdr_adaptive_odbc.conf.sample, configs/res_odbc.conf.sample: Merged revisions 89115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11195) ........ r89115 | qwell | 2007-11-08 12:45:15 -0600 (Thu, 08 Nov 2007) | 4 lines Avoid warnings on load when using sample configuration files. Issue 11195, patch by eliel. ........ 2007-11-08 17:32 +0000 [r89113-89114] Tilghman Lesher * apps/app_readfile.c, funcs/func_env.c: Add the FILE() dialplan function and deprecate ReadFile. * channels/chan_features.c: Fix missed conversion to linkedlists macro change 2007-11-08 16:51 +0000 [r89112] Mark Michelson * /: Blocking changes from previous 1.4 commit 2007-11-08 09:21 +0000 [r89108-89110] Luigi Rizzo * apps/app_voicemail.c: use %f instead of %lf (the 'l' is ignored anyways). * main/audiohook.c: use %d and cast to int instead of %zd for size_t object, this helps portability. * channels/chan_unistim.c: initialize a variable to silence compiler. The type of warnings emitted depends on the optimization level, at the lower levels the compiler doesn't always understand what the programmer has in mind. In this case I could not understand it either. 2007-11-08 05:36 +0000 [r89106-89107] Kevin P. Fleming * main/srv.c, /: Merged revisions 89105 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89105 | kpfleming | 2007-11-08 00:26:47 -0500 (Thu, 08 Nov 2007) | 2 lines fix a glaring bug in the new SRV record handling that would cause incorrect weight sorting ........ * main/autoservice.c, main/frame.c, apps/app_meetme.c, res/res_features.c, funcs/func_strings.c, main/devicestate.c, res/res_musiconhold.c, channels/chan_iax2.c, apps/app_followme.c, codecs/codec_zap.c, res/res_jabber.c, main/indications.c, main/astobj2.c, main/config.c, main/loader.c, main/cli.c, main/cdr.c, main/channel.c, main/manager.c, res/res_agi.c, main/logger.c, main/app.c, main/image.c, res/res_speech.c, main/sched.c, main/pbx.c, main/translate.c, res/res_crypto.c, channels/chan_agent.c, utils/astman.c, apps/app_queue.c, channels/iax2-parser.c, main/srv.c, include/asterisk/linkedlists.h, main/file.c, pbx/pbx_dundi.c, main/event.c, main/audiohook.c, res/res_odbc.c, main/asterisk.c, apps/app_voicemail.c: improve linked-list macros in two ways: - the *_CURRENT macros no longer need the list head pointer argument - add AST_LIST_MOVE_CURRENT to encapsulate the remove/add operation when moving entries between lists 2007-11-08 05:00 +0000 [r89104] Tilghman Lesher * /, doc/valgrind.txt: Merged revisions 89103 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89103 | tilghman | 2007-11-07 22:55:19 -0600 (Wed, 07 Nov 2007) | 2 lines Typo ........ 2007-11-08 02:28 +0000 [r89096-89102] Joshua Colp * /, channels/chan_sip.c: Merged revisions 89101 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89101 | file | 2007-11-07 22:26:48 -0400 (Wed, 07 Nov 2007) | 4 lines Do not add a sip: to the beginning of the To URI unless needed. (closes issue #10756) Reported by: goestelecom ........ * /, channels/chan_sip.c: Merged revisions 89099 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89099 | file | 2007-11-07 21:28:56 -0400 (Wed, 07 Nov 2007) | 6 lines Improve the devicestate logic for multiple devices. If any are available then the extension is considered available. (closes issue #10164) Reported by: nic_bellamy Patches: sip-hinting-svn-branch-1.4.patch uploaded by nic (license 299) ........ * /, channels/chan_sip.c: Merged revisions 89097 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89097 | file | 2007-11-07 21:11:25 -0400 (Wed, 07 Nov 2007) | 8 lines Add support for allowing one outgoing transaction. This means if a response comes back out of order chan_sip will still handle it. I dream of a chan_sip with real transaction support. (closes issue #10946) Reported by: flefoll (closes issue #10915) Reported by: ramonpeek (closes issue #9567) Reported by: atca_pres ........ * /, channels/chan_sip.c: Merged revisions 89095 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89095 | file | 2007-11-07 19:53:25 -0400 (Wed, 07 Nov 2007) | 4 lines If callerid is configured in sip.conf use that for checking the presence of an extension in the dialplan. (closes issue #11185) Reported by: spditner ........ 2007-11-07 23:47 +0000 [r89094] Tilghman Lesher * /, apps/app_queue.c: Merged revisions 89093 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89093 | tilghman | 2007-11-07 17:39:37 -0600 (Wed, 07 Nov 2007) | 7 lines The member refcount must be incremented, to avoid using it after deallocation. A huge thanks go to lvl- for patiently providing the necessary valgrind output that was necessary to finding this problem of memory corruption. Reported by: lvl- Patch by: tilghman Closes issue #11174 ........ 2007-11-07 23:18 +0000 [r89091-89092] Mark Michelson * apps/app_voicemail.c: If imapfolder has been specified in voicemail.conf, we should not connect to INBOX... ever. It may not exist. (closes issue #11151, reported by selsky, patched by me) * /, channels/chan_sip.c: Merged revisions 89090 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89090 | mmichelson | 2007-11-07 16:40:35 -0600 (Wed, 07 Nov 2007) | 6 lines This patch makes it possible for SIP phones to dial extensions defined with '#' characters in extensions.conf AND maintain their escaped characters when forming URI's (closes issue #10681, reported by cahen, patched by me, code review by file) ........ 2007-11-07 22:09 +0000 [r89089] Steve Murphy * /, res/res_jabber.c, cdr/cdr_tds.c: Merged revisions 89088 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89088 | murf | 2007-11-07 14:40:28 -0700 (Wed, 07 Nov 2007) | 1 line In response to 10578, I just ran 1.4 thru valgrind; some of the config leakage I've already fixed, but it doesn't hurt to double check. I found and fixed leaks in res_jabber, cdr_tds, pbx_ael. Nothing major, tho. ........ 2007-11-07 17:45 +0000 [r89086] Joshua Colp * channels/h323/ast_h323.cxx: Minor change so chan_h323 builds again. 2007-11-07 13:12 +0000 [r89082-89084] Luigi Rizzo * Makefile: remove enter/exit comments when handling subdirectory. If we really want them we can remove the --no-print-directory * main/loader.c: remove a debugging message which i forgot in. * Makefile: match changes in menuselect's Makefile 2007-11-07 04:21 +0000 [r89077-89081] Tilghman Lesher * apps/app_playback.c: Suppress erroneous warnings on load. Reported by: eliel Patch by: eliel Closes issue #11177 * /, configs/extensions.ael.sample: Merged revisions 89079 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89079 | tilghman | 2007-11-06 22:07:49 -0600 (Tue, 06 Nov 2007) | 5 lines Suppress AEL warnings on load. Reported by: eliel Patch by: eliel Closes issue #11178 ........ * channels/chan_zap.c, configs/zapata.conf.sample: Provide the ability to directly manipulate the TON/NPI bits in the dialstring. Reported by: thetatag Patch by: thetatag/stevens/tilghman Closes issue #5331 * contrib/utils/eagi_proxy.c (added): Add contributed EAGI proxy, which provides FastAGI functionality for EAGI, while also buffering the audio stream. Reported by: devil_slayer Patch by: devil_slayer Closes issue #8921 2007-11-07 00:16 +0000 [r89076] Russell Bryant * main/astmm.c: Fix another CLI command so it doesn't run the real code when called for initialization. 2007-11-07 00:04 +0000 [r89075] Mark Michelson * doc/tex/imapstorage.tex: Adding documentation regarding imapfolder, imapgreetings, and greetingsfolder options in voicemail.conf (closes issue #11133, reported by selsky, patched by blitzrage) 2007-11-07 00:00 +0000 [r89073-89074] Russell Bryant * include/asterisk/agi.h, res/res_agi.c, CHANGES: Print out the channel name as a prefix to the "agi debug" output. This makes AGI debugging on busy systems much easier. (closes issue #10730) Reported by: junky Patches: agi_debug_chan.diff uploaded by junky (license 177) 20070923_10730.diff uploaded by mvanbaak (license 7) * apps/app_meetme.c, CHANGES: Added the ability to do "meetme concise" with the "meetme" CLI command. This extends the concise capabilities of this CLI command to include listing all conferences, instead of an addition to the other sub commands for the "meetme" command. (closes issue #11078) Reported by: jthomas Patches: meetme-concise.patch uploaded by jthomas (license 293) 2007-11-06 23:08 +0000 [r89072] Joshua Colp * main/pbx.c: Fix up some PBX logic that became broken. The code would exit prematurely when it should have been collecting more digits. (closes issue #11175) Reported by: pj 2007-11-06 22:51 +0000 [r89071] Tilghman Lesher * channels/chan_jingle.c, channels/chan_phone.c, codecs/codec_g722.c, main/frame.c, channels/chan_sip.c, channels/chan_skinny.c, main/translate.c, channels/chan_h323.c, main/file.c, channels/chan_gtalk.c, include/asterisk/frame.h, main/rtp.c, channels/chan_mgcp.c, include/asterisk/translate.h: Commit some cleanups to the format type code. - Remove the AST_FORMAT_MAX_* types, as these are consuming 3 out of our available 32 bits. - Add a native slin16 type, so that 16kHz codecs can translate without losing resolution. (This doesn't affect anything immediately, until another codec has wb support.) 2007-11-06 22:36 +0000 [r89070] Mark Michelson * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Adding the queue strategy wrandom (closes issue #10942, reported and patched by julianjm, documentation changes by me) 2007-11-06 22:15 +0000 [r89069] Russell Bryant * apps/app_meetme.c, doc/tex/channelvariables.tex, CHANGES: Added the S() and L() options to the MeetMe application. These are pretty much identical to the S() and L() options to Dial(). They let you set timeouts for the conference, as well as have warning sounds played to let the caller know how much time is left, and when it is running out. (closes issue #8030) Reported by: areski Patches: meetme_timeout_timelimit_v2.patch uploaded by areski (license 29) 2007-11-06 22:05 +0000 [r89068] Mark Michelson * apps/app_queue.c: Added CLI and manager commands for changing a queue member's penalty (closes issue #9374, reported and initially patched by wuwu, intermediate patch by eliel, and final patch by me) 2007-11-06 22:01 +0000 [r89067] Matthew Fredrickson * channels/chan_zap.c: Add some more locking as well as API update for libss7 for new transport types 2007-11-06 21:08 +0000 [r89062] Steve Murphy * /, main/config.c: Merged revisions 89036 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89036 | murf | 2007-11-06 10:52:50 -0700 (Tue, 06 Nov 2007) | 1 line closes issue #8786 - where the [catname](!) and [catname](othercat1,othercat2,...) notation gets dropped across a ConfigUpdate (or any other thing that would cause a config file to be written). While I was at it, I also cleaned up some of the destroy routines to free up comments, which was not being done. Made sure the new struct I introduced is also cleaned up properly at destruction time. My code handles multiple template inclusions. Many thanks to ssokol for his patch, which, while not literally used in the final merge, served as a foundation for the fix. ........ 2007-11-06 20:55 +0000 [r89057] Joshua Colp * main/channel.c: Remove native bridging check for DTMF based transfers. Thanks to the last batch of RTP changes it is no longer required for the media stream to go through Asterisk if DTMF is going over signalling. It will simply reinvite back as needed. (closes issue #11172) Reported by: ibc 2007-11-06 20:32 +0000 [r89055] Mark Michelson * res/res_features.c: Instead of trying to callback a local channel on a failed attended transfer, call the device that made the transfer instead. This makes for much smoother calling back when queues are involved. (closes issue #11155, reported by IPetrov) Tremendous thanks to Russell for pulling me out of my block I was having on this one 2007-11-06 20:22 +0000 [r89052-89054] Russell Bryant * /, res/res_musiconhold.c: Merged revisions 89053 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89053 | russell | 2007-11-06 14:18:49 -0600 (Tue, 06 Nov 2007) | 3 lines Fix init_classes() so that classes that actually do have files loaded aren't treated as empty, and immediately destroyed ... ........ * main/astmm.c: Fix the memory show allocations CLI command so that it doesn't spew out all of the current memory allocations when you start Asterisk, when the command's handler gets called for initialization. 2007-11-06 19:40 +0000 [r89051] Steve Murphy * main/ast_expr2f.c, main/ast_expr2.fl: Hoping to avoid a crash in OSX for a problem blitzrage found 2007-11-06 19:23 +0000 [r89050] Olle Johansson * main/fskmodem.c: Formatting. Illegaly using some spare spaces from Russell's space-bucket. 2007-11-06 19:16 +0000 [r89049] Tilghman Lesher * /, include/asterisk/lock.h: Merged revisions 89045 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89045 | tilghman | 2007-11-06 13:09:06 -0600 (Tue, 06 Nov 2007) | 2 lines We went to the trouble of creating a method of tracking failed trylocks, then never turned it on (oops). ........ 2007-11-06 19:10 +0000 [r89048] Olle Johansson * main/tdd.c, include/asterisk/tdd.h: Additional TDD changes (preparing for SIP changes - adding TDD support to SIP) 2007-11-06 19:10 +0000 [r89047] Jason Parker * /, codecs/codec_zap.c: Merged revisions 89046 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89046 | qwell | 2007-11-06 13:09:30 -0600 (Tue, 06 Nov 2007) | 4 lines Correctly set the total number of channels from a zaptel transcoder board. SPD-49, patch by Matthew Nicholson. ........ 2007-11-06 19:04 +0000 [r89044] Mark Michelson * apps/app_readfile.c, res/res_features.c, apps/app_sayunixtime.c, apps/app_test.c, apps/app_chanisavail.c, res/res_musiconhold.c, apps/app_exec.c, apps/app_followme.c, apps/app_minivm.c, apps/app_mp3.c, apps/app_amd.c, apps/app_while.c, main/pbx.c, apps/app_nbscat.c, channels/chan_sip.c, apps/app_festival.c, apps/app_softhangup.c, apps/app_waitforsilence.c, channels/chan_agent.c, apps/app_morsecode.c, apps/app_getcpeid.c, apps/app_playback.c, res/res_monitor.c, apps/app_speech_utils.c, apps/app_forkcdr.c, apps/app_waitforring.c, apps/app_directed_pickup.c, apps/app_macro.c, apps/app_sms.c, res/res_indications.c, apps/app_chanspy.c, apps/app_mixmonitor.c, apps/app_stack.c: "show application " changes for clarity. (closes issue #11171, reported and patched by blitzrage) Many thanks! 2007-11-06 19:04 +0000 [r89043] Olle Johansson * /, main/tdd.c: Merged revisions 89042 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89042 | oej | 2007-11-06 19:53:37 +0100 (Tis, 06 Nov 2007) | 2 lines Bug fixes to tdd support in zaptel. ........ (Small changes for trunk) 2007-11-06 18:44 +0000 [r89041] Jason Parker * channels/chan_jingle.c, include/asterisk/jabber.h, channels/chan_gtalk.c, res/res_jabber.c: Allow gtalk and jingle to use TLS connections again. Closes issue #9972 2007-11-06 18:23 +0000 [r89038] Russell Bryant * /, res/res_musiconhold.c: Merged revisions 89037 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89037 | russell | 2007-11-06 12:20:07 -0600 (Tue, 06 Nov 2007) | 11 lines If someone were to delete the files used by an existing MOH class, and then issue a reload, further use of that class could result in a crash due to dividing by zero. This set of changes fixes up some places to prevent this from happening. (closes issue #10948) Reported by: jcomellas Patches: res_musiconhold_division_by_zero.patch uploaded by jcomellas (license 282) Additional changes added by me. ........ 2007-11-06 17:10 +0000 [r89034] Joshua Colp * /, channels/chan_sip.c: Merged revisions 89032 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89032 | file | 2007-11-06 13:08:05 -0400 (Tue, 06 Nov 2007) | 4 lines Make it so that if a peer is determined to be unreachable using qualify their devicestate will report back unavailable. (closes issue #11006) Reported by: pj ........ 2007-11-06 17:05 +0000 [r89031] Luigi Rizzo * main/loader.c: Fix embedding of modules on FreeBSD: the constructor for the list of modules was run after the constructors for the embedded modules (which appended entries to the list). As a result, the list appeared empty when it was time to use it. On linux the order of execution of constructor was evidently different (it may depend on the ordering of modules in the ELF file). This is only a workaround - there may be other situations where the execution of constructors causes problems, so if we manage to find a more general solution this workaround can go away. 2007-11-06 16:29 +0000 [r88974-88995] Joshua Colp * channels/chan_zap.c, /, configs/zapata.conf.sample: Merged revisions 88994 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88994 | file | 2007-11-06 12:24:56 -0400 (Tue, 06 Nov 2007) | 6 lines Fix improbable but possible memory leaks in chan_zap. (closes issue #11166) Reported by: eliel Patches: chan_zap.c.patch uploaded by eliel (license 64) ........ * channels/chan_agent.c: Update chan_agent documentation. Change a | to , as that is now the required way. (closes issue #11167) Reported by: eliel Patches: chan_agent.c.patch uploaded by eliel (license 64) 2007-11-06 15:01 +0000 [r88973] Tilghman Lesher * channels/chan_unistim.c, configure, include/asterisk/autoconfig.h.in, configure.ac: Set up detection of IP_PKTINFO in autoconf for chan_unistim 2007-11-06 14:17 +0000 [r88932-88937] Russell Bryant * channels/chan_unistim.c: convert uses of LOG_DEBUG to use ast_debug() * channels/chan_unistim.c, configs/unistim.conf.sample: Add jitterbuffer support to chan_unistim. (closes issue #11168) Reported by: IgorG Patches: unistimjb-88863-1.patch uploaded by IgorG (license 20) * main/pbx.c, /, channels/busy.h, channels/ringtone.h, include/asterisk/pbx.h: Merged revisions 88805 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88805 | russell | 2007-11-05 16:07:54 -0600 (Mon, 05 Nov 2007) | 12 lines After seeing crashes related to channel variables, I went looking around at the ways that channel variables are handled. In general, they were not handled in a thread-safe way. The channel _must_ be locked when reading or writing from/to the channel variable list. What I have done to improve this situation is to make pbx_builtin_setvar_helper() and friends lock the channel when doing their thing. Asterisk API calls almost all lock the channel for you as necessary, but this family of functions did not. (closes issue #10923, reported by atis) (closes issue #11159, reported by 850t) ........ * /, include/asterisk/lock.h: Merged revisions 88931 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88931 | russell | 2007-11-06 07:50:15 -0600 (Tue, 06 Nov 2007) | 8 lines Remove some checks to see if locks are initialized from the non-DEBUG_THREADS versions of the lock routines. These are incorrect for a number of reasons: - It breaks the build on mac. - If there is a problem with locks not getting initialized, then the proper fix is to find that place and fix the code so that it does get initialized. - If additional debug code is needed to help find the problem areas, then this type of things should _only_ be put in the DEBUG_THREADS wrappers. ........ 2007-11-06 08:17 +0000 [r88898-88913] Luigi Rizzo * channels/Makefile: explain that the host environment must be used to build gentone; Remove unset variables, they would be misleading. * Makefile: don't export variables that can be retrieved from makeopts in child subdirs 2007-11-06 02:53 +0000 [r88863] Kevin P. Fleming * /, include/asterisk/srv.h: Merged revisions 88862 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88862 | kpfleming | 2007-11-05 20:52:05 -0600 (Mon, 05 Nov 2007) | 2 lines update comment to match the state of the code ........ 2007-11-05 23:31 +0000 [r88827] Mark Michelson * main/channel.c, /: Merged revisions 88826 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88826 | mmichelson | 2007-11-05 17:29:29 -0600 (Mon, 05 Nov 2007) | 6 lines Reworked deadlock avoidance in __ast_read. Restored audio to callback agents. (closes issue #11071, reported by callguy, patched by me, tested by callguy and Ted Brown) ........ 2007-11-05 21:36 +0000 [r88770] Luigi Rizzo * Makefile, utils/Makefile: Move AUDIO_LIBS outside the top level Makefile. This too is used only in one place. 2007-11-05 21:35 +0000 [r88769] Russell Bryant * /, channels/chan_sip.c: Merged revisions 88768 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88768 | russell | 2007-11-05 15:33:56 -0600 (Mon, 05 Nov 2007) | 8 lines When traversing the list of channel variables here in transmit_invite(), the asterisk channel must be locked, as this data may change at any time. (I have seen numerous reports of crashes related to the handling of channel variables. There are a couple of issues on the bug tracker related to it, but it has also been noted on IRC and mailing lists. So, I am finding and fixing some places where channel variables are handled improperly.) ........ 2007-11-05 21:27 +0000 [r88767] Luigi Rizzo * Makefile, main/Makefile: Move the last instance of AST_LIBS to the only place it is used, namely main/Makefile . I am unclear where decisions on the build environment (CFLAGS, LDFLAGS, LIBS and so on) should be made - right now they are split here and there. As a first step in cleaning up this situation, i am trying to at least collect all instances of each variable in one place. 2007-11-05 21:23 +0000 [r88766] Russell Bryant * /, channels/chan_sip.c: Merged revisions 88765 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88765 | russell | 2007-11-05 15:21:39 -0600 (Mon, 05 Nov 2007) | 2 lines Fix up some indentation. ........ 2007-11-05 20:50 +0000 [r88764] Luigi Rizzo * Makefile.moddir_rules: comment out an unused variable. Remove it in a few days if no problems arise. 2007-11-05 20:44 +0000 [r88710-88740] Russell Bryant * main/srv.c, /, include/asterisk/srv.h: Merged revisions 88719 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88719 | russell | 2007-11-05 14:40:01 -0600 (Mon, 05 Nov 2007) | 7 lines Merge changes from asterisk/team/kpfleming/SRV-priority-handling Previously, the SRV record support in Asterisk was broken. There was no guarantee on what record Asterisk would choose to actually use. This set of changes improves the situation by ensuring that Asterisk will choose the highest priority record. ........ * main/channel.c, /: Merged revisions 88709 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88709 | russell | 2007-11-05 14:11:04 -0600 (Mon, 05 Nov 2007) | 20 lines Merge the last bit of changes from asterisk/team/russell/readq-1.4 The issue here is that the channel frame readq handling got broken when the code was converted to use the linked list macros. It caused corruption of the list head and tail pointers. So, I fixed up the usage of the linked list macros and in passing, simplified the code. I also documented what the code is doing, as it was a bit difficult to figure out at first. This bug showed itself with crashes showing messed up head/tail pointers for the readq. However, there are a couple of crashes that aren't quite as obvious, but I think may be related. So, if your bug gets closed by this commit, but you still have a problem, please reopen or create a new bug report. (closes issue #10936) (closes issue #10595) (closes issue #10368) (closes issue #11084) (closes issue #10040) (closes issue #10840) ........ 2007-11-05 19:22 +0000 [r88675] Luigi Rizzo * Makefile: Cleanup the installation of samples, avoiding repetitions. I am preserving the behaviour on *.adsi files, i.e. overwrite anything there without making a backup. However I am not sure that this is the intended behaviour. 2007-11-05 18:52 +0000 [r88673] Joshua Colp * /, channels/chan_sip.c: Merged revisions 88671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88671 | file | 2007-11-05 14:47:13 -0400 (Mon, 05 Nov 2007) | 7 lines If a SIP channel is put on hold multiple times do not keep incrementing the onHold value. (closes issue #11085) Reported by: francesco_r Tested by: blitzrage (closes issue #10474) Reported by: acennami ........ 2007-11-05 18:22 +0000 [r88653] Tilghman Lesher * CHANGES: Change wording to that suggested by MasterYoda 2007-11-05 18:00 +0000 [r88652] Luigi Rizzo * Makefile: simplify (hopefully) the printing of $(MAKE) in aligned output. 2007-11-05 17:52 +0000 [r88651] Russell Bryant * main/channel.c, /: Merged revisions 88624 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88624 | russell | 2007-11-05 11:46:02 -0600 (Mon, 05 Nov 2007) | 5 lines Fix up datastore handling in ast_do_masquerade(). The code is intended to move any channel datastores from the old channel to the new one. However, it did not use the linked list macros properly to accomplish the task. The existing code would only work if there was only a single datastore on the old channel. ........ 2007-11-05 17:44 +0000 [r88587-88615] Luigi Rizzo * Makefile: print messages when entering/leaving a directory so we know where we are (sometimes it is obvious, sometimes it is not). * Makefile.moddir_rules: merge two rules with the same right hand; document a bit what is done here. 2007-11-05 17:21 +0000 [r88586] Jason Parker * /, channels/chan_sip.c: Merged revisions 88585 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11163) ........ r88585 | qwell | 2007-11-05 11:19:41 -0600 (Mon, 05 Nov 2007) | 4 lines Make sure we destroy the config structure on configuration failure. Issue 11163, patch by eliel. ........ 2007-11-05 17:00 +0000 [r88584] Kevin P. Fleming * Makefile.rules: use a variable name that actually indicates what it is for 2007-11-05 16:41 +0000 [r88553] Luigi Rizzo * Makefile.rules: Put extra compiler flags into a variable so they are not repeated too many times. On passing, add some comments and fix indentation a bit. On passing, i suspect that the following pattern is wrong %.eoo: %.o but in case it will be fixed in a later commit. 2007-11-05 16:30 +0000 [r88540] Tilghman Lesher * /, res/res_odbc.c: Merged revisions 88539 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88539 | tilghman | 2007-11-05 10:20:13 -0600 (Mon, 05 Nov 2007) | 4 lines Don't check used pooled connections for connection status, as it will cause issues for prepared queries. Reported by: Nick Gorham (via -dev list) Patch by: tilghman ........ 2007-11-05 15:15 +0000 [r88525] Luigi Rizzo * main/db.c: remove a cygwin-specific function remap that does not work. 2007-11-05 13:11 +0000 [r88510] Joshua Colp * channels/chan_unistim.c: Fix memory leaks and deadlocks in chan_unistim. (closes issue #11158) Reported by: eliel Patches: chan_unistim.c.patch uploaded by eliel (license 64) 2007-11-04 22:42 +0000 [r88454-88490] Luigi Rizzo * /: block merging of not-applicable patch * main/channel.c, main/pbx.c, apps/app_meetme.c, channels/chan_sip.c, res/res_features.c, main/utils.c, channels/chan_iax2.c, include/asterisk/stringfields.h: Simplify the implementation and the API for stringfields; details and examples are in include/asterisk/stringfields.h. Not applicable to older branches except for 1.4 which will receive a fix for the routines that free memory pools. 2007-11-03 14:19 +0000 [r88437] Tilghman Lesher * main/term.c: Revert commit #86119. Some users intentionally do not want colorized terminals, so this was a misfeature. 2007-11-03 04:55 +0000 [r88422] James Golovich * main/db.c: Set CLI command to the correct name. Rev 85460 introduced two 'database show' commands when this one should have been 'database showkey' 2007-11-02 22:36 +0000 [r88368-88409] Russell Bryant * channels/chan_unistim.c: fix some issues with crashing on unload, when it didn't completely load cleanly * channels/chan_unistim.c: Convert the CLI commands to the new format * pbx/pbx_lua.c: propagate the DECLINE return value back to the loader * pbx/pbx_lua.c: Don't kill asterisk if extensions.lua is not present. * main/cli.c: Show the channel unique ID in the "show channel concise" output (closes issue #11148, requested by falves11, patched by me) * channels/chan_unistim.c (added), CREDITS, configs/unistim.conf.sample (added), CHANGES, doc/unistim.txt (added): Merge the code from asterisk/team/group/chan_unistim: This introduces a new channel driver, chan_unistim, that supports the Unistim VoIP protocol for Nortel phones. The following models have been confirmed to work: i2002, i2004 and i2050. (closes issue #8864) Reported by: c_hans Patches: chan_unistim.patch uploaded by c (license 304) ustm_no_conf.diff uploaded by junky (license 177) Tested by: c_hans, dbowerman, math, junky, loloski 2007-11-02 20:51 +0000 [r88329-88367] Joshua Colp * /, channels/chan_sip.c: Merged revisions 88366 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88366 | file | 2007-11-02 17:49:45 -0300 (Fri, 02 Nov 2007) | 4 lines Make subscribecontext behave as advertised. It will now look for the presence of a hint in the given context (be it subscribecontext or context). (closes issue #10702) Reported by: slavon ........ * /, channels/chan_sip.c: Merged revisions 88328 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88328 | file | 2007-11-02 17:20:21 -0300 (Fri, 02 Nov 2007) | 6 lines If an INFO request within a dialog is received with a content length of 0 simply send back a 200 OK. It is valid to do this and the remote side is probably using it to make sure the signalling is still alive. (closes issue #5747) Reported by: chandi Patches: infofix-81430-1.patch uploaded by IgorG (license 20) ........ 2007-11-02 20:13 +0000 [r88327] Russell Bryant * doc/tex/Makefile: Fix replacing the version number when it has a '/' in it, like SVN-group-chan_unistim-r88326M-/trunk 2007-11-02 17:34 +0000 [r88287] Tilghman Lesher * pbx/pbx_lua.c: Oops, some dev-mode changes for ISO C90 2007-11-02 16:54 +0000 [r88284] Jason Parker * /, main/say.c: Merged revisions 88283 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11147) ........ r88283 | qwell | 2007-11-02 11:51:08 -0500 (Fri, 02 Nov 2007) | 4 lines We need to make sure to specify a language to ast_fileexists, otherwise it may fail for anything besides en Issue 11147, fix discovered by both citats and myself (independently), with input from Corydon76 ........ 2007-11-02 16:26 +0000 [r88209-88267] Tilghman Lesher * CHANGES: Add a few bytes on LUA * main/pbx.c, utils/build-extensions-conf.lua (added), build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, configure.ac, pbx/pbx_lua.c (added), configs/extensions.lua.sample (added), include/asterisk/pbx.h, makeopts.in: Add pbx_lua as a method of doing extensions Reported by: mnicholson Patch by: mnicholson Closes issue #11140 * main/config.c: Don't re-cache the filename, but check to see if it already exists Reported by: jamesgolovich Patch by: jamesgolovich Closes issue #11144 * /, include/asterisk/lock.h: Merged revisions 88210 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88210 | tilghman | 2007-11-02 08:03:03 -0500 (Fri, 02 Nov 2007) | 5 lines Fix build on Solaris Reported by: snuffy Patch by: ys Closes issue #11143 ........ * main/pbx.c: 'h' extension doesn't execute past first priority Reported by: dimas Patch by: dimas Closes bug #11146 2007-11-02 03:09 +0000 [r88197] Joshua Colp * cdr/cdr_odbc.c: Restore building under 64-bit platforms. 2007-11-01 23:26 +0000 [r88184] Jason Parker * channels/chan_jingle.c, configure, include/asterisk/autoconfig.h.in, configure.ac, include/asterisk/jabber.h, channels/chan_gtalk.c, makeopts.in: Remove traces of gnutls, since we no longer use/need it. 2007-11-01 23:26 +0000 [r88182-88183] Tilghman Lesher * main/pbx.c: Modify WaitExten to include an optional dialtone Closes issue #10783 * UPGRADE.txt, cdr/cdr_odbc.c: Convert cdr_odbc to use res_odbc managed connections Closes issue #10614 2007-11-01 22:26 +0000 [r88166] Steve Murphy * apps/app_dial.c, main/pbx.c, cdr/cdr_sqlite3_custom.c, funcs/func_strings.c, funcs/func_cut.c, funcs/func_logic.c, apps/app_exec.c, apps/app_queue.c, apps/app_playback.c, res/ael/pval.c, pbx/pbx_loopback.c, funcs/func_odbc.c, apps/app_minivm.c, res/res_agi.c, main/logger.c, pbx/pbx_realtime.c, apps/app_macro.c, pbx/pbx_dundi.c, utils/extconf.c, include/asterisk/pbx.h, pbx/pbx_config.c, apps/app_mixmonitor.c, apps/app_rpt.c, cdr/cdr_custom.c, cdr/cdr_manager.c: This commits the performance mods that give the priority processing engine in the pbx, a 25-30% speed boost. The two updates used, are, first, to merge the ast_exists_extension() and the ast_spawn_extension() where they are called sequentially in a loop in the code, into a slightly upgraded version of ast_spawn_extension(), with a few extra args; and, second, I modified the substitute_variables_helper_full, so it zeroes out the byte after the evaluated string instead of demanding you pre-zero the buffer; I also went thru the code and removed the code that zeroed this buffer before every call to the substitute_variables_helper_full. The first fix provides about a 9% speedup, and the second the rest. These figures come from the 'PIPS' benchmark I describe in blogs, conf. reports, etc. 2007-11-01 22:19 +0000 [r88164-88165] Jason Parker * /: Crap, accidentally copied the props. Thanks for pointing this out mvanbaak. The odds are quite high that this will break automerge on every team branch. * /, include/asterisk/jabber.h, res/res_jabber.c: Switch res_jabber to use openssl rather than gnutls. Closes issue #9972, patch by phsultan. Copied from branch at http://svn.digium.com/svn/asterisk/team/phsultan/res_jabber-openssl/ 2007-11-01 17:25 +0000 [r88117] Tilghman Lesher * /, doc/valgrind.txt (added): Merged revisions 88116 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88116 | tilghman | 2007-11-01 12:17:56 -0500 (Thu, 01 Nov 2007) | 2 lines Add some notes on using valgrind ........ 2007-11-01 16:22 +0000 [r88079] Jason Parker * channels/chan_zap.c, /: Merged revisions 88078 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88078 | qwell | 2007-11-01 11:21:22 -0500 (Thu, 01 Nov 2007) | 4 lines Make sure we set the poll fds to NULL after free()ing it. Part of issue 11017, patch by tzafrir. ........ 2007-11-01 15:56 +0000 [r88062-88077] Russell Bryant * channels/chan_sip.c, pbx/pbx_dundi.c: Change some uses of free() to ast_free(). (No functional differences.) (closes issue #11138) Reported by: eliel Patches: pbx_dundi.c.patch uploaded by eliel (license 64) chan_sip.c.patch uploaded by eliel (license 64) * utils/Makefile: Remove another copied source file on "make clean". (closes issue #11137) Reported by: IgorG Patches: addonclean-87971-1.patch uploaded by IgorG (license 20) 2007-11-01 13:30 +0000 [r88027] Joshua Colp * /, apps/app_meetme.c: Merged revisions 88026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88026 | file | 2007-11-01 10:27:37 -0300 (Thu, 01 Nov 2007) | 2 lines Fix up commit for my Zap channel with spies in Meetme fix. (thanks Tony Mountifield!) ........ 2007-11-01 06:12 +0000 [r88007-88010] Tilghman Lesher * main/utils.c: Conditionally free lock_info->thread_name to avoid a useless warning Reported by: snuffy Patch by: snuffy Closes issue #11125 * apps/app_meetme.c, channels/chan_iax2.c: Janitor: use ast_free to pair calls of ast_malloc and ast_calloc Reported by: eliel Patch by: eliel Closes issue #11135 * cdr/cdr_adaptive_odbc.c: Fix memory leak Reported by: eliel Fixed by: tilghman Closes issue #11136 2007-11-01 01:55 +0000 [r87953-87971] Joshua Colp * /, apps/app_meetme.c: Merged revisions 87970 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87970 | file | 2007-10-31 22:53:55 -0300 (Wed, 31 Oct 2007) | 4 lines If a Zap channel contains a spy or a spy is added take it out of the conference in kernel space and make it go through Asterisk so the spy gets audio from both sides. (closes issue #10060) Reported by: mparker ........ * main/pbx.c: Drop any more references to type in the Exception dialplan function. (closes issue #11134) Reported by: blitzrage Patches: exception_patch.txt uploaded by blitzrage (license 10) 2007-10-31 21:23 +0000 [r87889-87909] Jason Parker * /, res/res_jabber.c: Merged revisions 87908 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11131) ........ r87908 | qwell | 2007-10-31 16:23:11 -0500 (Wed, 31 Oct 2007) | 4 lines Make sure we free some allocated memory before returning. Issue 11131, patch by eliel. ........ * channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged revisions 87906 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11130) (closes issue #11132) ........ r87906 | qwell | 2007-10-31 16:16:20 -0500 (Wed, 31 Oct 2007) | 4 lines Don't try to allocate memory that we're just going to re-allocate later anyways. Issues 11130 and 11132, patch by eliel. ........ * formats/format_sln.c, codecs/codec_adpcm.c, codecs/codec_gsm.c, formats/format_wav_gsm.c, res/res_musiconhold.c, codecs/codec_zap.c, formats/format_ilbc.c, res/res_smdi.c, formats/format_pcm.c, formats/format_h263.c, formats/format_h264.c, formats/format_jpeg.c, formats/format_gsm.c, res/res_speech.c, res/res_clioriginate.c, codecs/codec_lpc10.c, codecs/codec_g722.c, codecs/codec_a_mu.c, formats/format_wav.c, codecs/codec_speex.c, codecs/codec_alaw.c, res/res_adsi.c, res/res_convert.c, codecs/codec_g726.c, formats/format_ogg_vorbis.c, res/res_ael_share.c, formats/format_vox.c, codecs/codec_ulaw.c, formats/format_g723.c, res/res_indications.c, codecs/codec_ilbc.c, formats/format_g726.c, formats/format_g729.c: More changes to change return values from load_module functions. (issue #11096) Patches: codec_adpcm.c.patch uploaded by moy (license 222) codec_alaw.c.patch uploaded by moy (license 222) codec_a_mu.c.patch uploaded by moy (license 222) codec_g722.c.patch uploaded by moy (license 222) codec_g726.c.diff uploaded by moy (license 222) codec_gsm.c.patch uploaded by moy (license 222) codec_ilbc.c.patch uploaded by moy (license 222) codec_lpc10.c.patch uploaded by moy (license 222) codec_speex.c.patch uploaded by moy (license 222) codec_ulaw.c.patch uploaded by moy (license 222) codec_zap.c.patch uploaded by moy (license 222) format_g723.c.patch uploaded by moy (license 222) format_g726.c.patch uploaded by moy (license 222) format_g729.c.patch uploaded by moy (license 222) format_gsm.c.patch uploaded by moy (license 222) format_h263.c.patch uploaded by moy (license 222) format_h264.c.patch uploaded by moy (license 222) format_ilbc.c.patch uploaded by moy (license 222) format_jpeg.c.patch uploaded by moy (license 222) format_ogg_vorbis.c.patch uploaded by moy (license 222) format_pcm.c.patch uploaded by moy (license 222) format_sln.c.patch uploaded by moy (license 222) format_vox.c.patch uploaded by moy (license 222) format_wav.c.patch uploaded by moy (license 222) format_wav_gsm.c.patch uploaded by moy (license 222) res_adsi.c.patch uploaded by eliel (license 64) res_ael_share.c.patch uploaded by eliel (license 64) res_clioriginate.c.patch uploaded by eliel (license 64) res_convert.c.patch uploaded by eliel (license 64) res_indications.c.patch uploaded by eliel (license 64) res_musiconhold.c.patch uploaded by eliel (license 64) res_smdi.c.patch uploaded by eliel (license 64) res_speech.c.patch uploaded by eliel (license 64) 2007-10-31 18:53 +0000 [r87888] Steve Murphy * /: Merged revisions 87849 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87849 | murf | 2007-10-31 11:49:39 -0600 (Wed, 31 Oct 2007) | 1 line closes issue #11108 -- where the 'dialplan save' cli command saves a file where the semicolon is not escaped. Fixed this; User also wanted comments to be preserved across dialplan save, but this is impossible at this point in time, because comments are not stored in the dialplan. They are 'compiled' out of extensions.conf. The only way to preserve those comments is to use the config file reader/writer that the GUI uses to allow online user edits. extensions.conf is first and foremost, a config file, and is read in by the normal config-file reading routines. Then, it is processed into a dialplan (context/exten structs). (in the case of trunk, tho, no mods needed to be made -- works OK there -- just make sure you use ',' to sep app args!) ........ 2007-10-31 18:09 +0000 [r87854] Tilghman Lesher * Makefile, /: Merged revisions 87852 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87852 | tilghman | 2007-10-31 13:03:53 -0500 (Wed, 31 Oct 2007) | 2 lines Create samples for ALL of the available options in asterisk.conf ........ 2007-10-31 18:03 +0000 [r87833-87851] Joshua Colp * apps/app_mixmonitor.c: Add volume adjustment in. * apps/app_mixmonitor.c: Restore operation of the option that only writes when the channel is bridged. * apps/app_chanspy.c: Add volume adjustment to spy audiohook in app_chanspy. 2007-10-31 16:13 +0000 [r87817] Tilghman Lesher * CREDITS: Formatting cleanups, remove obsolete contributions (modules no longer in Asterisk), and obfuscate email addresses enough to stop most spam harvesters. 2007-10-31 16:07 +0000 [r87815] Joshua Colp * include/asterisk/channel.h: Remove old whisper remnants from channel.h 2007-10-31 15:46 +0000 [r87811] Tilghman Lesher * main/pbx.c: Optimize pbx_substitute_variables 2007-10-31 04:20 +0000 [r87776] Steve Murphy * res/ael/pval.c, /: Merged revisions 87775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87775 | murf | 2007-10-30 21:51:52 -0600 (Tue, 30 Oct 2007) | 1 line Included some verbage in the check_includes func, to inform the user that included contexts that have no match in the AEL, might be OK, as AEL cannot check in the extensions.conf or the in-memory contexts, as they may not be there at the time of the check. ........ 2007-10-30 23:08 +0000 [r87724-87740] Tilghman Lesher * /, include/asterisk/lock.h: Merged revisions 87739 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87739 | tilghman | 2007-10-30 18:02:22 -0500 (Tue, 30 Oct 2007) | 5 lines Fix for uninitialized mutexes on *BSD Reported by: ys Fixed by: ys Closes issue #11116 ........ * apps/app_exec.c: If no '?' is found in the arguments, don't attempt to continue. Reported by: blitzrage Fixed by: tilghman Closes issue #11111 2007-10-30 21:22 +0000 [r87687] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 87686 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87686 | russell | 2007-10-30 16:19:09 -0500 (Tue, 30 Oct 2007) | 11 lines Merge the changes from team/russell/iax2_poke_fix and iax2-poke-fix-trunk There was a race condition related to the handling of POKEing peers. Essentially, a reference to a peer is held by the scheduler when there are pending callbacks, but the reference count didn't reflect it. So, it was possible for a peer to hit a reference count of zero and have its destructor begin to be called at the same time that the scheduler thread ran a POKE related callback. If that happened, a crash would likely occur. (closes issue #11082, closes issue #11094) ........ 2007-10-30 20:30 +0000 [r87626-87651] Jason Parker * /, channels/Makefile: Merged revisions 87650 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87650 | qwell | 2007-10-30 15:29:41 -0500 (Tue, 30 Oct 2007) | 1 line Only try to clean out h323/ if the h323/Makefile exists. ........ * main/pbx.c: Update documentation to give an example of how to use the return status of RaiseException Closes issue #11117, patch by blitzrage (yay blitzrage) 2007-10-30 17:07 +0000 [r87573-87608] Mark Michelson * main/pbx.c: The priority gets incremented after raising an exception, so the priority should be set to 0 * main/pbx.c: Jumped the gun a bit in the RaiseException app. It would always return -1 since it checked for the existence of something that will never exist. 2007-10-30 16:15 +0000 [r87572] Joshua Colp * /, res/res_features.c: Merged revisions 87571 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87571 | file | 2007-10-30 13:13:39 -0300 (Tue, 30 Oct 2007) | 4 lines Add two more checks before printing out a warning message about bridging. If either channel has hungup of course the bridge will have failed. (closes issue #10009) Reported by: dimas ........ 2007-10-30 15:47 +0000 [r87568] Jason Parker * /, main/editline/np/vis.c: Merged revisions 87567 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11113) ........ r87567 | qwell | 2007-10-30 10:45:35 -0500 (Tue, 30 Oct 2007) | 4 lines Fix build of editline on Solaris. Issue 11113, patch by snuffy. ........ 2007-10-29 22:44 +0000 [r87462-87498] Kevin P. Fleming * utils/Makefile, utils, utils/hashtest2.c: UGH... while trying to fix #10995, I found all kinds of cruft in this Makefile. It should all be gone now, and as a side effect hashtest2 now builds with --enable-dev-mode enabled without a host of errors * agi/Makefile, utils/Makefile, codecs/g722/Makefile, main/editline/Makefile.in, Makefile.moddir_rules, codecs/ilbc/Makefile, codecs/lpc10/Makefile, main/db1-ast/Makefile: clean up assembler and preprocessor files if they are here too * utils, agi, codecs, apps, cdr, codecs/ilbc, formats, funcs, codecs/lpc10, main/db1-ast, codecs/g722, main/editline, main, codecs/gsm, main/minimime, pbx, res, channels: ignore preprocessor and assembler files if they are present * Makefile, /: Merged revisions 87460 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87460 | kpfleming | 2007-10-29 17:04:29 -0500 (Mon, 29 Oct 2007) | 2 lines don't put '-pipe' into ASTCFLAGS if '-save-temps' is already there (used when debugging preprocessor issues) because the compiler will whine about each compile command ........ 2007-10-29 21:34 +0000 [r87397-87428] Russell Bryant * apps/app_meetme.c: If a caller is listen-only, then don't bother with doing talker detection. (closes issue #10911, reported by junky, patched by me) * /, main/utils.c, include/asterisk/lock.h: Merged revisions 87396 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87396 | russell | 2007-10-29 15:22:07 -0500 (Mon, 29 Oct 2007) | 5 lines Add some more details to the output of "core show locks". When a thread is waiting for a lock, this will now show the details about who currently has it locked. (inspired by issue #11100) ........ 2007-10-29 20:13 +0000 [r87395] Mark Michelson * UPGRADE.txt, apps/app_queue.c: Adding the more flexible QUEUE_MEMBER function to replace the QUEUE_MEMBER_COUNT function. A deprecation notice will be issued the first time QUEUE_MEMBER_COUNT is used. 2007-10-29 20:02 +0000 [r87394] Joshua Colp * main/rtp.c: Drop the RTCP Read too short message to debug. There are some phones out there that send a sort of keep alive packet in the RTCP that trigger this every 5 seconds. 2007-10-29 19:56 +0000 [r87393] Jason Parker * apps/app_record.c: Make sure we set flags to a 0 value before trying to use it. Pointed out by seanbright while I was debugging issue 11109. 2007-10-29 19:47 +0000 [r87392] Russell Bryant * /, main/astmm.c: Merged revisions 87373 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87373 | russell | 2007-10-29 14:21:06 -0500 (Mon, 29 Oct 2007) | 5 lines Remove a lock that doesn't make any sense. The regions lock needs to be held when traversing the list of allocated chunks so that they can be printed out to the CLI. (Thanks to eliel on #asterisk-dev for pointing this out!) ........ 2007-10-29 17:22 +0000 [r87343] Joshua Colp * /, channels/chan_sip.c: Merged revisions 87342 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87342 | file | 2007-10-29 14:20:28 -0300 (Mon, 29 Oct 2007) | 6 lines Fix issue where if both sides of the dialog cancelled the dialog at the same time chan_sip could kepe retransmitting a response for no reason. (closes issue #9566) Reported by: atca_pres Patches: bug9566.patch uploaded by oej ........ 2007-10-29 16:38 +0000 [r87295-87327] Joshua Colp * apps/app_voicemail.c: Remove duplicate stdlib.h include. (closes issue #11105) Reported by: eliel Patches: app_voicemail.c.patch uploaded by eliel (license 64) * channels/chan_misdn.c, configure, include/asterisk/autoconfig.h.in, configure.ac: Add autoconf checks for extra suppserv definitions that are not present in releases yet. chan_misdn should now build against the latest release. (closes issue #11103) Reported by: IgorG * /, main/utils.c: Merged revisions 87294 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87294 | file | 2007-10-29 11:23:49 -0300 (Mon, 29 Oct 2007) | 6 lines Fix issue with ast_unescape_semicolon going into an endless loop. (closes issue #10550) Reported by: ramonpeek Patches: unescape-85177-1.patch uploaded by IgorG (license 20) ........ 2007-10-28 14:16 +0000 [r87263-87264] Tilghman Lesher * funcs/func_dialgroup.c (added): Add a simple dialgroup function. By taking one of the simpler uses of Queue away from Queue, we simplify the lives of people who do not need all the bells and whistles. Also, this is part of the functions that people need to reimplement Queue in the dialplan, as a set of logic, rather than as a single app with hundreds of options. * /, funcs/func_odbc.c, funcs/func_strings.c, funcs/func_cut.c, funcs/func_realtime.c: Merged revisions 87262 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87262 | tilghman | 2007-10-28 08:46:55 -0500 (Sun, 28 Oct 2007) | 7 lines Add autoservice to several more functions which might delay in their responses. Also, make sure that func_odbc functions have a channel on which to set variables. Reported by russell Fixed by tilghman Closes issue #11099 ........ 2007-10-27 15:41 +0000 [r87233-87247] Russell Bryant * configure, configure.ac: Update the configure script for the last libss7 API change * funcs/func_shell.c, funcs/func_lock.c: Make sure a channel exists before attempting to start or stop channel autoservice in func_lock and func_shell. 2007-10-27 00:48 +0000 [r87231-87232] Matthew Fredrickson * channels/chan_zap.c: Add Circuit Group Queury message code * channels/chan_zap.c: Make sure we turn on the DSP when we answer the call 2007-10-26 22:21 +0000 [r87217] Mark Michelson * CHANGES: Forgot to update CHANGES when I committed the linear queue strategy. Thank you Russell, for pointing this out! 2007-10-26 21:37 +0000 [r87202] Jason Parker * channels/chan_local.c, channels/chan_zap.c, channels/chan_agent.c, channels/chan_features.c, res/res_crypto.c, res/res_realtime.c, res/res_monitor.c: Correctly use defined return values in (some) load_module functions. (issue #11096) Patches: chan_agent.c.patch uploaded by eliel (license 64) chan_local.c.patch uploaded by eliel (license 64) chan_features.c.patch uploaded by eliel (license 64) chan_zap.c.patch uploaded by eliel (license 64) res_monitor.c.patch uploaded by eliel (license 64) res_realtime.c.patch uploaded by eliel (license 64) res_crypto.c.patch uploaded by eliel (license 64) 2007-10-26 17:39 +0000 [r87187] Steve Murphy * res/ael/pval.c, /, include/asterisk/pval.h, res/ael/ael.tab.c, res/ael/ael.y, pbx/pbx_ael.c, res/ael/ael_lex.c, res/ael/ael.tab.h, utils/ael_main.c, pbx/ael/ael-test/ref.ael-test16, res/ael/ael.flex, utils/conf2ael.c, pbx/ael/ael-test/ref.ael-test19: Merged revisions 87168 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87168 | murf | 2007-10-26 10:34:02 -0600 (Fri, 26 Oct 2007) | 1 line closes issue #11086 where a user complains that references to following contexts report a problem; The problem was REALLy that he was referring to empty contexts, which were being ignored. Reporter stated that empty contexts should be OK. I checked it out against extensions.conf, and sure enough, empty contexts ARE ok. So, I removed the restriction from AEL. This, though, highlighted a problem with multiple contexts of the same name. This should be OK, also. So, I added the extend keyword to AEL, and it can preceed the 'context' keyword (mixed with 'abstract', if nec.). This will turn off the warnings in AEL if the same context name is used 2 or more times. Also, I now call ast_context_find_or_create for contexts now, instead of just ast_context_create; I did this because pbx_config does this. The 'extend' keyword thus becomes a statement of intent. AEL can now duplicate the behavior of pbx_config, ........ 2007-10-26 15:19 +0000 [r87153-87154] Mark Michelson * configs/queues.conf.sample, apps/app_queue.c: Added queue strategy "linear". This strategy is useful for those who always wish for their phones to be rung in a specific order. (closes issue #7279, reported and initially patched by diLLec, patch reworked by me) * configs/queues.conf.sample: Remove information about the roundrobin strategy from trunk's queues.conf.sample since it no longer exists 2007-10-26 14:00 +0000 [r87103-87121] Tilghman Lesher * funcs/func_curl.c, /: Merged revisions 87120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87120 | tilghman | 2007-10-26 08:54:30 -0500 (Fri, 26 Oct 2007) | 7 lines The addition of autoservice to func_curl additionally made func_curl dependent on the existence of a channel, with no real reason. This should make func_curl once again work without a channel. Reported by jmls. Fixed by tilghman. Closes issue #11090 ........ * include/asterisk/app.h, funcs/func_strings.c, funcs/func_cut.c, main/app.c: Use the same delimited character as the FILTER function in FIELDQTY and CUT. 2007-10-25 23:11 +0000 [r87070] Kevin P. Fleming * main/channel.c, /, include/asterisk/linkedlists.h: Merged revisions 87069 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87069 | kpfleming | 2007-10-25 18:03:11 -0500 (Thu, 25 Oct 2007) | 2 lines appending one list to another should leave the first list empty, and not require the user to do that ........ 2007-10-25 18:59 +0000 [r87040] Russell Bryant * apps/app_meetme.c: Add support for a muted user to request to talk. The '2' option in the user menu will adjust this status if a user is muted. The talk request status will be reflected in the CLI commands as well as the manager interface. (closes issue #9418) Reported by: imesper Patches: app_meetme_v2.patch uploaded by imesper (license 275) 2007-10-25 16:21 +0000 [r87024] Steve Murphy * main/ast_expr2.y, res/res_config_sqlite.c, main/ast_expr2.c: closes issue #11045 - each file needs to define ASTERISK_FILE_VERSION, if you are going to set MTX_PROFILE in the compiler flags; the problem was that the fixes were getting made to the generated .c file, and erased the next time someone regenerated that file from the corresponding .y or .flex file. Moral of story: keep your eyes open and make mods to the .y (or flex input file) and re-run bison (or flex) as the Makefile directs for that file, and then check in both. Also, res_config_sqlite was kinda missed, and has the same issue. 2007-10-24 21:26 +0000 [r86985] Mark Michelson * configs/queues.conf.sample, apps/app_queue.c: Adding the general option "shared_lastcall" to queues so that a member's wrapuptime may be used across multiple queues. (closes issue #9777, reported and patched by eliel) 2007-10-24 20:59 +0000 [r86983] Jason Parker * channels/chan_zap.c, /: Merged revisions 86982 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11079) ........ r86982 | qwell | 2007-10-24 15:56:47 -0500 (Wed, 24 Oct 2007) | 5 lines Correctly respect hidecalleridname configuration option. Simplify code slightly in the process. Issue 11079, reported by ddv2005 ........ 2007-10-24 13:21 +0000 [r86900-86967] Steve Murphy * pbx/ael/ael-test/ref.ael-ntest22, pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3, res/ael/ael_lex.c, pbx/ael/ael-test/ref.ael-test4, res/ael/ael.flex: closes issue #11005, where #include uses the current dir instead of the config dir (/etc/asterisk) for relative path includes for AEL * /, res/ael/ael.tab.c, res/ael/ael.y: Merged revisions 86936 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86936 | murf | 2007-10-23 22:14:28 -0600 (Tue, 23 Oct 2007) | 1 line closes issue #11037 -- unable to specify app:spec in hint arguments ........ * /, funcs/func_logic.c: Merged revisions 86902 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86902 | murf | 2007-10-23 15:18:08 -0600 (Tue, 23 Oct 2007) | 1 line closes issue #11052 -- where nothing after the ? will allow un-initialized variable values to corrupt and crash asterisk on 64-bit platforms ........ * /, main/ast_expr2f.c: Merged revisions 86880 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86880 | murf | 2007-10-23 14:20:54 -0600 (Tue, 23 Oct 2007) | 1 line This should get rid of a really, really irritating warning generated by some 64-bit platforms from libc, where free(0) is frowned upon ........ * /, main/Makefile: Merged revisions 86881 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86881 | murf | 2007-10-23 14:22:25 -0600 (Tue, 23 Oct 2007) | 1 line this update to Makefile corrects how ast_expr2f.c should be generated ........ 2007-10-22 21:37 +0000 [r86835-86839] Russell Bryant * /, include/asterisk/lock.h: Merged revisions 86836 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86836 | russell | 2007-10-22 16:36:12 -0500 (Mon, 22 Oct 2007) | 9 lines If lock tracking is not enabled, then we can not attempt to log any mutex failures. If so, we could end up in infinite recursion. The only lock that is affected by this is a mutex in astmm.c used when MALLOC_DEBUG is enabled. (closes issue #11044) Reported by: ys Patches: lock.h.diff uploaded by ys (license 281) ........ * apps/app_playback.c: Convert some spaces to tabs and make it so the CLI command is only registered once instead of 3 times. (closes issue #11053) Reported by: seanbright Patches: app_playback.patch uploaded by seanbright (license 71) 2007-10-22 20:05 +0000 [r86820] Jason Parker * main/udptl.c, channels/chan_local.c, main/frame.c, res/res_features.c, main/threadstorage.c, channels/chan_iax2.c, main/astobj2.c, main/config.c, main/cli.c, channels/chan_skinny.c, main/http.c, pbx/pbx_ael.c, channels/chan_alsa.c, main/db.c, main/pbx.c, channels/chan_agent.c, channels/iax2-provision.c, apps/app_playback.c, channels/chan_misdn.c, channels/chan_features.c, res/res_indications.c, pbx/pbx_config.c, apps/app_mixmonitor.c, main/asterisk.c, res/res_odbc.c, apps/app_voicemail.c, apps/app_meetme.c, main/utils.c, channels/chan_gtalk.c, res/res_musiconhold.c, res/res_jabber.c, codecs/codec_zap.c, res/res_config_sqlite.c, main/channel.c, main/cdr.c, apps/app_osplookup.c, main/manager.c, res/res_agi.c, apps/app_minivm.c, main/logger.c, res/res_realtime.c, main/image.c, apps/app_rpt.c, channels/chan_mgcp.c, res/res_clioriginate.c, res/res_config_pgsql.c, main/dnsmgr.c, channels/chan_zap.c, channels/chan_sip.c, res/res_limit.c, main/translate.c, res/res_convert.c, res/res_crypto.c, include/asterisk/cli.h, apps/app_queue.c, channels/chan_oss.c, main/rtp.c, channels/chan_jingle.c, channels/chan_usbradio.c, main/file.c, channels/chan_h323.c, pbx/pbx_dundi.c, main/astmm.c, funcs/func_devstate.c: Switch from AST_CLI (formerly NEW_CLI) to AST_CLI_DEFINE, since the former didn't make much sense 2007-10-22 17:40 +0000 [r86790] Tilghman Lesher * /, main/astmm.c: Merged revisions 86787 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86787 | tilghman | 2007-10-22 12:38:13 -0500 (Mon, 22 Oct 2007) | 2 lines Minor FreeBSD build fix ........ 2007-10-22 16:36 +0000 [r86755-86757] Joshua Colp * /, channels/chan_sip.c: Merged revisions 86756 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86756 | file | 2007-10-22 13:35:22 -0300 (Mon, 22 Oct 2007) | 4 lines After reading online I have confirmed that Record-Route headers should be copied to 1xx responses as well. (closes issue #10113) Reported by: makoto ........ * /, apps/app_controlplayback.c: Merged revisions 86754 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86754 | file | 2007-10-22 13:15:18 -0300 (Mon, 22 Oct 2007) | 4 lines Make sure res is a positive value before performing the check to determine whether the user stopped it or not. (closes issue #11023) Reported by: cfc ........ 2007-10-22 15:57 +0000 [r86734-86751] Russell Bryant * main/channel.c, /: Merged revisions 86750 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86750 | russell | 2007-10-22 10:52:48 -0500 (Mon, 22 Oct 2007) | 8 lines Don't leak a frame in the case that an END frame is received and the time since the BEGIN is less than that of the defined minimum DTMF duration. (closes issue #11051) Reported by: casper Patches: channel.c.86664.diff uploaded by casper (license 55) ........ * channels/chan_zap.c: There is a really fun game that you can play before committing code, and it's called "make". :) * /, include/asterisk/lock.h: Merged revisions 86726 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86726 | russell | 2007-10-22 10:43:30 -0500 (Mon, 22 Oct 2007) | 4 lines Update the static mutex initializer to include the initialization of the internal mutex used to protect the lock debugging data. (closes issue #11044, patch suggested by Ivan) ........ 2007-10-22 14:59 +0000 [r86697] Kevin P. Fleming * channels/chan_zap.c, configs/zapata.conf.sample: resetinterval defaulting to something other than 'never' doesn't seem to accomplish any good and causes problems for plenty of people... 2007-10-22 14:58 +0000 [r86696] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 86694 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86694 | mmichelson | 2007-10-22 09:48:46 -0500 (Mon, 22 Oct 2007) | 5 lines Account for the fact that sometimes headers may be terminated with \r\n instead of just \n (closes issue #11043, reported by yehavi) ........ 2007-10-22 14:56 +0000 [r86695] Kevin P. Fleming * main/loader.c: merging patches that don't compile is bad... mmkay? 2007-10-22 14:28 +0000 [r86631-86664] Joshua Colp * main/channel.c, /: Merged revisions 86663 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86663 | file | 2007-10-22 11:27:03 -0300 (Mon, 22 Oct 2007) | 6 lines Move log message to before the frame it references is freed. (closes issue #11050) Reported by: slavon Patches: channel.c.86662.diff uploaded by casper (license 55) ........ * /, pbx/pbx_dundi.c: Merged revisions 86661 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86661 | file | 2007-10-22 11:05:26 -0300 (Mon, 22 Oct 2007) | 6 lines Fix tab completion for dundi show peer. (closes issue #11041) Reported by: jsmith Patches: asterisk-dundicomplete.diff.txt uploaded by jamesgolovich (license 176) ........ * /, main/acl.c, main/loader.c: Merged revisions 86630 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86630 | file | 2007-10-22 10:33:23 -0300 (Mon, 22 Oct 2007) | 6 lines Fixes for building under OpenSolaris. (closes issue #11047) Reported by: snuffy Patches: 11047-fixes.diff uploaded by snuffy (license 35) ........ 2007-10-22 10:18 +0000 [r86616-86617] Christian Richter * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged revisions 86598 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86598 | crichter | 2007-10-22 11:21:15 +0200 (Mo, 22 Okt 2007) | 1 line we send DISCONNECT instead of RELEASE/RELEASE_COMPLETE if the dialplan does not match after an overlap call. Also added out_cause=1 ........ * channels/misdn/isdn_lib.h, channels/chan_misdn.c, channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c: started to add some basic support for supplementary services like CallForwarding and so forth 2007-10-21 22:52 +0000 [r86585] Russell Bryant * /, include/asterisk/cli.h, main/asterisk.c, main/cli.c: Merged revisions 85532 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85532 | russell | 2007-10-13 00:24:33 -0500 (Sat, 13 Oct 2007) | 8 lines Properly handle the case where read() may return the text for more than one CLI command at once for a remote console. (closes issue #10888) Reported by: jamesgolovich Patches: asterisk-climultiple.diff.txt uploaded by jamesgolovich (license 176) ........ 2007-10-20 19:56 +0000 [r86572] Matthew Fredrickson * configs/zapata.conf.sample: Improved comments and organization for zapata.conf (#10904) 2007-10-19 18:46 +0000 [r86549] Matthew Fredrickson * channels/chan_zap.c: Add better support for blocking and unblocking of CICs (#10965) 2007-10-19 18:29 +0000 [r86534-86536] Jason Parker * main/udptl.c, channels/chan_local.c, main/frame.c, res/res_features.c, main/threadstorage.c, channels/chan_iax2.c, main/astobj2.c, main/config.c, main/cli.c, channels/chan_skinny.c, main/http.c, pbx/pbx_ael.c, channels/chan_alsa.c, main/db.c, main/pbx.c, channels/chan_agent.c, channels/iax2-provision.c, apps/app_playback.c, channels/chan_misdn.c, channels/chan_features.c, res/res_indications.c, pbx/pbx_config.c, apps/app_mixmonitor.c, main/asterisk.c, res/res_odbc.c, apps/app_voicemail.c, apps/app_meetme.c, main/utils.c, channels/chan_gtalk.c, res/res_musiconhold.c, res/res_jabber.c, codecs/codec_zap.c, res/res_config_sqlite.c, main/channel.c, main/cdr.c, apps/app_osplookup.c, main/manager.c, res/res_agi.c, apps/app_minivm.c, main/logger.c, res/res_realtime.c, main/image.c, apps/app_rpt.c, channels/chan_mgcp.c, res/res_clioriginate.c, res/res_config_pgsql.c, main/dnsmgr.c, channels/chan_zap.c, channels/chan_sip.c, res/res_limit.c, main/translate.c, res/res_convert.c, res/res_crypto.c, include/asterisk/cli.h, apps/app_queue.c, channels/chan_oss.c, main/rtp.c, channels/chan_jingle.c, channels/chan_usbradio.c, main/file.c, channels/chan_h323.c, pbx/pbx_dundi.c, main/astmm.c, funcs/func_devstate.c: Convert NEW_CLI to AST_CLI. Closes issue #11039, as suggested by seanbright. * channels/chan_usbradio.c, res/res_config_pgsql.c, channels/chan_misdn.c, channels/chan_h323.c, res/res_indications.c, channels/chan_iax2.c, codecs/codec_zap.c, res/res_config_sqlite.c, main/config.c, main/rtp.c: More changes to NEW_CLI. Also fixes a few cli messages and some minor formatting. (closes issue #11001) Reported by: seanbright Patches: newcli.1.patch uploaded by seanbright (license 71) newcli.2.patch uploaded by seanbright (license 71) newcli.4.patch uploaded by seanbright (license 71) newcli.5.patch uploaded by seanbright (license 71) newcli.6.patch uploaded by seanbright (license 71) newcli.7.patch uploaded by seanbright (license 71) 2007-10-19 16:40 +0000 [r86470-86503] Joshua Colp * /, main/app.c: Merged revisions 86502 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86502 | file | 2007-10-19 13:38:29 -0300 (Fri, 19 Oct 2007) | 4 lines When returning a DTMF digit from ast_control_streamfile cast it as a char so that 0 does not overlap with the success return code. (closes issue #11023) Reported by: cfc ........ * /, channels/chan_sip.c: Merged revisions 86471 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86471 | file | 2007-10-19 12:33:49 -0300 (Fri, 19 Oct 2007) | 6 lines Fix two issues with domains and transfers. If a port was given in the hostname it was treated as part of the hostname. If domains were configured but external domains were not enabled all transfers would be considered remote. (closes issue #11027) Reported by: ramonpeek Patches: 11027-1.diff uploaded by ramonpeek (license 266) ........ * /, channels/chan_sip.c: Merged revisions 86469 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86469 | file | 2007-10-19 12:08:12 -0300 (Fri, 19 Oct 2007) | 4 lines Set port number in received as information for registrations as well. (closes issue #11028) Reported by: brad-x ........ 2007-10-19 01:56 +0000 [r86439] TransNexus OSP Development * apps/app_osplookup.c: Fixed a buffer size issue. 2007-10-18 22:03 +0000 [r86407-86408] Jason Parker * Makefile, /: Merged revisions 86405 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes issue #11029) ........ r86405 | qwell | 2007-10-18 16:58:44 -0500 (Thu, 18 Oct 2007) | 4 lines Add documentation for options in asterisk.conf Issue 11029, patch by eserra ........ 2007-10-18 18:40 +0000 [r86350] Mark Michelson * channels/chan_zap.c: Fixing a segfault from tab-completing a "zap restart" CLI command. (patch made by seanbright, pointed out in #asterisk-dev on IRC) 2007-10-18 18:06 +0000 [r86331] Russell Bryant * main/channel.c, /, include/asterisk/channel.h: Merged revisions 86330 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86330 | russell | 2007-10-18 13:03:10 -0500 (Thu, 18 Oct 2007) | 10 lines The channel needs to stay locked while running timer callbacks, as they access and modify channel data that may change elsewhere. I went through every timer callback in the source tree to make sure that none of them did any additional locking that could introduce deadlocks, and all is well. (closes issue #10765) Reported by: Ivan Patches: ast_1_4_11_svn_patch_channel_rc.diff uploaded by Ivan (license 229) ........ 2007-10-18 17:40 +0000 [r86298-86329] Mark Michelson * /, apps/app_queue.c: Merged revisions 86328 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86328 | mmichelson | 2007-10-18 12:38:26 -0500 (Thu, 18 Oct 2007) | 5 lines If a non-existent file is specified to be played either as a periodic announcement or as a hold/position announcement, the caller would be kicked out of the queue. No longer does this happen. ........ * apps/app_queue.c: Changed some spaces to tabs 2007-10-18 15:57 +0000 [r86297] Russell Bryant * /, codecs/codec_zap.c: Merged revisions 86296 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86296 | russell | 2007-10-18 10:45:55 -0500 (Thu, 18 Oct 2007) | 3 lines Execute the RELEASE operation on transcoder channels in the destroy callback. (patch from jsloan) ........ 2007-10-18 07:23 +0000 [r86277-86278] Tilghman Lesher * main/acl.c: Code cleanup of acl.c Reported by dimas Closes issue #10784 * res/res_musiconhold.c: On reload, re-read the files in the specified moh directory (closes issue #10536) 2007-10-18 04:41 +0000 [r86238] Russell Bryant * /, main/utils.c: Merged revisions 86237 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86237 | russell | 2007-10-17 23:40:52 -0500 (Wed, 17 Oct 2007) | 9 lines Revert a change that I made for issue #10979 which, as has been pointed out to me in issue #11018, doesn't really make sense. There is no reason to have the base64 decode function force a '\0' terminated buffer, when the result is almost always binary, anyway. In fact, this caused some breakage, as some code in res_crypto passed in a buffer exactly the right size to get its binary result, which got stomped on by this patch. (closes issue #11018, reported by dimas) ........ 2007-10-17 21:41 +0000 [r86208] Mark Michelson * /, apps/app_queue.c: Merged revisions 86202 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86202 | mmichelson | 2007-10-17 16:39:05 -0500 (Wed, 17 Oct 2007) | 6 lines Changing the strategy field of the call_queue struct to be signed instead of unsigned, since the code attempts to set the strategy to -1 if you specify a bogus strategy. While this isn't a huge issue in 1.4, it could be a problem for someone who, say, tries to use the roundrobin strategy in trunk (despite all the deprecation warnings in 1.4). ........ 2007-10-17 21:16 +0000 [r86195-86197] Tilghman Lesher * main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2.c: Simplify some preprocessor logic by using #elif * CHANGES, configs/meetme.conf.sample: Document the changes made earlier today to meetme 2007-10-17 20:06 +0000 [r86180-86182] Steve Murphy * utils/hashtest2.c, utils/check_expr.c, utils/clicompat.c: and then, I noticed the clicompat stuff. * utils/check_expr.c: more stub routines to allow linkage in stand-alone environment, with thread debugs turned on * utils/hashtest2.c: more stub routines to allow linkage in stand-alone environment, with thread debugs turned on 2007-10-17 18:01 +0000 [r86150] Russell Bryant * /, channels/chan_sip.c: Merged revisions 86149 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86149 | russell | 2007-10-17 12:57:45 -0500 (Wed, 17 Oct 2007) | 8 lines If Asterisk is in the middle of shutting down, respond to OPTIONS with 503 Unavailable. (closes issue #10994) Reported by: eserra Patches: sip-options-503.patch uploaded by eserra (license 45) ........ 2007-10-17 17:06 +0000 [r86119] Tilghman Lesher * main/term.c: Support color on certain platforms, even when started at boot (before TERM is set) Closes issue #9048 2007-10-17 17:00 +0000 [r86118] Joshua Colp * /, channels/chan_sip.c: Merged revisions 86117 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86117 | file | 2007-10-17 13:58:03 -0300 (Wed, 17 Oct 2007) | 4 lines Whoops, forgot to remove the original sip_scheddestroy. (closes issue #11010) Reported by: vadim ........ 2007-10-17 16:09 +0000 [r86104] Jason Parker * channels/chan_usbradio.c, channels/xpmr/xpmr.c: Allow chan_usbradio to compile again. Closes issue #11014, patch by seanbright. 2007-10-17 15:39 +0000 [r86079] Tilghman Lesher * /, main/asterisk.c: Merged revisions 86066 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86066 | tilghman | 2007-10-17 10:23:51 -0500 (Wed, 17 Oct 2007) | 3 lines When runuser/rungroup is specified, a remote console could only be attained by root (Closes issue #9999) ........ 2007-10-17 15:30 +0000 [r86067] Joshua Colp * channels/chan_usbradio.c: Change dependency for chan_usbradio to asound. Let's keep everything uniform. (closes issue #11013) Reported by: seanbright 2007-10-17 15:13 +0000 [r86065] Tilghman Lesher * apps/app_meetme.c: Enhancements to realtime (closes issue #9609) 2007-10-17 15:09 +0000 [r86064] Joshua Colp * /, channels/chan_sip.c: Merged revisions 86063 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86063 | file | 2007-10-17 12:06:36 -0300 (Wed, 17 Oct 2007) | 4 lines Don't schedule dialog destruction if a MESSAGE is received using an existing dialog. (closes issue #11010) Reported by: vadim ........ 2007-10-16 23:36 +0000 [r86029-86033] Mark Michelson * /, configs/queues.conf.sample: Merged revisions 86032 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86032 | mmichelson | 2007-10-16 18:35:31 -0500 (Tue, 16 Oct 2007) | 3 lines Since monitor-join is deprecated now, remove the example from the sample queues.conf file ........ * apps/app_queue.c: Removed the monitor-join option. If one wishes to mix audio, they should instead use monitor-type=mixmonitor. (related to issue #10885) 2007-10-16 22:36 +0000 [r85995-85998] Russell Bryant * /, include/asterisk/lock.h: Merged revisions 85997 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85997 | russell | 2007-10-16 17:36:16 -0500 (Tue, 16 Oct 2007) | 1 line really picky formatting tweak ... ........ * /, include/asterisk/lock.h: Merged revisions 85994 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85994 | russell | 2007-10-16 17:14:36 -0500 (Tue, 16 Oct 2007) | 16 lines Some locking errors exposed the fact that the lock debugging code itself was not thread safe. How ironic! Anyway, these changes ensure that the code that is accessing the lock debugging data is thread-safe. Many thanks to Ivan for finding and fixing the core issue here, and also thanks to those that tested the patch and provided test results. (closes issue #10571) (closes issue #10886) (closes issue #10875) (might close some others, as well ...) Patches: (from issue #10571) ivan_ast_1_4_12_rel_patch_lock.h.diff uploaded by Ivan (license 229) - a few small changes by me ........ 2007-10-16 21:51 +0000 [r85959-85992] Mark Michelson * apps/app_queue.c: Fixing the build. * apps/app_read.c: Fixing app_read so that if a timeout of less than 1 ms is specified, assume that 1 ms is desired. (closes issue #11000, reported and patched by michael-fig, with a warning line added by me) * /, apps/app_queue.c: Merged revisions 85958 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85958 | mmichelson | 2007-10-16 16:14:34 -0500 (Tue, 16 Oct 2007) | 5 lines Trying to remove a non-dynamic queue member via dynamic means can lead to some interesting (read nasty) situations. This patch clears up the issue by making only dynamic queue members removable via dynamic methods. ........ 2007-10-16 20:55 +0000 [r85957] Matthew Fredrickson * channels/chan_zap.c: Don't hangup the call for SS7 if we get an alarm 2007-10-16 20:32 +0000 [r85944] Russell Bryant * channels/chan_sip.c: This fixes SIP subscriptions in trunk. Don't improperly memset() over an ast_str. This was leftover from before it got changed to use ast_str. (closes issue #11003, reported by pj) (closes issue #10770, reported by yehavi) (patched by me) 2007-10-16 19:47 +0000 [r85943] Tilghman Lesher * /, main/stdtime/localtime.c: Merged revisions 85921 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85921 | tilghman | 2007-10-16 14:41:40 -0500 (Tue, 16 Oct 2007) | 4 lines Also set up gmtoff (this is used in the %z gnu extension to strftime) Reported and fixed by jcmoore Closes issue #11002 ........ 2007-10-16 19:12 +0000 [r85897] Russell Bryant * /, apps/app_voicemail.c: Merged revisions 85896 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85896 | russell | 2007-10-16 14:10:01 -0500 (Tue, 16 Oct 2007) | 2 lines Remove a pointless lock. ........ 2007-10-16 16:40 +0000 [r85853-85883] Mark Michelson * apps/app_voicemail.c: Fix IMAP compilation error. (closes issue #10986, reported and patched by snuffy) * /: Blocking changes from previous commit 2007-10-16 15:15 +0000 [r85819-85851] Joshua Colp * /, funcs/func_vmcount.c: Merged revisions 85850 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85850 | file | 2007-10-16 11:52:22 -0300 (Tue, 16 Oct 2007) | 4 lines Check to make sure a value has been given to the VMCOUNT dialplan function. (closes issue #10996) Reported by: marsosa ........ * main/threadstorage.c: Permit building under DEBUG_THREADLOCALS. Thanks snuff. * /, main/threadstorage.c: Merged revisions 85818 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85818 | file | 2007-10-16 11:19:39 -0300 (Tue, 16 Oct 2007) | 6 lines Fix memory allocation issue in threadstorage. (closes issue #10995) Reported by: snuffy Patches: new-patch.diff uploaded by snuffy (license 35) ........ 2007-10-16 10:38 +0000 [r85777-85787] Philippe Sultan * channels/chan_jingle.c, channels/chan_gtalk.c: Fix CLI help output * channels/chan_jingle.c: Added two CLI functions, taken from chan_gtalk : - jingle reload ; - jingle show channels. * channels/chan_jingle.c: Make an audio path under the following call configuration : SIP Phone 1 --- [chan_sip]Asterisk 1[chan_jingle] --- [chan_jingle]Asterisk 2[chan_sip] --- SIP Phone 2 Modifications : - set bridge type to partial ; - process media candidates from the remote peer properly. Now we have Jingle audio, at least between two Asterisk Jingle clients. 2007-10-15 23:20 +0000 [r85764] Jason Parker * configs/dundi.conf.sample, channels/chan_sip.c, channels/chan_h323.c, main/acl.c, UPGRADE.txt, channels/iax2-provision.c, doc/tex/qos.tex, pbx/pbx_dundi.c, channels/chan_iax2.c, channels/chan_mgcp.c: Switch dundi to new tos config format. Remove old unused defines for old style. Closes issue 10860, patch by IgorG. 2007-10-15 21:11 +0000 [r85718-85721] Russell Bryant * /, apps/app_queue.c: Merged revisions 85720 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85720 | russell | 2007-10-15 16:10:02 -0500 (Mon, 15 Oct 2007) | 3 lines Ensure that no pending state changes are leaked when the device state change thread gets stopped on module unload. ........ * /, main/say.c: Merged revisions 85686 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85686 | russell | 2007-10-15 15:21:27 -0500 (Mon, 15 Oct 2007) | 7 lines Add a small fix for the tw version of saying dates. (closes issue #7827) Reported by: sharkey Patches: say.nits.patch uploaded by sharkey (license 172) ........ 2007-10-15 20:16 +0000 [r85685] Jason Parker * Makefile, /: Merged revisions 85684 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #10938) ........ r85684 | qwell | 2007-10-15 15:15:51 -0500 (Mon, 15 Oct 2007) | 5 lines Properly use DESTDIR in 'config' target. Do not try to run chkconfig or similar if using DESTDIR. Issue 10938, patch by cabal95. ........ 2007-10-15 20:09 +0000 [r85648-85683] Russell Bryant * doc/tex/channelvariables.tex: add TOUCH_MONITOR_PREF to the channel var docs * res/res_features.c, CHANGES: Added support for reading the TOUCH_MONITOR_PREFIX channel variable. It allows you to configure a prefix for auto-monitor recordings. (closes issue #6353) Reported by: ivanfm Patches: asterisk_automon_v4.patch uploaded by ivanfm (original patch) - updated patch: 6353-touch_monitor_prefix.diff uploaded by qwell (license 4) * /, main/utils.c: Merged revisions 85649 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85649 | russell | 2007-10-15 14:22:45 -0500 (Mon, 15 Oct 2007) | 2 lines Be pedantic about handling memory allocation failure. ........ * /, main/utils.c: Merged revisions 85647 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85647 | russell | 2007-10-15 14:11:38 -0500 (Mon, 15 Oct 2007) | 5 lines The loop in the handler for the "core show locks" could potentially block for some amount of time. Be a little bit more careful and prepare all of the output in an intermediary buffer while holding a global resource. Then, after releasing it, send the output to ast_cli(). ........ 2007-10-15 17:51 +0000 [r85633] Tilghman Lesher * funcs/func_strings.c: Document my changes from Friday 2007-10-15 16:59 +0000 [r85605] Russell Bryant * /, channels/chan_sip.c: Merged revisions 85604 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85604 | russell | 2007-10-15 11:54:57 -0500 (Mon, 15 Oct 2007) | 6 lines Make the default for the srvlookup option to be yes. It doesn't really make sense for it to default to off. The default configuration file has it on, and proper RFC behavior, as indicated by a comment in the code, is for it to be on. So, let's have it on by default to make lives easier. (closes issue #10954, suggested by jtodd) ........ 2007-10-15 16:41 +0000 [r85578] Joshua Colp * /, configs/features.conf.sample: Merged revisions 85571 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85571 | file | 2007-10-15 13:39:59 -0300 (Mon, 15 Oct 2007) | 4 lines Document that DTMF based features only work when two channels are bridged together. (closes issue #10773) Reported by: pbayley ........ 2007-10-15 16:36 +0000 [r85562] Russell Bryant * /, include/asterisk/strings.h: Merged revisions 85561 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85561 | russell | 2007-10-15 11:34:13 -0500 (Mon, 15 Oct 2007) | 4 lines Make a few changes so that characters in the upper half of the ISO-8859-1 character set don't get stripped when reading configuration. (closes issue #10982, dandre) ........ 2007-10-15 16:23 +0000 [r85560] Joshua Colp * /, main/rtp.c: Merged revisions 85559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85559 | file | 2007-10-15 13:22:02 -0300 (Mon, 15 Oct 2007) | 4 lines Bring both DTMF begin and end frames up through to the core for DTMF feature handling. (closes issue #10826) Reported by: dimas ........ 2007-10-15 15:55 +0000 [r85557-85558] Russell Bryant * pbx/dundi-parser.c: Simplify buffer handling in dundi-parser.c. This also makes the code a bit safer by removing various assumptions about sizes. (No vulnerabilities, though) (closes issue #10977) Reported by: dimas Patches: dundiparser.patch uploaded by dimas (license 88) * /, pbx/pbx_dundi.c: Merged revisions 85556 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85556 | russell | 2007-10-15 10:40:45 -0500 (Mon, 15 Oct 2007) | 9 lines Ensure the buffer passed to ast_canmatch_extension() is properly initialized so that it is null terminated. (issue #10977) Reported by: dimas Patches: pbxdundi.patch uploaded by dimas (license 88) - small mods by me ........ 2007-10-15 15:26 +0000 [r85555] Philippe Sultan * channels/chan_jingle.c: Allow RTP structure registration 2007-10-15 15:07 +0000 [r85553-85554] Joshua Colp * main/frame.c: Add packetization data for G.722. (closes issue #10900) Reported by: andrew Patches: frame.diff uploaded by andrew (license 240) * /, main/rtp.c: Merged revisions 85552 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85552 | file | 2007-10-15 11:55:04 -0300 (Mon, 15 Oct 2007) | 4 lines If Monitor or a spy was added to a P2P or native bridged channel bring the channel back to the generic bridging core so the monitor or spy operations work. (closes issue #10943) Reported by: julianjm ........ 2007-10-15 13:51 +0000 [r85551] Philippe Sultan * res/res_jabber.c: Allocate more space for the base64 output we need to generate. Closes issue #10913, reported by tootai, who graciously granted us access to his Asterisk server, thanks! Daniel, feel free to reopen the bug in case you can reproduce this on 1.4. 2007-10-15 13:44 +0000 [r85539-85550] Russell Bryant * main/cli.c: Move the CLI commands that were in builtins[] into the cli_cli[] array of CLI commands and remove the cli_iterator struct. This gets tab completion working again. (closes issue #10970) Reported by: jamesgolovich Patches: asterisk-clicomplete.diff.txt uploaded by jamesgolovich (license 176) * doc/tex/jitterbuffer.tex, doc/tex/extensions.tex, doc/tex/channelvariables.tex, doc/tex/ael.tex, doc/tex/queues-with-callback-members.tex, doc/tex/realtime.tex, doc/tex/dundi.tex, doc/tex/security.tex, doc/tex/configuration.tex, doc/tex/ajam.tex, doc/tex/cliprompt.tex, doc/tex/manager.tex, doc/tex/misdn.tex, doc/tex/imapstorage.tex, doc/tex/privacy.tex, doc/tex/sla.tex, doc/tex/app-sms.tex, doc/tex/billing.tex, apps/app_zapateller.c, doc/tex/localchannel.tex, doc/tex/cdrdriver.tex, doc/tex/queuelog.tex: Another major doc directory update from IgorG. This patch includes - Many uses of the astlisting environment around verbatim text to ensure that it gets properly formatted and doesn't run off the page. - Update some things that have been deprecated. - Add escaping as needed - and more ... (closes issue #10978) Reported by: IgorG Patches: texdoc-85542-1.patch uploaded by IgorG (license 20) * /, main/asterisk.c: Merged revisions 85545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85545 | russell | 2007-10-15 08:05:45 -0500 (Mon, 15 Oct 2007) | 7 lines Make sure remote consoles unmute themselves again after reconnecting. (closes issue #10847) Reported by: atis Patches: console_unmute_on_reconnect.patch uploaded by atis (license 242) ........ * /, main/utils.c: Merged revisions 85543 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85543 | russell | 2007-10-15 07:48:10 -0500 (Mon, 15 Oct 2007) | 8 lines Make sure that the base64 decoder returns a terminated string. (closes issue #10979) Reported by: ys Patches: util.c.diff uploaded by ys (license 281) - small mods by me ........ * configure, configure.ac: Change the configure script to check for a function that was recently added to libss7. * /, pbx/pbx_config.c: Merged revisions 85540 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85540 | russell | 2007-10-14 10:24:52 -0500 (Sun, 14 Oct 2007) | 7 lines Don't create the context for users in users.conf until we know at least one user exists. (closes issue #10971) Reported by: dimas Patches: pbxconfig.patch uploaded by dimas (license 88) ........ * doc/tex/backtrace.tex (added): When merging the last documentation update, I forgot to "svn add" a file. Here it is. (closes issue #10962) 2007-10-13 08:38 +0000 [r85535] James Golovich * main/cli.c: Fix compiling cli.c due to differences with new cli system (closes issue 0010966) 2007-10-13 05:53 +0000 [r85534] Russell Bryant * include/asterisk/logger.h, /, main/asterisk.c, main/cli.c: Merged revisions 85533 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85533 | russell | 2007-10-13 01:48:10 -0400 (Sat, 13 Oct 2007) | 12 lines Fix an issue with console verbosity when running asterisk -rx to execute a command and retrieve its output. The issue was that there was no way for the main Asterisk process to know that the remote console was connecting in the -rx mode. The way that James has fixed this is to have all remote consoles muted by default. Then, regular remote consoles automatically execute a CLI command to unmute themselves when they first start up. (closes issue #10847) Reported by: atis Patches: asterisk-consolemute.diff.txt uploaded by jamesgolovich (license 176) ........ 2007-10-12 20:06 +0000 [r85527] Mark Michelson * configs/queues.conf.sample, apps/app_queue.c: Allow for the position announcement to be turned off if desired. (closes issue #8515, reported by bruno_rocha, initial patch by bruno_rocha, final patch by qwell) 2007-10-12 19:41 +0000 [r85525-85526] Matthew Fredrickson * channels/chan_zap.c, doc/tex/channelvariables.tex: Trying to finish the last of the charge_number patch up #10916 * channels/chan_zap.c: Add support for receive charge number in dialplan #10916 2007-10-12 18:37 +0000 [r85522-85524] Tilghman Lesher * doc/asterisk-mib.txt, doc/PEERING, /, LICENSE: Merged revisions 85523 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85523 | tilghman | 2007-10-12 13:30:55 -0500 (Fri, 12 Oct 2007) | 2 lines Change Digium address ........ * funcs/func_strings.c: Enable ranges, hexadecimal, octal, and special backslashed characters for the FILTER function 2007-10-12 15:50 +0000 [r85516-85519] Russell Bryant * doc/tex/odbcstorage.tex, doc/tex/extensions.tex, doc/tex/channelvariables.tex, doc/tex/ael.tex, doc/tex/queues-with-callback-members.tex, doc/tex/dundi.tex, doc/tex/enum.tex, doc/tex/cliprompt.tex, doc/tex/manager.tex, doc/tex/privacy.tex, doc/tex/sla.tex, doc/tex/app-sms.tex, doc/tex/localchannel.tex, doc/tex/ices.tex, doc/tex/cdrdriver.tex, doc/tex/asterisk.tex: Many doc directory improvements, including: - Added development section (backtrace.tex) - Correct filesystem path formating - Replace all "|" argument separator to "," - Endless count of spaces at the end of line - Using astlisting to make listings do not take so much place - Take back ASTRISKVERSION on first page - Make localchannel.tex readable by inserting extra end of lines (closes issue #10962) Reported by: IgorG Patches: texdoc-85177-1.patch uploaded by IgorG (license 20) * res/res_smdi.c, /: Merged revisions 85517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85517 | russell | 2007-10-12 10:45:09 -0500 (Fri, 12 Oct 2007) | 3 lines Fix a spelling error in a log message. SMDI, not SDMI. (closes issue #10959) ........ * /, pbx/pbx_realtime.c: Merged revisions 85515 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85515 | russell | 2007-10-12 10:40:35 -0500 (Fri, 12 Oct 2007) | 7 lines Fix the potential use of an uninitialized buffer in a log message. (closes issue #10958) Reported by: dimas Patches: realtime.patch uploaded by dimas (license 88) ........ 2007-10-11 22:42 +0000 [r85474-85499] Matthew Fredrickson * apps/app_dial.c: Make sure we propogate ANI2 to the outbound channel * funcs/func_callerid.c: See if I can fix this borked ANI2 code I added * channels/chan_zap.c: Make sure we set the ANI2 field for PRI * funcs/func_callerid.c: Add ANI2 support to func_callerid * channels/chan_zap.c: Add SS7 ANI2 support tx and rx. #10916 * channels/chan_zap.c: Add CCR test support #10916 2007-10-11 19:03 +0000 [r85460] Russell Bryant * main/udptl.c, main/threadstorage.c, res/res_limit.c, main/translate.c, res/res_crypto.c, res/res_convert.c, channels/iax2-provision.c, channels/chan_gtalk.c, channels/chan_oss.c, main/astobj2.c, main/cli.c, main/cdr.c, main/channel.c, apps/app_osplookup.c, channels/chan_skinny.c, pbx/pbx_ael.c, main/file.c, pbx/pbx_dundi.c, main/image.c, pbx/pbx_config.c, apps/app_mixmonitor.c, apps/app_rpt.c, main/asterisk.c, main/db.c, channels/chan_mgcp.c, res/res_clioriginate.c: Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) 2007-10-11 17:17 +0000 [r85431-85444] Matthew Fredrickson * channels/chan_zap.c: Let's hard code this until I fix it * channels/chan_zap.c: Make sure we are clean to build without libpri 2007-10-11 04:40 +0000 [r85357] Tilghman Lesher * main/pbx.c, /: Merged revisions 85356 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85356 | tilghman | 2007-10-10 23:35:33 -0500 (Wed, 10 Oct 2007) | 2 lines A dollar sign by itself, not indicating a start of a variable or expression prematurely ends substitution (closes issue #10939) ........ 2007-10-10 16:01 +0000 [r85317] Russell Bryant * include/asterisk/file.h, /: Merged revisions 85316 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85316 | russell | 2007-10-10 10:56:23 -0500 (Wed, 10 Oct 2007) | 6 lines I introduced a new member to the ast_filestream struct in 1.4.12, but put it in the middle of the struct, instead of at the end. One of the Debian folks, paravoid, pointed out that this breaks binary compatability with modules compiled against older headers. So, I'm moving the new member to the end of the struct to resolve the situation. ........ 2007-10-10 14:43 +0000 [r85281] Joshua Colp * /, channels/chan_sip.c: Merged revisions 85280 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85280 | file | 2007-10-10 11:42:00 -0300 (Wed, 10 Oct 2007) | 4 lines If devicestate is passed a port number strip it out. (closes issue #10930) Reported by: ibc ........ 2007-10-10 14:38 +0000 [r85279] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 85276 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85276 | mmichelson | 2007-10-10 09:26:31 -0500 (Wed, 10 Oct 2007) | 5 lines A bunch of changes from sprintf to snprintf. See security advisory AST-2002-022 ........ 2007-10-10 14:30 +0000 [r85234-85278] Joshua Colp * /, channels/chan_sip.c: Merged revisions 85277 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85277 | file | 2007-10-10 11:28:18 -0300 (Wed, 10 Oct 2007) | 6 lines Add support for handling a 182 Queued response. (closes issue #10924) Reported by: ramonpeek Patches: queued-182.diff uploaded by ramonpeek (license 266) ........ * /, apps/app_voicemail.c: Merged revisions 85242 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85242 | file | 2007-10-10 11:14:56 -0300 (Wed, 10 Oct 2007) | 6 lines Close voicemail message description file if duration did not meet the minimum, or else we will eventually run out of file descriptors. (closes issue #10918) Reported by: brak2718 Patches: vm1.4.12.1.patch uploaded by brak2718 (license 279) ........ * main/logger.c: Process outstanding log messages before shutting down the logger thread. (closes issue #10933) Reported by: sperreault 2007-10-10 06:48 +0000 [r85197] Luigi Rizzo * bootstrap.sh: Adapt the autotools names to different versions of FreeBSD (and open the way to better adaptation for other platforms as well). 2007-10-10 06:41 +0000 [r85196] Kevin P. Fleming * /, include/asterisk/frame.h: Merged revisions 85195 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85195 | kpfleming | 2007-10-10 08:24:41 +0200 (Wed, 10 Oct 2007) | 2 lines use a macro instead of an inline function, so that backtraces will report the caller of ast_frame_free() properly ........ 2007-10-09 22:35 +0000 [r85177] Mark Michelson * apps/app_queue.c: Patch to add one-touch parking for queues. (closes issue #10869, reported and patched by bluecrow76) 2007-10-09 22:21 +0000 [r85140-85176] Tilghman Lesher * main/channel.c, /, main/utils.c, include/asterisk/lock.h: Merged revisions 85158 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85158 | tilghman | 2007-10-09 16:55:06 -0500 (Tue, 09 Oct 2007) | 5 lines This commit fixes the following issues: - Deadlock in ast_write (issue #10406) - Deadlock in ast_read (issue #10406) - Possible mutex initialization error in lock.h (issue #10571) ........ * apps/app_dial.c, channels/chan_jingle.c, channels/chan_misdn.c, apps/app_festival.c, apps/app_minivm.c, apps/app_zapras.c, utils/astman.c, apps/app_adsiprog.c, utils/check_expr.c: Remove redundant includes (patch by snuffy) (Closes issue #10922) 2007-10-09 15:12 +0000 [r85097-85098] Russell Bryant * CHANGES: Note jitterbuffer support for chan_local in CHANGES * channels/chan_local.c, doc/tex/localchannel.tex: Add jitterbuffer support for chan_local. To enable it, you use the 'j' option in the Dial command. The 'j' option _must_ be used in conjunction with the 'n' option. This feature will allow you to use the existing jitterbuffer implementation to put a jitterbuffer on incoming SIP calls connecting to Asterisk applications by putting a local channel in the middle. 2007-10-09 14:31 +0000 [r84991-85094] Joshua Colp * /, channels/chan_sip.c: Merged revisions 85093 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85093 | file | 2007-10-09 11:30:16 -0300 (Tue, 09 Oct 2007) | 4 lines Don't perform a reinvite if a transfer is in progress. (issue #10915) Reported by: ramonpeek ........ * /, main/rtp.c: Merged revisions 85057 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85057 | file | 2007-10-08 17:06:33 -0300 (Mon, 08 Oct 2007) | 4 lines Only update codec information if the channel has a technology private structure. (issue #10915) Reported by: ramonpeek ........ * res/res_limit.c, utils/hashtest2.c, utils/conf2ael.c, main/ast_expr2.c, utils/check_expr.c: Fix up tree so that it compiles when MTX Profiling is enabled. (closes issue #10898) Reported by: snuffy Patches: 10898-mtx_prof.diff uploaded by qwell (license 4) * /, main/rtp.c: Merged revisions 85023 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85023 | file | 2007-10-08 12:37:46 -0300 (Mon, 08 Oct 2007) | 4 lines Update codec information as well as address when doing hold reinvites. (issue #10868) Reported by: mavince ........ * main/channel.c, /: Merged revisions 84990 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84990 | file | 2007-10-08 12:03:07 -0300 (Mon, 08 Oct 2007) | 4 lines Don't keep trying to native bridge if either of the channels are involved in a masquerade operation to be done. (closes issue #10696) Reported by: tbelder ........ 2007-10-08 03:29 +0000 [r84958] Russell Bryant * /, Makefile.rules: Merged revisions 84957 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84957 | russell | 2007-10-07 22:28:34 -0500 (Sun, 07 Oct 2007) | 6 lines Enable file dependency tracking for _all_ builds, and not just for builds with dev-mode enabled. I have seen enough problems caused by this that I don't think it's worth keeping. I want to continue to encourage anybody that is interested to continue to run Asterisk from svn. Furthermore, I do not want their systems to break when we change a structure definition in a header file. :) ........ 2007-10-07 16:28 +0000 [r84891-84939] Philippe Sultan * configs/jabber.conf.sample, include/asterisk/jabber.h, res/res_jabber.c: Make the status and priority configurable. Closes issue #10785, patch by Luke-Jr, thanks! * /, res/res_jabber.c: Merged revisions 84902 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84902 | phsultan | 2007-10-07 18:15:39 +0200 (Sun, 07 Oct 2007) | 5 lines Presence packets from a client who's connected with our Jabber ID are valid, therefore, those clients must be considered as buddies. The resource string helps us make the distinction between clients. Closes issue #10707, reported by yusufmotiwala. ........ * res/res_jabber.c: Fix indentation * /, res/res_jabber.c: Merged revisions 84890 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84890 | phsultan | 2007-10-07 17:52:44 +0200 (Sun, 07 Oct 2007) | 5 lines Prevent Asterisk from crashing when receiving a presence packet without resource from a buddy that is known to have a resource list. Revert a change I previously made, where Asterisk could point to a freed memory location. ........ 2007-10-05 19:48 +0000 [r84852] Tilghman Lesher * /, main/db.c: Merged revisions 84851 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84851 | tilghman | 2007-10-05 14:42:21 -0500 (Fri, 05 Oct 2007) | 2 lines Log exactly why we can't open the database, if we fail (closes issue #10887) ........ 2007-10-05 18:57 +0000 [r84819] Joshua Colp * /, main/rtp.c: Merged revisions 84818 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84818 | file | 2007-10-05 15:55:36 -0300 (Fri, 05 Oct 2007) | 4 lines Update the remembered RTP peer information when putting an endpoint on hold or taking it off hold so that the RTP stack does not initiate a needless reinvite. (closes issue #10868) Reported by: mavince ........ 2007-10-05 16:49 +0000 [r84743-84784] Russell Bryant * channels/chan_zap.c, /: Merged revisions 84783 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84783 | russell | 2007-10-05 11:44:21 -0500 (Fri, 05 Oct 2007) | 4 lines Do deadlock avoidance in a couple more places. You can't lock two channels at the same time without doing extra work to make sure it succeeds. (closes issue #10895, patch by me) ........ * main/manager.c, /: Merged revisions 84742 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84742 | russell | 2007-10-04 20:39:07 -0500 (Thu, 04 Oct 2007) | 3 lines Fix a copy/paste error in the description of UpdateConfig that was pointed out by JerJer on #asterisk-dev ........ 2007-10-04 22:58 +0000 [r84693-84726] Mark Michelson * apps/app_queue.c: A two-in-one patch from the bugtracker 1) Fix some bad logic in the counting of statistics for QueueSummary manager event. Variables were not being reset for each additional queue, so cumulative totals were reported on each successive queue. 2) Add a longest hold time stat to QueueSummary manager event. * /, apps/app_queue.c: Merged revisions 84692 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84692 | mmichelson | 2007-10-04 16:57:03 -0500 (Thu, 04 Oct 2007) | 5 lines Don't allocate space for queue members unless it's needed. You end up deleting dynamic members on a reload. Not good. closes issue (#10879, reported by dazza76, patched by me) ........ 2007-10-04 21:38 +0000 [r84691] Kevin P. Fleming * channels/chan_zap.c, /: Merged revisions 84690 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84690 | kpfleming | 2007-10-04 16:36:56 -0500 (Thu, 04 Oct 2007) | 2 lines callers of sig2str already add the word 'signalling' in the appropriate place, so don't duplicate it ........ 2007-10-04 16:56 +0000 [r84671] Tilghman Lesher * res/res_jabber.c: Update to current coding standards, also changing the argument delimiter to ',' (Closes issue #10876) 2007-10-04 14:54 +0000 [r84613-84638] Joshua Colp * /, apps/app_queue.c: Merged revisions 84637 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84637 | file | 2007-10-04 11:51:57 -0300 (Thu, 04 Oct 2007) | 4 lines Create a duplicate of the channel's member name as the tab completion stuff will free it. (closes issue #10884) Reported by: adamg ........ * main/pbx.c: Don't register the exception function with module information. Since it is in the core there is none and it will explode. 2007-10-03 23:05 +0000 [r84580-84582] Tilghman Lesher * /, main/rtp.c: Merged revisions 84581 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84581 | tilghman | 2007-10-03 17:59:17 -0500 (Wed, 03 Oct 2007) | 2 lines When an RFC 2833 event is sent that we don't recognize, ignore it, don't queue a NULL digit (closes issue #10877) ........ * main/pbx.c, doc/tex/extensions.tex, include/asterisk/pbx.h: Create a universal exception handling extension, "e" (closes issue #9785) 2007-10-03 18:23 +0000 [r84512-84545] Steve Murphy * /: blocked 84544 from trunk; it only applies to 1.4; 10870 -- the CUT in AEL * res/ael/pval.c, pbx/ael/ael-test/ref.ael-vtest17, /, pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test5, pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18, pbx/ael/ael-test/ref.ael-test19, pbx/ael/ael-test/ref.ael-vtest13: Merged revisions 84511 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84511 | murf | 2007-10-03 08:23:00 -0600 (Wed, 03 Oct 2007) | 1 line closes issue #10834 ; where a null input to a switch statement results in a hangup; since switch is implemented with extensions, and the default case is implemented with a '.', and the '.' matches 1 or more remaining characters, the case where 0 characters exist isn't matched, and the extension isn't matched, and the goto fails, and a hangup occurs. Now, when a default case is generated, it also generates a single fixed extension that will match a null input. That extension just does a goto to the default extension for that switch. I played with an alternate solution, where I just tack an extra char onto all the patterns and the goto, but not the default case's pattern. Then even a null input will still have at least one char in it. But it made me nervous, having that extra char in , even if that's a pretty secret and low-level issue. ........ 2007-10-02 20:07 +0000 [r84475] Russell Bryant * Makefile, /, build_tools/prep_tarball: Merged revisions 84474 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84474 | russell | 2007-10-02 15:06:07 -0500 (Tue, 02 Oct 2007) | 5 lines * Don't build the menuselect-tree for the tarball, as it requires running the configure script first * Change the Makefile to note that menuselect-tree depends on the configure script. ........ 2007-10-02 19:02 +0000 [r84432-84440] Jason Parker * /, res/res_features.c: Merged revisions 84410 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #10821) ........ r84410 | qwell | 2007-10-02 13:52:55 -0500 (Tue, 02 Oct 2007) | 4 lines Finish up on transferee channel before return on failure. Issue 10821, patch by Ivan ........ 2007-10-02 18:12 +0000 [r84405] Tilghman Lesher * main/pbx.c: Add MSet for people who prefer the old, deprecated syntax of Set (Closes issue #10549) 2007-10-02 14:13 +0000 [r84371] Russell Bryant * /, channels/chan_sip.c: Merged revisions 84370 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84370 | russell | 2007-10-02 09:12:35 -0500 (Tue, 02 Oct 2007) | 6 lines Use snprintf instead of sprintf in one place. There is no vulnerability here due to various buffer sizes around the code, but I still didn't like seeing a non length-limited copy of data coming off of the wire into a stack buffer, as this would be a problem in the future if buffer sizes elsewhere got changed or size limitations removed ... ........ 2007-10-02 13:58 +0000 [r84368] Joshua Colp * main/rtp.c: Don't swap channel priority if using epoll as polling should/will only happen off the first channel. (closes issue #10867) Reported by: phsultan 2007-10-01 23:33 +0000 [r84327-84331] Steve Murphy * utils/check_expr.c: OK. THis a DEBUG_THREADS situation. * utils/check_expr.c: picky gcc versions... sigh. * utils/check_expr.c: This mod will allow check_expr to compile in the presence of DEBUG_THREAD situations. At least, it does for me. And it's less expensive than several other approaches I tried. * res/ael/pval.c, /, res/ael/ael.tab.c, res/ael/ael.y, pbx/pbx_ael.c: Merged revisions 84239 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84239 | murf | 2007-10-01 14:27:52 -0600 (Mon, 01 Oct 2007) | 1 line closes issue #10777 -- by returning a null for the parse tree when there's really nothing there, and making sure we don't try to do checking on a null tree. ........ 2007-10-01 21:54 +0000 [r84300] Jason Parker * Makefile, /, Makefile.rules, channels/Makefile: Merged revisions 84291 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84291 | qwell | 2007-10-01 16:52:45 -0500 (Mon, 01 Oct 2007) | 6 lines Add dist-clean support for subdirs. Change h323 to only remove the Makefile on a dist-clean, rather than a clean. This fixes a bug I found with trying to run make after a make clean ........ 2007-10-01 21:31 +0000 [r84275] Dwayne M. Hubbard * main/channel.c, main/manager.c, /, channels/chan_agent.c: Merged revisions 84274 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84274 | dhubbard | 2007-10-01 16:25:37 -0500 (Mon, 01 Oct 2007) | 1 line moved get_base_channel() code from action_redirect to ast_channel_masquerade() for issue 7706 and BE-160 ........ 2007-10-01 21:15 +0000 [r84207-84272] Russell Bryant * /, main/utils.c, include/asterisk/lock.h: Merged revisions 84271 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84271 | russell | 2007-10-01 16:07:06 -0500 (Mon, 01 Oct 2007) | 4 lines Fulfull a feature request from Qwell on the "core show locks" output. It will now note the lock type for each lock that a thread holds. (mutex, rdlock, or wrlock) ........ * /, res/res_agi.c: Merged revisions 84236 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84236 | russell | 2007-10-01 14:56:28 -0500 (Mon, 01 Oct 2007) | 5 lines Add another sanity check in the AGI read loop. We really don't care about EAGAIN unless we didn't read an entire line. If there is a newline at the end if the read buffer, break, because we got the whole thing. (reported and patched by bmd) ........ * /, include/asterisk/lock.h: Merged revisions 84206 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84206 | russell | 2007-10-01 14:34:12 -0500 (Mon, 01 Oct 2007) | 2 lines Show rwlocks in the "core show locks" output. Before, it only showed mutexes. ........ 2007-10-01 15:57 +0000 [r84176] Joshua Colp * channels/chan_sip.c: Check to make sure a structure pointer is non-NULL before touching it... crashing is bad, mmmk? (closes issue #10831) Reported by: eliel Patches: chan_sip.c.patch uploaded by eliel (license 64) 2007-10-01 15:34 +0000 [r84167-84174] Russell Bryant * main/say.c: Change simple uses of snprintf to ast_copy_string. This was provided by mvanbaak as a part of issue #10843, but this part didn't apply because of a patch I applied right beforehand. * channels/chan_misdn.c, main/frame.c, res/res_config_odbc.c, apps/app_meetme.c, channels/chan_sip.c, channels/chan_skinny.c, main/say.c, apps/app_minivm.c, pbx/dundi-parser.c, channels/chan_iax2.c, channels/iax2-parser.c, main/asterisk.c, main/rtp.c, channels/chan_mgcp.c: Corydon posted this janitor project to the bug tracker and mvanbaak provided a patch for it. It replaces a bunch of simple calls to snprintf with ast_copy_string (closes issue #10843) Reported by: Corydon76 Patches: 2007092900_10843.diff uploaded by mvanbaak (license 7) * main/say.c: Simplify code by using the -= and %= operators. (closes issue #10848) Reported by: opticron Patches: saymod.diff uploaded by opticron (license 267) * codecs/g722/Makefile, /, res/Makefile, channels/Makefile: The trunk version of this patch also includes a couple more small clean fixes from IgorG. Merged revisions 84170 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84170 | russell | 2007-10-01 10:00:56 -0500 (Mon, 01 Oct 2007) | 3 lines Remove another file in "make clean". (closes issue #10814, paravoid) ........ * main/cli.c: Don't set the full command string until after verifying that there is not another CLI command with the same command text registered. This prevents a crash if someone accidentally calls ast_cli_register() on the same CLI command data twice. This also fixes a small bug where the helpers list would get unlocked without being locked if building the full command failed. (closes issue #10858, reported by jamesgolovich, patched by me) * configs/musiconhold.conf.sample, res/res_musiconhold.c: Add a new option for files-based music on hold to ensure that the sort order of the files is alphabetical. (closes issue #10855) Reported by: jamesgolovich Patches: asterisk-mohsortalpha.diff.txt uploaded by jamesgolovich (license 176) * apps/app_dial.c, /: Merged revisions 84166 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84166 | russell | 2007-10-01 09:24:49 -0500 (Mon, 01 Oct 2007) | 2 lines Simplify the CAN_EARLY_BRIDGE macro a bit. ........ 2007-10-01 14:21 +0000 [r84159-84165] Joshua Colp * channels/chan_sip.c: Add MP4 to part of the SDP code. (closes issue #10820) Reported by: ruikubo Patches: chan_sip.patch uploaded by ruikubo (license 250) * main/dnsmgr.c: Don't register the dnsmgr refresh CLI command twice. (closes issue #10856) Reported by: jamesgolovich Patches: asterisk-dnsmgrclireg.diff.txt uploaded by jamesgolovich (license 176) * /, res/res_musiconhold.c: Merged revisions 84160 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84160 | file | 2007-10-01 10:57:42 -0300 (Mon, 01 Oct 2007) | 6 lines Fix randomness. save_pos was being set to 0 initially instead of -1, causing it to jump to position 0 when moh started. (closes issue #10859) Reported by: jamesgolovich Patches: asterisk-mohpos2.diff.txt uploaded by jamesgolovich (license 176) ........ * apps/app_dial.c, /: Merged revisions 84158 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84158 | file | 2007-10-01 10:49:36 -0300 (Mon, 01 Oct 2007) | 4 lines Only attempt early bridging if the options given to Dial() permit it. (closes issue #10861) Reported by: peekyb ........ 2007-09-30 20:06 +0000 [r84143-84147] Russell Bryant * /, include/asterisk/module.h: Merged revisions 84146 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84146 | russell | 2007-09-30 16:02:16 -0400 (Sun, 30 Sep 2007) | 4 lines Fix the AST_MODULE_INFO macro for C++ modules. The load and reload parameters were in the wrong place. (closes issue #10846, alebm) ........ * funcs/func_lock.c: * The documentation for the LOCK() function says that it will block for up to 3 seconds while waiting on a lock when other locks are currently held to avoid deadlocks. Change the code to reflect this. * Since trying to grab a lock may block for some time, put the channel in autoservice so that audio is still read from the channel and that any active generators on the channel don't pause. 2007-09-29 23:47 +0000 [r84134-84137] Steve Murphy * /, res/ael/ael_lex.c, res/ael/ael.flex: Merged revisions 84133 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84133 | murf | 2007-09-29 15:47:53 -0600 (Sat, 29 Sep 2007) | 1 line This issue sort of closes 10786; All config files support #include with globbing (you know, *,[chars],?,{list,list},etc), so I've updated the AEL system to support this also. ........ * pbx/ael/ael-test/ael-ntest22/t2 (added), pbx/ael/ael-test/ael-ntest22/t3 (added), pbx/ael/ael-test/ael-ntest22/extensions.ael (added), pbx/ael/ael-test/ael-ntest22 (added), pbx/ael/ael-test/ael-ntest22/t1/a.ael (added), pbx/ael/ael-test/ael-ntest22/t1/b.ael (added), pbx/ael/ael-test/ael-ntest22/t1/c.ael (added), pbx/ael/ael-test/ael-ntest22/t2/d.ael (added), pbx/ael/ael-test/ael-ntest22/t2/e.ael (added), pbx/ael/ael-test/ael-ntest22/t2/f.ael (added), pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-ntest22 (added), pbx/ael/ael-test/ael-ntest22/t3/g.ael (added), pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ael-ntest22/t3/h.ael (added), pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ael-ntest22/t3/i.ael (added), pbx/ael/ael-test/ael-ntest22/t3/j.ael (added), pbx/ael/ael-test/ael-ntest22/qq.ael (added), pbx/ael/ael-test/ael-ntest22/t1 (added): the last commit for AEL affected a small number of tests. Added a regression test for glob'd includes 2007-09-29 18:21 +0000 [r84130] Tilghman Lesher * cdr/cdr_manager.c: Set enablecdr at the end of re-reading the config file (Closes issue #10852) 2007-09-29 00:19 +0000 [r84115] Matthew Fredrickson * main/translate.c: Let's use process time instead of wall clock time for show translation 2007-09-28 14:35 +0000 [r84050-84080] Tilghman Lesher * configure, configure.ac: Autoconf requires version 2.60, not 2.59, to process (Closes issue #10842) * /, main/say.c: Merged revisions 84078 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84078 | tilghman | 2007-09-28 09:13:47 -0500 (Fri, 28 Sep 2007) | 2 lines Correct pronunciations of numbers for .nl (Closes issue #10837) ........ * main/channel.c, /: Merged revisions 84049 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84049 | tilghman | 2007-09-28 00:30:22 -0500 (Fri, 28 Sep 2007) | 3 lines Avoid a deadlock with ALL of the locks in the masquerade function, not just the pairs of channels. (Closes issue #10406) ........ 2007-09-27 23:18 +0000 [r84019] Dwayne M. Hubbard * main/manager.c, /, channels/chan_agent.c, include/asterisk/channel.h: Merged revisions 84018 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84018 | dhubbard | 2007-09-27 18:12:25 -0500 (Thu, 27 Sep 2007) | 1 line if an Agent is redirected, the base channel should actually be redirected. This was causing multiple issues, especially issue 7706 and BE-160 ........ 2007-09-27 00:08 +0000 [r83978-83986] Kevin P. Fleming * /, channels/chan_alsa.c: Merged revisions 83974 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r83974 | kpfleming | 2007-09-26 16:53:03 -0700 (Wed, 26 Sep 2007) | 2 lines avoid the weird usage of assert() in the ALSA header files that gcc 4.2 wants to complain about ........ * res/ael/ael.tab.c, res/ael/ael.y: deal with more gcc 4.2 const pointer warnings 2007-09-27 00:02 +0000 [r83911-83977] Russell Bryant * /, pbx/pbx_dundi.c: Merged revisions 83976 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r83976 | russell | 2007-09-26 19:01:29 -0500 (Wed, 26 Sep 2007) | 1 line remove a todo item that has been completed ........ * /, channels/chan_sip.c: Merged revisions 83943 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r83943 | russell | 2007-09-26 16:35:23 -0500 (Wed, 26 Sep 2007) | 2 lines I changed my mind ... I think this should be a LOG_NOTICE. ........ * /, channels/chan_sip.c: Merged revisions 83941 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r83941 | russell | 2007-09-26 16:15:15 -0500 (Wed, 26 Sep 2007) | 5 lines Add a log message that was requested by the masses in the developer tutorial session at Astricon. chan_sip did not output any message when a call was rejected because the extension was not found. This adds a verbose message (at verbose level 3) to note when this happens. ........ * /: Merged revisions 83910 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r83910 | russell | 2007-09-26 15:50:09 -0500 (Wed, 26 Sep 2007) | 3 lines Fix building chan_misdn under dev-mode. (please run the configure script with --enable-dev-mode so this doesn't happen again ...) ........ 2007-09-26 18:43 +0000 [r83880] Tilghman Lesher * channels/chan_zap.c, /: Merged revisions 83879 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r83879 | tilghman | 2007-09-26 13:35:56 -0500 (Wed, 26 Sep 2007) | 2 lines Remove unused 4k of memory on the program stack (closes issue #10827) ........ 2007-09-26 06:53 +0000 [r83849-83864] Russell Bryant * include/asterisk/event.h: fix a typo in a comment * include/asterisk/file.h: Change function documentation to use doxygen tags. (Really, I just needed to make some minor change in trunk to test something with automerge ...) 2007-09-25 23:14 +0000 [r83834] Matthew Fredrickson * doc/ss7.txt: Fix typo in readme 2007-09-25 21:06 +0000 [r83819] Russell Bryant * include/asterisk/devicestate.h: Don't note that functions are deprecated in favor of themselves. This was found by showing a very poor example doxygen function in a presentation this morning. :) 2007-09-25 16:34 +0000 [r83804] Philippe Sultan * res/res_jabber.c: Added a CLI command that shows our buddy list, as suggested by Daniel McKeehan, thanks! 2007-09-25 14:18 +0000 [r83774] Tilghman Lesher * /, main/app.c: Merged revisions 83773 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r83773 | tilghman | 2007-09-25 09:13:25 -0500 (Tue, 25 Sep 2007) | 2 lines jmls pointed out that unsetting the group and setting the group to the blank string aren't quite the same. ........ 2007-09-25 13:41 +0000 [r83758] Joshua Colp * res/ael/pval.c: Fix minor memory leak in pval.c. Overwriting a value without freeing the previous result is bad, mmmk? 2007-09-25 09:07 +0000 [r83743] Philippe Sultan * channels/chan_jingle.c, include/asterisk/jingle.h: Comply with latest XEP-0166, XEP-0167, XEP-0176. No real Jingle implementation being available, testing was made using two Asterisk servers relaying SIP calls over their Jingle channels: SIP Phone 1 --- [chan_sip]Asterisk 1[chan_jingle] --- [chan_jingle]Asterisk 2[chan_sip] --- SIP Phone 2 Thus, it was possible to test the code in both ways, and make the Jingle channel comply with the latest specifications. No sound available yet. Main modifications include : - modified the 'jingle_candidate' structure and the 'jingle_create_candidates' function according to XEP-0176 ; - modified the 'jingle_action' function in order to properly terminate a Jingle session, in conformance with XEP-0166 ; - modified username format used in STUN requests ; - actually make the bindaddr configuration field useable. Todo : - set audio paths up (no native bridging) ; - make the CLI gtalk functions available to jingle ; - clean up the storage space used in strings. 2007-09-25 08:09 +0000 [r83741] Russell Bryant * utils/Makefile, utils: Add some files to the utils directory svn:ignore and Makefile clean target (closes issue #10808, reported by mvanbaak) 2007-09-24 22:06 +0000 [r83696-83726] Tilghman Lesher * Makefile, main/asterisk.c: Permit custom locations for astdb and the keys directory (though default to the current locations) (Closes issue #10267) * /, build_tools/make_defaults_h: Merged revisions 83695 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r83695 | tilghman | 2007-09-24 12:22:08 -0500 (Mon, 24 Sep 2007) | 4 lines In the source, keys are relative to the datadir, not varlib (which is the same in most cases, but it's good to be accurate). Closes issue #10811 ........ 2007-09-24 17:10 +0000 [r83671] Dwayne M. Hubbard * channels/chan_sip.c, configs/sip.conf.sample: merged jcmoore's patch for configurable SDP origin-field username and session field, closes issue# 10795 2007-09-24 17:00 +0000 [r83656] Mark Michelson * apps/app_queue.c: interface_exists_global was never returning 1. Most likely an error from my merge on Friday. (closes issue #10817, reported and patched by snar, patch simplified by me) 2007-09-24 16:42 +0000 [r83654-83655] Tilghman Lesher * /, main/app.c: Merged revisions 83637 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r83637 | tilghman | 2007-09-24 10:17:06 -0500 (Mon, 24 Sep 2007) | 3 lines Making change to group splitting, as discussed on the -dev list. The main effect of this will be to permit Set(GROUP([cat])=), i.e. unsetting a group. ........ 2007-09-22 19:54 +0000 [r83575-83590] Steve Murphy * res/ael/pval.c, /: Merged revisions 83589 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r83589 | murf | 2007-09-22 13:39:16 -0600 (Sat, 22 Sep 2007) | 1 line This closes issue #10788 -- The exact same fixes are made here for the first arg in the for(arg1; arg2; arg3) {} statement, as were done for the 3rd arg. It can now be an assignment that will embedded in a Set() app, or a macro call, or an app call. ........ * res/ael/pval.c, /, pbx/pbx_ael.c: Merged revisions 83558 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r83558 | murf | 2007-09-22 10:41:43 -0600 (Sat, 22 Sep 2007) | 1 line This closes issue #10788 -- the 3rd arg in the for statement is now wrapped in Set() only if there's an '=' in that string. Otherwise, if it begins with '&', then a Macro call is generated; otherwise it is made into an app call. A bit more accomodating, keeps the new guys happy, and the guys with ael-1 code should be happy, too ........ 2007-09-22 17:37 +0000 [r83574] Matthew Fredrickson * doc/ss7.txt: Fix potential point of confusion 2007-09-22 14:45 +0000 [r83517-83545] Tilghman Lesher * utils/Makefile, utils/hashtest2.c, utils/clicompat.c (added): Fix build of check_expr and hashtest2 when DEBUG_THREADLOCAL is defined * main/manager.c, apps/app_meetme.c: Add the MeetmeList and Reload manager commands, which supplement the need to have Command privilege. (closes issue #10736) * configure, include/asterisk/autoconfig.h.in, main/ast_expr2.h, main/ast_expr2.y, configure.ac, main/ast_expr2.c: Fixes for FreeBSD... testing for every conceivable math function now 2007-09-21 19:55 +0000 [r83500] Russell Bryant * channels/chan_zap.c: Fix compilation errors in CLI command updates to SS7 CLI commands 2007-09-21 19:54 +0000 [r83499] Matthew Fredrickson * doc/ss7.txt (added): Add an SS7 readme for setup and use of libss7 and asterisk 2007-09-21 18:41 +0000 [r83484] Tilghman Lesher * apps/app_queue.c: Fix some areas where we were still using '|' for an argument delimiter (closes issue #10793) 2007-09-21 18:27 +0000 [r83483] Russell Bryant * apps/app_queue.c: Update app_queue to use commas as application argument separators. (closes issue #10793, snar) 2007-09-21 17:36 +0000 [r83466] Tilghman Lesher * cdr/cdr_manager.c: Fix cdr_manager, such that if the config file is created past load, it'll start logging (and conversely, if the config file is destroyed or deactivated, the logging is disabled). Reported by Juggie via IRC, fix by me. 2007-09-21 14:40 +0000 [r83433] Russell Bryant * res/res_config_pgsql.c, main/dnsmgr.c, /, channels/chan_sip.c, main/db1-ast/hash/hash.c, include/asterisk/channel.h, channels/chan_iax2.c, main/rtp.c, channels/misdn_config.c, main/cdr.c, main/channel.c, channels/chan_misdn.c, main/ast_expr2f.c, main/file.c, include/asterisk/sched.h, channels/chan_h323.c, utils/ael_main.c, pbx/pbx_dundi.c, main/sched.c, channels/chan_mgcp.c, main/ast_expr2.fl: Merged revisions 83432 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r83432 | russell | 2007-09-21 09:37:20 -0500 (Fri, 21 Sep 2007) | 4 lines gcc 4.2 has a new set of warnings dealing with cosnt pointers. This set of changes gets all of Asterisk (minus chan_alsa for now) to compile with gcc 4.2. (closes issue #10774, patch from qwell) ........ 2007-09-21 14:25 +0000 [r83431] Tilghman Lesher * configure, include/asterisk/autoconfig.h.in, main/ast_expr2.h, main/ast_expr2.y, configure.ac, main/ast_expr2.c: Check for the presence of trunc and round, and make the ISOC99 detection a little more sane (closes issue #10776) 2007-09-20 23:14 +0000 [r83381] Jason Parker * apps/app_minivm.c, main/astmm.c, apps/app_playback.c: More NEW_CLI conversions. (issue #10724) Patches: app_playback.c.patch uploaded by moy (license 222) app_minivm.c.patch uploaded by eliel (license 64) astmm.c.patch uploaded by eliel (license 64) 2007-09-20 21:37 +0000 [r83350-83351] Mark Michelson * /: Oops. Getting rid of svnmerge-integrated and automerge stuff * /, apps/app_queue.c: Merging changes from queue_refcount_trunk into trunk. Refcounted queues now in place. 2007-09-20 21:17 +0000 [r83293-83349] Russell Bryant * /, main/asterisk.c: Merged revisions 83348 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r83348 | russell | 2007-09-20 16:16:48 -0500 (Thu, 20 Sep 2007) | 4 lines When daemonizing, don't change working directory to "/". It makes it not be able to do a core dump when not running as uid=root. (closes issue #10766, xrg) ........ * /, contrib/scripts/safe_asterisk: Merged revisions 83316 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r83316 | russell | 2007-09-20 16:01:20 -0500 (Thu, 20 Sep 2007) | 3 lines Change safe_asterisk to explicitly ask for /bin/bash, as it uses bashisms. (closes issue #10772, reported by culrich) ........ * main/dsp.c: trivial formatting change * main/asterisk.c: trivial formatting change * main/app.c: minor spelling fixes in a comment * main/app.c: minor grammar fix * channels/chan_sip.c: fix spelling in a comment * main/asterisk.c: trivial formatting change 2007-09-20 19:05 +0000 [r83251-83278] Jason Parker * doc/modules.txt: Fix a trivial typo, to test our new commit bot * /, apps/app_disa.c: Merged revisions 83246 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r83246 | qwell | 2007-09-20 12:09:14 -0500 (Thu, 20 Sep 2007) | 8 lines If # is pressed after dialing an extension in DISA, stop trying to collect more digits. (closes issue #10754) Reported by: atis Patches: app_disa.c.branch.patch uploaded by atis (license 242) app_disa.c.trunk.patch uploaded by atis (license 242) ........ 2007-09-20 16:28 +0000 [r83234] Joshua Colp * /, channels/chan_sip.c: Merged revisions 83232 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r83232 | file | 2007-09-20 13:25:30 -0300 (Thu, 20 Sep 2007) | 7 lines Make sure the minimum T1 timer value is obeyed in all cases. (closes issue #10768) Reported by: flefoll Patches: chan_sip.c.trunk.83071.retrans-patch uploaded by flefoll (license 244) chan_sip.c.br14.83070.retrans-patch uploaded by flefoll (license 244) ........ 2007-09-20 16:27 +0000 [r83233] Russell Bryant * main/asterisk.c: Don't start the event processing thread until after forking. (reported by Simon on the -dev list, thanks!) 2007-09-20 16:19 +0000 [r83229-83231] Joshua Colp * /, channels/chan_sip.c: Merged revisions 83230 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r83230 | file | 2007-09-20 13:17:24 -0300 (Thu, 20 Sep 2007) | 7 lines Fix a minor spelling error. (closes issue #10769) Reported by: flefoll Patches: chan_sip.c.trunk.83071.inita-patch uploaded by flefoll (license 244) chan_sip.c.br14.83070.inita-patch uploaded by flefoll (license 244) ........ * pbx/pbx_dundi.c, cdr/cdr_pgsql.c, main/config.c: Fix memory leaks in pbx_dundi, cdr_pgsql, and the configuration file parser. 2007-09-19 23:16 +0000 [r83213] Jason Parker * channels/chan_zap.c, apps/app_meetme.c, apps/app_queue.c, apps/app_voicemail.c: More conversions to NEW_CLI (issue #10724) Patches: chan_zap.c.patch uploaded by moy (license 222) app_queue.c.patch uploaded by eliel (license 64) app_voicemail.c.patch uploaded by eliel (license 64) app_meetme.c.patch uploaded by eliel (license 64) 2007-09-19 20:06 +0000 [r83182-83183] Joshua Colp * cdr/cdr_csv.c: Clean up code in cdr_csv. (Are you sensing a theme for me today?) * res/res_adsi.c: Clean up code in res_adsi. 2007-09-19 19:54 +0000 [r83176-83181] Russell Bryant * funcs/func_shell.c: put the channel in autoservice when executing func_shell * /, apps/app_system.c: Merged revisions 83179 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r83179 | russell | 2007-09-19 14:50:48 -0500 (Wed, 19 Sep 2007) | 5 lines The System() and TrySystem() applications can take a substantial amount of time to execute while not servicing the channel. So, put the channel in autoservice while the command is being executed. (closes issue #10726, reported by mnicholson) ........ * funcs/func_curl.c, /: Merged revisions 83177 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r83177 | russell | 2007-09-19 14:34:25 -0500 (Wed, 19 Sep 2007) | 4 lines Using curl can take a substantial amount of time, so the channel should be autoserviced while waiting for it to complete. (closes issue #10725, reported by mnicholson) ........ * /, channels/chan_iax2.c: Merged revisions 83175 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r83175 | russell | 2007-09-19 14:13:29 -0500 (Wed, 19 Sep 2007) | 8 lines When handling a reload of chan_iax2, don't use an ao2_callback() to POKE all peers. Instead, use an iterator. By using an iterator, the peers container is not locked while the POKE is being done. It can cause a deadlock if the peers container is locked because poking a peer will try to lock pvt structs, while there is a lot of other code that will hold a pvt lock when trying to go lock the peers container. (reported to me directly by Loic Didelot. Thank you for the debug info!) ........ 2007-09-19 17:22 +0000 [r83155-83157] Joshua Colp * apps/app_db.c: Fix indentation in app_db. * apps/app_authenticate.c: Clean up code in app_authenticate. * apps/app_adsiprog.c: Clean up code in app_adsiprog. 2007-09-19 15:11 +0000 [r83126] Russell Bryant * main/manager.c, /: Merged revisions 83121 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r83121 | russell | 2007-09-19 10:10:14 -0500 (Wed, 19 Sep 2007) | 4 lines Fix up another potential race condition. Do the loop decrementing use count on events with the eventq protected from being changed. (reported on IRC by Ivan) ........ 2007-09-19 15:08 +0000 [r83105-83114] Joshua Colp * apps/app_disa.c: DISA only needs to know about the end of DTMF, not the beginning/duration. * apps/app_disa.c: Clean up app_disa code a bit. 2007-09-19 13:55 +0000 [r83076] Philippe Sultan * channels/chan_jingle.c: Replace Google namespace occurrences with Jingle. The former namespace is handled by chan_gtalk. 2007-09-19 13:49 +0000 [r83073-83075] Joshua Colp * /, apps/app_queue.c: Merged revisions 83074 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r83074 | file | 2007-09-19 10:47:59 -0300 (Wed, 19 Sep 2007) | 6 lines Protect the CDR record from modification by pbx_exec so that the application data contains the Queue data. (closes issue #10761) Reported by: snar Patches: app-queue-mixmonitor.patch uploaded by snar (license 245) ........ * main/manager.c: Extend manager show connected with additional information. (closes issue #10757) Reported by: outtolunc Patches: manager.c.sessionstart.diff uploaded by outtolunc (license 237) 2007-09-19 13:29 +0000 [r83072] Philippe Sultan * channels/chan_jingle.c: Remove namespaces in payload-type tags. 2007-09-19 13:21 +0000 [r83071] Joshua Colp * /, channels/chan_sip.c: Merged revisions 83070 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r83070 | file | 2007-09-19 10:18:22 -0300 (Wed, 19 Sep 2007) | 6 lines (closes issue #10760) Reported by: dimas Patches: chan_sip.patch uploaded by dimas (license 88) Read in subscribecontext option in general to be the default. ........ 2007-09-19 12:23 +0000 [r83055] Philippe Sultan * channels/chan_jingle.c, include/asterisk/jingle.h: Transmit proper invitation, thus conforming to XEP-0166 (Jingle general specifications), XEP-0167 (Jingle Audio via RTP) and XEP-0176 (Jingle ICE Transport). 2007-09-19 09:48 +0000 [r83025] Christian Richter * channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h, channels/misdn_config.c: Merged revisions 83023-83024 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r83023 | crichter | 2007-09-19 11:31:55 +0200 (Mi, 19 Sep 2007) | 1 line added 'astdtmf' option to allow configuring the asterisk dtmf detector instead of the mISDN_dsp ones. also added the patch from irroot #10190, so that dtmf tones detected by the asterisk detector are passed outofband to asterisk, to make any use of dtmf tones at all. ........ r83024 | crichter | 2007-09-19 11:32:42 +0200 (Mi, 19 Sep 2007) | 1 line removed comment which violates the coding guidelines. ........ 2007-09-19 00:21 +0000 [r82993] Russell Bryant * /, apps/app_flash.c: Merged revisions 82992 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82992 | russell | 2007-09-18 19:19:49 -0500 (Tue, 18 Sep 2007) | 4 lines Change the description of app_flash to note how it can be a useful tool instead of just saying that it is generally a worthless feature. (Thanks to Jim Van Meggelen for pointing it out and providing the proposed text) ........ 2007-09-18 23:42 +0000 [r82962] Joshua Colp * /, apps/app_queue.c: Merged revisions 82961 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82961 | file | 2007-09-18 20:41:02 -0300 (Tue, 18 Sep 2007) | 2 lines Initialize a variable to NULL to make the world happy. ........ 2007-09-18 22:46 +0000 [r82931] Russell Bryant * include/asterisk/agi.h, /, res/res_agi.c: Merged revisions 82929 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82929 | russell | 2007-09-18 17:42:27 -0500 (Tue, 18 Sep 2007) | 11 lines Add a new patch to handle interrupting the fgets() call when using FastAGI. This version of the patch maintains the original behavior of the code when not using FastAGI. (closes issue #10553) Reported by: juggie Patches: res_agi_fgets-4.patch uploaded by juggie (license 24) res_agi_fgets_1.4svn.patch uploaded by juggie (license 24) Slight mods by me Tested by: juggie, festr ........ 2007-09-18 22:43 +0000 [r82871-82930] Jason Parker * main/pbx.c, main/frame.c, main/dnsmgr.c, channels/chan_local.c, channels/chan_sip.c, res/res_features.c, channels/chan_agent.c, res/res_musiconhold.c, res/res_jabber.c, main/manager.c, res/res_agi.c, channels/chan_features.c, main/logger.c, main/http.c, channels/chan_alsa.c, res/res_realtime.c, res/res_odbc.c: (issue #10724) Reported by: eliel Patches: res_features.c.patch uploaded by eliel (license 64) res_agi.c.patch uploaded by seanbright (license 71) res_musiconhold.c.patch uploaded by seanbright (license 71) pbx.c.patch uploaded by moy (license 222) logger.c.patch uploaded by moy (license 222) frame.c.patch uploaded by moy (license 222) manager.c.patch uploaded by moy (license 222) http.c.patch uploaded by moy (license 222) dnsmgr.c.patch uploaded by moy (license 222) res_realtime.c.patch uploaded by eliel (license 64) res_odbc.c.patch uploaded by seanbright (license 71) res_jabber.c.patch uploaded by eliel (license 64) chan_local.c.patch uploaded by eliel (license 64) chan_agent.c.patch uploaded by eliel (license 64) chan_alsa.c.patch uploaded by eliel (license 64) chan_features.c.patch uploaded by eliel (license 64) chan_sip.c.patch uploaded by eliel (license 64) RollUp.1.patch (includes all of the above patches) uploaded by seanbright (license 71) Convert many CLI commands to the NEW_CLI format. * configs/voicemail.conf.sample, apps/app_voicemail.c: (closes issue #10739) Reported by: ruffle Patches: app_voicemail.c.diff uploaded by ruffle (license 201) 10739-moveheard.diff uploaded by qwell (license 4) Tested by: callguy, ruffle Add an option to disable the automatic moving of "heard" messages to the Old folder. 2007-09-18 20:59 +0000 [r82868] Russell Bryant * main/manager.c, /: Merged revisions 82867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82867 | russell | 2007-09-18 15:56:43 -0500 (Tue, 18 Sep 2007) | 10 lines Fix a memory leak that can occur on systems under higher load. The issue is that when events are appended to the master event queue, they use the number of active sessions as a use count so it will know when all active sessions at the time the event happened have consumed it. However, the handling of the number of sessions was not properly synchronized, so the use count was not always correct, causing an event to disappear early, or get stuck in the event queue for forever. (closes issue #9238, reported by bweschke, patch from Ivan, modified by me) ........ 2007-09-18 20:10 +0000 [r82866] Mark Michelson * /, apps/app_queue.c: Merged revisions 82865 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82865 | mmichelson | 2007-09-18 15:09:02 -0500 (Tue, 18 Sep 2007) | 4 lines Moving the logic for handling an empty membername to the create_member function so that there is a common place where this occurs instead of being spread out to several different places. ........ 2007-09-18 19:06 +0000 [r82835] Kevin P. Fleming * /, apps/app_queue.c: Merged revisions 82834 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82834 | kpfleming | 2007-09-18 13:59:52 -0500 (Tue, 18 Sep 2007) | 2 lines there is no need for conditional logic to select ->interface or ->membername, snince ->membername will always be populated ........ 2007-09-18 16:34 +0000 [r82803] Russell Bryant * /, pbx/pbx_dundi.c: Merged revisions 82802 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82802 | russell | 2007-09-18 11:31:01 -0500 (Tue, 18 Sep 2007) | 4 lines When copying the contents from the wildcard peer, do a deep copy instead of shallow copy so that it doesn't crash when beging destroyed. (closes issue #10546, patch by me) ........ 2007-09-18 16:16 +0000 [r82800] Jason Parker * configs/queues.conf.sample, apps/app_queue.c: (closes issue #10755) Reported by: snar Patches: app-queue-cdr-trunk.patch uploaded by snar (license 245) queues.conf.patch uploaded by snar (license 245) Add an updatecdr option to queues.conf, so that if a "member name" is specified, the cdr record will be updated with that, rather than the channel. 2007-09-18 16:14 +0000 [r82776-82793] Russell Bryant * include/asterisk/threadstorage.h: Make sure that libpthread doesn't try to call free() directly when MALLOC_DEBUG is enabled. If it does, Asterisk will crash as the address isn't the real beginning of the allocation. * channels/chan_zap.c: Don't use ast_channel_lock_both() here, it only exists in one of my branches. This is theoretically a potential deadlock, but it's the way it was before so I'm going to leave it this way for now. 2007-09-18 15:29 +0000 [r82752] Jason Parker * /, configs/sip.conf.sample: Merged revisions 82751 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes issue #10753) ........ r82751 | qwell | 2007-09-18 10:28:21 -0500 (Tue, 18 Sep 2007) | 4 lines Correct the allowexternaldomains option in SIP sample config. Issue 10753 ........ 2007-09-17 22:59 +0000 [r82728] Russell Bryant * channels/chan_local.c, channels/chan_zap.c, apps/app_zapscan.c, channels/chan_agent.c, channels/chan_alsa.c, channels/chan_iax2.c, channels/chan_mgcp.c: convert various places that access the channel lock directly to use the channel lock wrappers 2007-09-17 21:52 +0000 [r82710-82712] Jason Parker * cdr/cdr_sqlite3_custom.c: Don't try to continue loading cdr_sqlite3_custom on a module load failure (such as the config not existing) Closes issue #10749, patch by seanbright. * configs/http.conf.sample: Fix the sample redirect to point to a valid file in the Asterisk GUI. Closes issue #10748, patch by bkruse 2007-09-17 20:24 +0000 [r82595-82679] Russell Bryant * doc/res_config_sqlite.txt, res/res_config_sqlite.c: Add support for #include, var_metric, and cat_metric in res_config_sqlite (closes issue #10738, rbraun_proformatique) * /, main/stdtime/localtime.c, apps/app_voicemail.c: Merged revisions 82676 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82676 | russell | 2007-09-17 15:16:25 -0500 (Mon, 17 Sep 2007) | 4 lines Put a memset in ast_localtime() instead of a couple places in app_voicemail to prevent the problem everywhere instead of just a couple of places. (related to issue #10746) ........ * /, apps/app_voicemail.c: Merged revisions 82644 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82644 | russell | 2007-09-17 15:00:32 -0500 (Mon, 17 Sep 2007) | 6 lines Initialize some memory to fix crashes when leaving voicemail. This problem was fixed by running Asterisk under valgrind. (closes issue #10746, reported by arcivanov, patched by me) *** IMPORTANT NOTE: We need to check to see if this same bug exists elsewhere. ........ * apps/app_dial.c, res/ael/pval.c, include/asterisk/utils.h, apps/app_meetme.c, channels/chan_sip.c, channels/chan_skinny.c, res/res_features.c, apps/app_queue.c, channels/chan_iax2.c, pbx/pbx_config.c: Make the MALLOC_DEBUG output for free() useful again. After changing calls to free to be ast_free, astmm said all calls to free were coming from utils.h * /, res/res_features.c: Merged revisions 82594 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82594 | russell | 2007-09-17 11:46:59 -0500 (Mon, 17 Sep 2007) | 5 lines Handle the case where there are multiple dynamic features with the same digit mapping, but won't always match the activated on/by access controls. In that case, the code needs to keep trying features for a match. (reported by Atis on the asterisk-dev list, patched by me) ........ 2007-09-17 16:44 +0000 [r82593] Kevin P. Fleming * /, apps/app_queue.c: Merged revisions 82590,82592 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82590 | kpfleming | 2007-09-17 11:33:30 -0500 (Mon, 17 Sep 2007) | 2 lines fix a couple of places where a logical member name (if specified) was not used, but instead the direct interface was listed ........ r82592 | kpfleming | 2007-09-17 11:40:12 -0500 (Mon, 17 Sep 2007) | 2 lines revert a change that wasn't supposed to be committed... doh! ........ 2007-09-17 14:58 +0000 [r82568] Doug Bailey * main/http.c: Fix memory leak introduced when POST support was added. 2007-09-17 02:20 +0000 [r82516-82546] Joshua Colp * res/res_features.c: (closes issue #10715) Reported by: the-chopper Don't bother hanging up the new channel if it does not exist yet. * main/pbx.c, /: Merged revisions 82514 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82514 | file | 2007-09-16 23:00:59 -0300 (Sun, 16 Sep 2007) | 4 lines (closes issue #10734) Reported by: asgaroth Instead of passing a NULL pointer into snprintf pass "". It makes Solaris much happier. ........ 2007-09-16 15:32 +0000 [r82496] Tilghman Lesher * apps/app_voicemail.c: Option maxmessage should be maxsecs per-folder, too (closes issue #10729) 2007-09-14 21:30 +0000 [r82457] Steve Murphy * main/cdr.c, /: Merged revisions 82444 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82444 | murf | 2007-09-14 15:19:27 -0600 (Fri, 14 Sep 2007) | 1 line closes issue #10668; thanks to arkadia for his patch; had to leave out the bit about ending the previous cdr in the fork; it would destroy current implementations. ........ 2007-09-14 21:21 +0000 [r82454] Russell Bryant * /, configs/zapata.conf.sample: Merged revisions 82435 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82435 | russell | 2007-09-14 16:17:08 -0500 (Fri, 14 Sep 2007) | 3 lines Add a note to help clarify the value set with the echocancel option. (inspired by Malcolm's blog post on blogs.digium.com about HPEC) ........ 2007-09-14 19:49 +0000 [r82401] Jason Parker * channels/chan_skinny.c, configs/skinny.conf.sample: Add support in chan_skinny for sending RTP directly to the endpoints. Closes issue #9154, patch by DEA 2007-09-14 18:37 +0000 [r82397-82400] Mark Michelson * /: Blocking revision 82398 * /, apps/app_queue.c: Merged revisions 82396 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82396 | mmichelson | 2007-09-14 13:28:36 -0500 (Fri, 14 Sep 2007) | 5 lines Adding member name field to manager events where they were missing before (closes issue #10721, reported by snar) ........ 2007-09-14 17:51 +0000 [r82395] Jason Parker * channels/chan_zap.c, /: Merged revisions 82394 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82394 | qwell | 2007-09-14 12:48:05 -0500 (Fri, 14 Sep 2007) | 5 lines If a channel does not have an owner, do not try to set a channel variable. This will end up making the channel variable global, which is not right. Closes issue #10720, patch by flefoll. ........ 2007-09-14 17:29 +0000 [r82393] Tilghman Lesher * include/asterisk/res_odbc.h, res/res_odbc.c: Add a direct execute method to res_odbc (closes issue #10722) 2007-09-14 16:02 +0000 [r82386-82391] Russell Bryant * channels/xpmr/xpmr.h, channels/xpmr/LICENSE (removed), channels/xpmr/sinetabx.h, channels/xpmr/xpmr.c, channels/xpmr/xpmr_coef.h: use the standard license header for the xpmr files * channels/chan_usbradio.c (added), channels/xpmr (added): Add chan_usbradio to trunk * /, build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Merged revisions 82385 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82385 | russell | 2007-09-14 10:50:49 -0500 (Fri, 14 Sep 2007) | 3 lines Add checking for libusb here, so nobody has to deal with conflicts in the chan_usbradio-1.4 branch every time the configure script gets changed ........ 2007-09-14 14:44 +0000 [r82377] Mark Michelson * doc/CODING-GUIDELINES, /: Merged revisions 82376 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82376 | mmichelson | 2007-09-14 09:42:29 -0500 (Fri, 14 Sep 2007) | 5 lines Fixing a typo in the coding guidelines (closes issue #10717, reported and patched by leedm777) ........ 2007-09-14 13:02 +0000 [r82373] Philippe Sultan * channels/chan_jingle.c: Fix DTMF following what has been done in issue #9401. Thanks irroot. 2007-09-13 23:12 +0000 [r82359] Jason Parker * pbx/pbx_spool.c, /: Merged revisions 82358 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82358 | qwell | 2007-09-13 18:11:27 -0500 (Thu, 13 Sep 2007) | 4 lines Fix a small typo. retrytime > waittime ........ 2007-09-13 21:53 +0000 [r82347-82352] Mark Michelson * apps/app_queue.c: Changed "in" to "queue" in "queue {pause|unpause} member" command to be more clear. Also added check to be sure that sixth argument is the word "reason" if full command is given * CHANGES, apps/app_queue.c: Added the ability to pause and unpause members via the CLI * /, apps/app_queue.c: Merged revisions 82346 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82346 | mmichelson | 2007-09-13 15:16:37 -0500 (Thu, 13 Sep 2007) | 4 lines Preemptively fixing a possible segfault. It is possible that queuename is NULL (meaning pause ALL queues), so use q->name instead. ........ 2007-09-13 20:13 +0000 [r82345] Jason Parker * /, cdr/cdr_csv.c: Merged revisions 82344 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82344 | qwell | 2007-09-13 15:11:40 -0500 (Thu, 13 Sep 2007) | 9 lines Fix a crash that could occur in cdr_csv when mutliple threads tried to close the same file. Do we actually need the locking here? What happens if you open the same file twice, and two threads try to write to it at the same time? Is fputs() going to write out the entire line at once? I suspect that it could be possible for the second fopen to run during the first fputs, so the position could be in the middle of the previously written line... Issue 10347, initial patch by explidous (but I removed all of the paranoia stuff..) ........ 2007-09-13 19:16 +0000 [r82338-82341] Russell Bryant * /, main/astobj2.c: Merged revisions 82339 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82339 | russell | 2007-09-13 13:57:08 -0500 (Thu, 13 Sep 2007) | 1 line resolve a warning when not building under dev mode ........ * include/asterisk.h, /, main/astobj2.c, main/asterisk.c: Merged revisions 82337 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82337 | russell | 2007-09-13 13:45:59 -0500 (Thu, 13 Sep 2007) | 4 lines Only compile in tracking astobj2 statistics if dev-mode is enabled. Also, when dev mode is enabled, register the CLI command that can be used to run the astobj2 test and print out statistics. ........ 2007-09-13 18:13 +0000 [r82336] Kevin P. Fleming * /, LICENSE: Merged revisions 82335 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r82335 | kpfleming | 2007-09-13 11:12:00 -0700 (Thu, 13 Sep 2007) | 10 lines Merged revisions 82334 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r82334 | kpfleming | 2007-09-13 11:10:12 -0700 (Thu, 13 Sep 2007) | 2 lines clarify the OpenSSL and OpenH323 license exceptions ........ ................ 2007-09-13 16:58 +0000 [r82329] Joshua Colp * channels/chan_zap.c, CHANGES, configs/zapata.conf.sample: Add setvar support to chan_zap. Just like you can in chan_sip and chan_iax2 you can now use it with zaptel channels. (done while in Montreal at the Asterisk bootcamp!) 2007-09-13 16:27 +0000 [r82327] Mark Michelson * /, apps/app_queue.c: Merged revisions 82326 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82326 | mmichelson | 2007-09-13 11:25:59 -0500 (Thu, 13 Sep 2007) | 7 lines Added logic to handle the unlikely case that someone has two queues with the same name. Asterisk will log a warning message letting the user know that one was already defined with that name and is it skipping all further instances. This also will work for realtime queues but in order for that to happen, the user would have to trigger a perfectly timed reload as a realtime queue is being looked up, which is highly unlikely (but taken care of nonetheless). ........ 2007-09-13 15:26 +0000 [r82321] Russell Bryant * include/asterisk/doxyref.h, doc/res_config_sqlite.txt, res/res_config_sqlite.c, configs/res_config_sqlite.conf: Various code and documentation cleanups for res_config_sqlite (closes issue #10711, rbraun_proformatique) 2007-09-13 15:25 +0000 [r82312-82320] Philippe Sultan * channels/chan_jingle.c: Modify rule filters to match with the Jingle namespace constant * include/asterisk/jingle.h: Assign namespace properly * channels/chan_jingle.c, include/asterisk/jingle.h: Changed Jingle and Jingle DTMF namespaces. As both specifications are in the Experimental status, the namespaces specified therein shall be of the form "http://www.xmpp.org/extensions/xep-XXXX.html#ns". See the Namespace issuance section in XEP-0053 : http://www.xmpp.org/extensions/xep-0053.html#namespaces * channels/chan_jingle.c: Reflect Jingle DTMF specification changes 2007-09-13 13:34 +0000 [r82311] Russell Bryant * apps/app_queue.c: Fix a missing unref of a member struct. This was pointed out by Marta. Thanks! This function in 1.4 didn't have the problem. 2007-09-13 11:54 +0000 [r82310] Philippe Sultan * /, channels/chan_gtalk.c: Merged revisions 82309 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82309 | phsultan | 2007-09-13 13:47:14 +0200 (Thu, 13 Sep 2007) | 4 lines Closes issue #9401, reported and patched by irrot, with slight modifications by me. Handle DTMF sent by Asterisk properly. ........ 2007-09-12 21:57 +0000 [r82297] Russell Bryant * /, res/res_agi.c: Merged revisions 82296 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82296 | russell | 2007-09-12 16:56:32 -0500 (Wed, 12 Sep 2007) | 3 lines Fix a check of the wrong pointer, as pointed out by an XXX comment left in the code. The problem was harmless, however. ........ 2007-09-12 21:55 +0000 [r82294] Jason Parker * channels/chan_iax2.c: After some discussions, we decided that the return values here were a bit messy. This also fixes a bug on reload, where peers may not have reregistered properly. 2007-09-12 21:32 +0000 [r82290-82292] Tilghman Lesher * /, main/stdtime/tzfile.h: Merged revisions 82291 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82291 | tilghman | 2007-09-12 16:28:33 -0500 (Wed, 12 Sep 2007) | 2 lines Oops, wrong location for FreeBSD zone files ........ * main/stdtime/private.h, /, main/stdtime/tzfile.h, funcs/func_strings.c, apps/app_sms.c, include/asterisk/localtime.h, main/stdtime/localtime.c: Merged revisions 82285 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82285 | tilghman | 2007-09-12 15:12:06 -0500 (Wed, 12 Sep 2007) | 4 lines Working on issue #10531 exposed a rather nasty 64-bit issue on ast_mktime, so we updated the localtime.c file from source. Next we'll have to write ast_strptime to match. ........ 2007-09-12 21:17 +0000 [r82289] Mark Michelson * apps/app_queue.c: Removed an unneeded ao2_ref. This was a problem because unless get_member_status returned QUEUE_NORMAL, a NULL member would be unreferenced. While this didn't cause any crashes or anything terrible, it still is incorrect 2007-09-12 20:50 +0000 [r82288] Steve Murphy * main/config.c: This fix closes issue #10642 -- it's not perfect, but should retain most blank lines in config files, via read/write cycles. 2007-09-12 20:47 +0000 [r82287] Dwayne M. Hubbard * /, apps/app_meetme.c: Merged revisions 82286 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82286 | dhubbard | 2007-09-12 15:24:24 -0500 (Wed, 12 Sep 2007) | 1 line remove a race condition for the creation of recordthread's, and fix a small memory leak. This closes issue# 10636 ........ 2007-09-12 16:24 +0000 [r82283] Mark Michelson * main/pbx.c, main/app.c, main/asterisk.c: Fixes Solaris build warnings (closes issue #10698, reported and patched by snuffy) 2007-09-12 15:53 +0000 [r82279-82282] Russell Bryant * utils/hashtest2.c: Change the traversal to use ao2_callback() instead of an ao2_iterator. Using ao2_callback() is a much more efficient way of performing an operation on every item in the container. This change makes hashtest2 run in about 25% of the time it ran before on my system. In general, I would say that it makes the most sense to use an ao2_iterator if the operation being performed is going to take a long time and you don't want to keep the container locked while you work with each object. Otherwise, the use of ao2_callback is preferred. * /, main/asterisk.c: Merged revisions 82280 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82280 | russell | 2007-09-12 10:16:49 -0500 (Wed, 12 Sep 2007) | 4 lines Clean up the output of "asterisk -h". This tweaks the wording and wraps lines at 80 characters. (closes issue #10699, seanbright) ........ * /, res/res_agi.c: Merged revisions 82278 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82278 | russell | 2007-09-12 10:11:11 -0500 (Wed, 12 Sep 2007) | 3 lines revert patch from issue #10553, as someone not using fastagi reported that this broke their system. ........ 2007-09-12 14:31 +0000 [r82275-82277] Mark Michelson * /: Blocking changes from revision 82276 * /, apps/app_queue.c: Merged revisions 82274 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82274 | mmichelson | 2007-09-12 09:24:53 -0500 (Wed, 12 Sep 2007) | 6 lines We should only initialize a realtime queue when it is allocated, not every time we access it. This prevents the members ao2_container from being reallocated every time the queue is accessed. I also removed a debug message I had accidentally left in on a previous commit. ........ 2007-09-11 23:07 +0000 [r82273] Matthew Fredrickson * channels/chan_zap.c: Fix to make sure we don't hangup a call when getting a RLC without sending REL. Found making sure we are Q.784 (the SS7 test specification) compliant 2007-09-11 22:38 +0000 [r82269-82270] Russell Bryant * main/config.c: remove unused functions that made this file not build under dev mode * /, apps/app_queue.c: Merged revisions 82267 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82267 | russell | 2007-09-11 17:37:17 -0500 (Tue, 11 Sep 2007) | 3 lines Fix incorrect uses of ao2_find(). Every one of these calls was reading bogus memory ... ........ 2007-09-11 22:37 +0000 [r82268] Steve Murphy * utils/Makefile, main/config.c: This solves an unreported solaris compile problem (missing -lnsl -lsocket). 2007-09-11 21:43 +0000 [r82266] Joshua Colp * /, codecs/gsm/src/long_term.c, codecs/gsm/src/lpc.c: Merged revisions 82265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82265 | file | 2007-09-11 18:41:49 -0300 (Tue, 11 Sep 2007) | 4 lines (closes issue #10679) Reported by: andrew Build under dev mode when K6OPTS is enabled. ........ 2007-09-11 20:50 +0000 [r82264] Russell Bryant * /, apps/app_queue.c: Merged revisions 82263 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82263 | russell | 2007-09-11 15:49:34 -0500 (Tue, 11 Sep 2007) | 5 lines Fix another missing unref of member objects. This one was pointed out by Marta. When building the outgoing list in try_calling(), a member reference is stored in each outgoing entry. However, when this list got destroyed, the reference was not released. ........ 2007-09-11 20:49 +0000 [r82262] Steve Murphy * main/cdr.c, /: Merged revisions 82261 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82261 | murf | 2007-09-11 14:36:15 -0600 (Tue, 11 Sep 2007) | 1 line this change should fix issue # 10659 -- what I worry about is how many other bug reports it may generate. Hopefully, we can please the/a majority. Hopefully. We shall see. Calls not marked ANSWERED and with only one channel name will not be posted. This should eliminate the double CDR's. ........ 2007-09-11 18:37 +0000 [r82257-82258] Joshua Colp * configs/sip.conf.sample: Lil' bit more documentation to keep folks happy. * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: (closes issue #9433) Reported by: junky Patches: register_trying.diff.txt uploaded by jcmoore Disable sending 100 Trying on REGISTER attempts and make it an option. This has been signed off by oej. 2007-09-11 17:16 +0000 [r82256] Steve Murphy * utils/Makefile: fixing up the pthread stuff for hashtest2 2007-09-11 16:15 +0000 [r82254] Christian Richter * channels/chan_misdn.c, channels/misdn/isdn_lib.c: Merged revisions 82249 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82249 | crichter | 2007-09-11 18:01:27 +0200 (Di, 11 Sep 2007) | 1 line fixed a hold/retrieve issue. ........ 2007-09-11 16:12 +0000 [r82253] Mark Michelson * /, apps/app_queue.c: Merged revisions 82252 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82252 | mmichelson | 2007-09-11 11:05:56 -0500 (Tue, 11 Sep 2007) | 6 lines All instances of ao2_iterators which were just named 'i' have been renamed to 'mem_iter' so that when refcounted queues are merged into trunk, there will be little confusion regarding iterator names, especially when a queue and member iterator are used in the same function. ........ 2007-09-11 16:05 +0000 [r82251] Russell Bryant * /, pbx/pbx_dundi.c: Merged revisions 82250 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82250 | russell | 2007-09-11 11:03:42 -0500 (Tue, 11 Sep 2007) | 4 lines The sample dundi.conf claims support for a wildcard peer entry - [*], but the code did not support it. This patch makes it work. (closes issue #10546, patch by dds, with some changes by me) ........ 2007-09-11 15:34 +0000 [r82248] Joshua Colp * main/cdr.c: (closes issue #10666) Reported by: arkadia Patches: cdr_lockorder.patch uploaded by arkadia (license 233) Optimize CDR stuff a bit. 2007-09-11 15:31 +0000 [r82246-82247] Russell Bryant * res/res_agi.c: Remove an unused variable. I have no idea why this was marked with the unused attribute instead of just removing it. :) * /, res/res_agi.c: Merged revisions 82245 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82245 | russell | 2007-09-11 10:26:51 -0500 (Tue, 11 Sep 2007) | 9 lines (closes issue #10553) Reported by: juggie Patches: res_agi_fgets-2.patch uploaded by juggie (license 24) Tested by: juggie When using fastagi, fgets() can return before a full line is read. Add explicit handling for the case where it gets interrupted. ........ 2007-09-11 14:58 +0000 [r82242-82244] Joshua Colp * /, pbx/pbx_dundi.c: Merged revisions 82243 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82243 | file | 2007-09-11 11:56:39 -0300 (Tue, 11 Sep 2007) | 6 lines (closes issue #10577) Reported by: jamesgolovich Patches: asterisk-dundifree.diff.txt uploaded by jamesgolovich (license 176) Don't leak memory when unloading DUNDi. ........ * apps/app_meetme.c: (closes issue #10560) Reported by: ruffle Patches: rb uploaded by ruffle (license 201) Show whether the conference is locked or not on the CLI. 2007-09-11 14:35 +0000 [r82237-82241] Russell Bryant * /, apps/app_queue.c: Merged revisions 82240 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82240 | russell | 2007-09-11 09:34:12 -0500 (Tue, 11 Sep 2007) | 2 lines Add a couple more missing unrefs of queue member objects ........ * /, apps/app_queue.c: Merged revisions 82238 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82238 | russell | 2007-09-11 09:21:17 -0500 (Tue, 11 Sep 2007) | 2 lines Add a missing unref of a queue member in an error handling block ........ * /, apps/app_queue.c: Merged revisions 82236 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82236 | russell | 2007-09-11 09:09:43 -0500 (Tue, 11 Sep 2007) | 2 lines Document why membercount can not simply be replaced by ao2_container_count() ........ 2007-09-11 13:46 +0000 [r82231-82235] Joshua Colp * utils/Makefile: Include string compatibility file in hashtest2. * utils/hashtest2.c: Include compat.h to hopefully make it compatible with FreeBSD. * utils/hashtest2.c: Fix building under FreeBSD. Make sure alloca.h exists before including it. * main/manager.c: (closes issue #10695) Reported by: junky Patches: count_showconn.diff uploaded by junky (license 177) Provide a count of connected users to manager. * main/minimime/minimime.c, main/minimime/tests/create.c, main/minimime/mm_mem.c, main/minimime/tests/parse.c: (closes issue #10692) Reported by: snuffy Patches: minivm.diff uploaded by snuffy (license 35) Instead of using err (which is not available under Solaris) use fdprintf with stderr. 2007-09-10 20:03 +0000 [r82200] Tilghman Lesher * UPGRADE.txt, channels/chan_iax2.c: Change the IAXPeers command to have manager-style output, instead of CLI-style output (closes issue #8254) 2007-09-10 19:10 +0000 [r82185] Mark Michelson * apps/app_queue.c: Fixing a problem where NULL channels would cause a crash when calling indisposed queue members (i.e. paused, wrapup time not completed, etc.) 2007-09-10 18:32 +0000 [r82178] Tilghman Lesher * /, apps/app_queue.c: Merged revisions 82155 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82155 | tilghman | 2007-09-10 13:02:02 -0500 (Mon, 10 Sep 2007) | 2 lines Convert struct member to use refcounts (closes issue #10199) ........ 2007-09-10 17:39 +0000 [r82154] Jason Parker * main/db.c: Add a counter to the 'database deltree' CLI command. Note: this is slightly different than the initial patch, because I felt that using res <= 0 would be a change in behavior. Closes issue #10687, patch by junky 2007-09-10 16:59 +0000 [r82140] Steve Murphy * utils/Makefile, utils/hashtest2.c (added): Committing my test for astobj2, hashtest2.c, along with makefile changes in utils. 2007-09-10 16:24 +0000 [r82125] Jason Parker * main/db.c: Add counter to 'database show' CLI command. (also a minor whitespace change that I found along the way) Closes issue #10683, patch by junky 2007-09-10 16:19 +0000 [r82124] Steve Murphy * main/astobj2.c: Changes applied from marta's team/marta/astobj2 branch to solve a race condition 2007-09-10 15:05 +0000 [r82092] Mark Michelson * /, configs/misdn.conf.sample: Merged revisions 82091 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82091 | mmichelson | 2007-09-10 10:02:12 -0500 (Mon, 10 Sep 2007) | 5 lines Removing non-existent options from misdn configuration sample. (closes issue #10678, reported and patched by IgorG) ........ 2007-09-10 14:26 +0000 [r82062-82077] Joshua Colp * channels/chan_sip.c: (closes issue #10688) Reported by: casper Patches: chan_sip.c.82076.diff uploaded by casper (license 55) Remove double check for zombie flag and optimize things a bit. * res/res_agi.c: (closes issue #10684) Reported by: junky Patches: debug.diff uploaded by junky (license 177) Fix issue with debug always showing up. * apps/app_meetme.c: (closes issue #10686) Reported by: junky Patches: meet.diff uploaded by junky (license 177) Change NOTICE message to DEBUG. 2007-09-09 02:45 +0000 [r82029] Tilghman Lesher * /, include/asterisk/lock.h: Merged revisions 82028 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r82028 | tilghman | 2007-09-08 21:35:18 -0500 (Sat, 08 Sep 2007) | 2 lines Fix inline compiles on really old compilers (who uses gcc 2.7 anymore, really?) (closes issue #10675) ........ 2007-09-08 19:01 +0000 [r81998-81999] Russell Bryant * include/asterisk/slinfactory.h: Add doxygen documentation for slinfactory_destroy(), mainly just noting that it doesn't free the slinfactory itself. (This isn't related to a bug, i'm just looking over random code) * /, main/asterisk.c: Merged revisions 81997 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81997 | russell | 2007-09-08 13:41:32 -0500 (Sat, 08 Sep 2007) | 2 lines Fix a small memory leak. ast_unregister_atexit() did not free the entry it removed. ........ 2007-09-08 16:37 +0000 [r81984] Mark Michelson * apps/app_voicemail.c: Make Callerid more consistent in IMAP mail headers (closes issue #10056, reported and patched by jaroth, with small modification by me) 2007-09-08 13:45 +0000 [r81953] Russell Bryant * /, .cleancount: Merged revisions 81952 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81952 | russell | 2007-09-08 08:42:26 -0500 (Sat, 08 Sep 2007) | 11 lines (closes issue #10672) Bump the cleancount so that a "make clean" will be forced. This is needed because my fix in revision 81599 made a change to a data structure in file.h, and since file dependency tracking is only on with dev-mode enabled, file format modules that don't get rebuilt may crash, as is the case with this issue. This makes me wonder - how much faster does the code build without the file dependency tracking enabled? If it doesn't make much of a difference, then it may be worth just keeping it on all of the time, or perhaps just not in release tarballs, so that this type of issue is avoided. ........ 2007-09-07 19:53 +0000 [r81910-81924] Jason Parker * /, apps/app_queue.c: Merged revisions 81923 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #10671) ........ r81923 | qwell | 2007-09-07 14:48:00 -0500 (Fri, 07 Sep 2007) | 5 lines Allow the MEMBERINTERFACE variable to be used as the mixmonitor filename. This moves the setting of the MEMBERINTERFACE variable to before mixmonitor. Issue 10671, patch by sim. ........ * apps/app_queue.c: Add an optional reason parameter to PauseQueueMember/UnpauseQueueMember applications and manager events. Issue 8738, patch by rgollent 2007-09-07 15:29 +0000 [r81891] Mark Michelson * /, configs/queues.conf.sample: Merged revisions 81886 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81886 | mmichelson | 2007-09-07 10:25:19 -0500 (Fri, 07 Sep 2007) | 3 lines Moving the explanation for joinempty to a more appropriate place ........ 2007-09-07 12:32 +0000 [r81858-81873] Joshua Colp * configure, configure.ac: Don't check for epoll support when cross compiling. * main/channel.c, main/audiohook.c: Fix memory issue that crept up with Russell's testing. It is *not* proper to free the frame we get in ast_write. 2007-09-06 22:32 +0000 [r81839-81849] Russell Bryant * channels/chan_sip.c: fix the build ... oops * /, channels/chan_sip.c: Merged revisions 81832 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81832 | russell | 2007-09-06 17:28:57 -0500 (Thu, 06 Sep 2007) | 16 lines (closes issue #9724, closes issue #10374) Reported by: kenw Patches: 9724.txt uploaded by russell (license 2) Tested by: kenw, russell Resolve a deadlock that occurs when doing a SIP transfer to parking. I come across this type of deadlock fairly often it seems. It is very important to mind the boundary between the channel driver and the core in respect to the channel lock and the channel-pvt lock. Channel drivers lock to lock the pvt and then the channel once it calls into the core, while the core will do it in the opposite order. The way this is avoided is by having channel drivers either release their pvt lock while calling into the core, or such as in this case, unlocking the pvt just long enough to acquire the channel lock. ........ 2007-09-06 22:06 +0000 [r81827] Jason Parker * Makefile, /: Merged revisions 81826 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81826 | qwell | 2007-09-06 17:05:02 -0500 (Thu, 06 Sep 2007) | 1 line We added COPTS for ASTCFLAGS additions, but not LDOPTS for ASTLDFLAGS. This adds LDOPTS ........ 2007-09-06 21:01 +0000 [r81814] Joshua Colp * channels/iax2-parser.c: Initialize iax_frames variable to NULL, keeps valgrind happy. 2007-09-06 20:54 +0000 [r81783-81813] Russell Bryant * CHANGES, funcs/func_extstate.c (added): Add EXTENSION_STATE() function that can retrieve the state of an extension that has a hint. (closes issue #10635, adamgundy) * CHANGES: s/DEVSTATE/DEVICE_STATE/ * funcs/func_devstate.c: Rename the DEVSTATE() function to DEVICE_STATE() to better conform to how other functions are named. (inspired by issue #10635) * CHANGES, funcs/func_devstate.c: Merge HINT() dialplan function from my sandbox branch into trunk. This function will let you retrieve the list of devices or name associated with a hint. (inspired by issue #10635) 2007-09-06 20:16 +0000 [r81782] Joshua Colp * channels/chan_skinny.c, CHANGES: (closes issue #10377) Reported by: mvanbaak Patches: chan_skinny_info.diff uploaded by mvanbaak (license 7) Add skinny show device, skinny show line, and skinny show settings CLI commands. 2007-09-06 20:05 +0000 [r81781] Russell Bryant * configs/extensions.conf.sample: Fix the syntax of declaring a hint with a name to be compatible with trunk 2007-09-06 20:00 +0000 [r81779] Jason Parker * /, include/asterisk/astobj2.h: Merged revisions 81778 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81778 | qwell | 2007-09-06 14:59:07 -0500 (Thu, 06 Sep 2007) | 2 lines This should fix a build issue that people building against uClibc were seeing with the addition of astobj2 ........ 2007-09-06 19:43 +0000 [r81777] Joshua Colp * /, apps/app_meetme.c: Merged revisions 81776 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81776 | file | 2007-09-06 16:40:37 -0300 (Thu, 06 Sep 2007) | 7 lines (closes issue #10122) Reported by: stevefeinstein Patches: meetme-unmute-manager.diff uploaded by qwell (license 4) Tested by: stevefeinstein After looking over the code I agree with Qwell. Setting the file descriptor to conference each time just causes a fight back and forth. ........ 2007-09-06 17:00 +0000 [r81745] Philippe Sultan * /, include/asterisk/jabber.h, channels/chan_gtalk.c: Merged revisions 81743 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81743 | phsultan | 2007-09-06 18:56:29 +0200 (Thu, 06 Sep 2007) | 1 line Various string length fixes. Removed an unused variable in aji_client structure (context) ........ 2007-09-06 16:57 +0000 [r81744] Tilghman Lesher * contrib/scripts/safe_asterisk: Incorporate the ability to log output of safe_asterisk to syslog (closes issue #9882) 2007-09-06 16:38 +0000 [r81742] Matthew Fredrickson * channels/chan_zap.c: Patch on 10575. Add support for unequipped CIC (UCIC) message as well as improve some of our CIC flags in chan_zap 2007-09-06 16:31 +0000 [r81730] Mark Michelson * /, apps/app_queue.c: Merged revisions 81713 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81713 | mmichelson | 2007-09-06 11:25:40 -0500 (Thu, 06 Sep 2007) | 6 lines Fixes an issue where valid DTMF had to be pressed twice to exit a queue if a member's phone was ringing. (closes issue #10655, reported by strider2k, patched by me) ........ 2007-09-06 15:43 +0000 [r81712] Luigi Rizzo * include/asterisk/astobj2.h, main/astobj2.c: various changes to the documentation, and redefinition of ao2_hash_fn and ao2_callback_fn typedefs, in preparation to more cleanup of the _search_flags Please do not merge this change to 1.4 yet - there are no functional changes anyways. 2007-09-06 15:21 +0000 [r81683] Mark Michelson * /, res/res_features.c: Merged revisions 81682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81682 | mmichelson | 2007-09-06 10:20:36 -0500 (Thu, 06 Sep 2007) | 5 lines Fixes a memory leak (closes issue #10658, reported and patched by Ivan) ........ 2007-09-06 14:24 +0000 [r81651] Philippe Sultan * /, res/res_jabber.c: Merged revisions 81650 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81650 | phsultan | 2007-09-06 16:20:54 +0200 (Thu, 06 Sep 2007) | 3 lines According to both RFC 3920 - section 9.1.2 - and Google's XMPP server complaint, if set, the 'from' attribute must be set to the user's full JID. ........ 2007-09-05 21:59 +0000 [r81632] Mark Michelson * apps/app_queue.c: Not having this epoll specific code in wait_for_answer was causing app_queue to infinitely loop. This makes it so it doesn't. Thanks to file for pointing out where the problem was and showing a similar function in app_dial as an example of how to fix it. 2007-09-05 21:45 +0000 [r81631] Tilghman Lesher * /, include/asterisk/lock.h: Merged revisions 81569 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81569 | tilghman | 2007-09-05 12:18:24 -0500 (Wed, 05 Sep 2007) | 2 lines Solaris x86 compatibility fix ........ 2007-09-05 20:58 +0000 [r81601] Dwayne M. Hubbard * apps/app_zapateller.c: added ZAPATELLERSTATUS to app_zapateller 2007-09-05 20:58 +0000 [r81600] Russell Bryant * include/asterisk/file.h, /, main/say.c, res/res_features.c, main/file.c, include/asterisk/channel.h: Merged revisions 81599 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81599 | russell | 2007-09-05 15:53:41 -0500 (Wed, 05 Sep 2007) | 11 lines Fix an issue that can occur when you do an attended transfer to parking. If you complete the transfer before the announcement of the parking spot finishes, then the channel being parked will hear the remainder of the announcement. These changes make it so that will not happen anymore. Basically, res_features sets a flag on the channel is playing the announcement to so that the file streaming core knows that it needs to watch out for a channel masquerade, and if it occurs, to abort the announcement. (closes BE-182) ........ 2007-09-05 16:48 +0000 [r81568] Tilghman Lesher * utils: Add two more generated files (requested by mvanbaak via irc) 2007-09-05 16:31 +0000 [r81560] Jason Parker * include/asterisk/devicestate.h, res/res_config_odbc.c, channels/chan_sip.c, include/asterisk/audiohook.h, main/sha1.c, res/res_features.c, include/asterisk/astobj2.h, res/res_crypto.c, include/asterisk/strings.h, main/audiohook.c, res/res_jabber.c, res/res_config_sqlite.c, include/asterisk/sha1.h, include/asterisk/stringfields.h, include/asterisk/features.h: Doxygen cleanups/fixes. Closes issue #10654, patch by snuffy 2007-09-05 15:32 +0000 [r81526-81535] Mark Michelson * apps/app_queue.c: Weird. When I merged my changes from 1.4, they merged into the wrong function. This should fix the build for trunk. * /, apps/app_queue.c: Merged revisions 81525 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81525 | mmichelson | 2007-09-05 10:19:47 -0500 (Wed, 05 Sep 2007) | 4 lines Fixing the build... ........ 2007-09-05 15:16 +0000 [r81524] Jason Parker * channels/chan_phone.c, /: Merged revisions 81523 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #10651) ........ r81523 | qwell | 2007-09-05 10:14:30 -0500 (Wed, 05 Sep 2007) | 5 lines Do not try to unregister a NULL channel tech. Also changed load_module function to use defines rather than numbers for return values. Issue 10651, patch by rbraun_proformatique, with additions by me. ........ 2007-09-05 15:04 +0000 [r81522] Mark Michelson * /, apps/app_queue.c: Merged revisions 81520 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81520 | mmichelson | 2007-09-05 10:03:22 -0500 (Wed, 05 Sep 2007) | 6 lines Reverting behavior of QUEUE_MEMBER_COUNT to only count members who are logged in and available. (related to issue #10652, reported by wuwu) ........ 2007-09-05 14:47 +0000 [r81519] Steve Murphy * include/asterisk/config.h, main/config.c: this set of changes fixes issue # 10643 by keeping track of the last object defined in a file, and attaching any accumulated comments to that object (category header or variable declaration). The file_save routine also had to be upgraded to output these trailing comments. Config.h was modified to include the trailing comment list on categories and variables. 2007-09-05 13:13 +0000 [r81459-81493] Joshua Colp * main/editline/sys.h: Finish up commit from revision 81452 by removing last remnants of strlcat/strlcpy checks. 2007-09-04 20:59 +0000 [r81454-81456] Jason Parker * /, apps/app_followme.c: Merged revisions 81455 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #10634) ........ r81455 | qwell | 2007-09-04 15:54:51 -0500 (Tue, 04 Sep 2007) | 4 lines Rather than attempt to play a file, we can just check whether it exists. Issue 10634, patch by me, testing by pabelanger, sanity checked by bweschke ........ * /, configs/followme.conf.sample: Merged revisions 81453 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #10644) ........ r81453 | qwell | 2007-09-04 14:56:06 -0500 (Tue, 04 Sep 2007) | 4 lines Change default followme config file to point to the correct files. Issue 10644, patch by pabelanger ........ 2007-09-04 19:51 +0000 [r81445-81452] Russell Bryant * main/editline/configure, main/editline/configure.in: Don't check for and include strlcpy and strlcat in editline. We also include them directly in Asterisk. For platforms that need them (like my mac), you will get a linker error due to the functions being included twice. * /, include/asterisk/astobj2.h, channels/chan_iax2.c, main/astobj2.c: Merged revisions 81448 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81448 | russell | 2007-09-04 13:37:44 -0500 (Tue, 04 Sep 2007) | 4 lines Remove the typedefs on ao2_container and ao2_iterator. This is simply because we don't typedef objects anywhere else in Asterisk, so we might as well make this follow the same convention. ........ * include/asterisk/logger.h: logger.h depends on options.h, so go ahead and include it 2007-09-04 16:41 +0000 [r81443] Kevin P. Fleming * /, channels/chan_sip.c: Merged revisions 81442 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81442 | kpfleming | 2007-09-04 11:40:39 -0500 (Tue, 04 Sep 2007) | 2 lines there is no point in sending 401 Unauthorized to a UAS that sent us a properly-formatted Authentication header with the expected username and nonce but an incorrect response (which indicates the shared secret does not match)... instead, let's send 403 Forbidden so that the UAS doesn't retry with the same authentication credentials repeatedly ........ 2007-09-04 14:28 +0000 [r81436-81441] Joshua Colp * configs/extensions.ael.sample: (closes issue #10633) Reported by: pabelanger Patches: extensions.ael.sample.patch uploaded by pabelanger (license 224) Update extensions.ael.sample with voicemail and | changes. * /, channels/chan_iax2.c: Merged revisions 81439 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81439 | file | 2007-09-04 11:23:18 -0300 (Tue, 04 Sep 2007) | 6 lines (closes issue #10632) Reported by: jamesgolovich Patches: asterisk-iaxfirmwareleak.diff.txt uploaded by jamesgolovich (license 176) Fix memory leak when unloading chan_iax2. The firmware files were not being freed. ........ * main/channel.c, /: Merged revisions 81437 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81437 | file | 2007-09-04 10:46:23 -0300 (Tue, 04 Sep 2007) | 4 lines (closes issue #10476) Reported by: mdu113 Only look for the end of a digit when waiting for a digit. This in turn disables emulation in the core. ........ * /, main/dns.c: Merged revisions 81435 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81435 | file | 2007-09-04 10:10:56 -0300 (Tue, 04 Sep 2007) | 7 lines (closes issue #10610) Reported by: john Patches: dns.c.patch uploaded by john (license 218) Tested by: mvanbaak Don't return a match if no SRV record actually exists. ........ 2007-09-03 18:59 +0000 [r81434] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 81433 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81433 | russell | 2007-09-03 13:57:53 -0500 (Mon, 03 Sep 2007) | 5 lines Remove a couple of calls to ast_string_field_free_pools() on peers in error handling blocks in the code for building peers. The peer object destructor does this and doing it twice will cause a crash. (closes issue #10625, reported by and patched by pnlarsson) ........ 2007-09-03 18:01 +0000 [r81430-81432] Tilghman Lesher * main/config.c: Once we get past the file checks, we're loading, so clear the FILEUNCHANGED flag (fixes #include) (closes issue #10629) * /, funcs/func_logic.c: Merged revisions 81415 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81415 | tilghman | 2007-08-31 14:16:52 -0500 (Fri, 31 Aug 2007) | 2 lines The IF() function was not allowing true values that had embedded colons (closes issue #10613) ........ * main/config.c: We shouldn't use a filename blindly without checking to make sure it's unused first 2007-09-01 06:03 +0000 [r81427] Mark Michelson * /, include/asterisk/astobj2.h, main/astobj2.c: Merged revisions 81426 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81426 | mmichelson | 2007-09-01 01:02:06 -0500 (Sat, 01 Sep 2007) | 4 lines Making match_by_addr into ao2_match_by_addr and making it available everywhere since it could be a handy callback to have ........ 2007-08-31 21:29 +0000 [r81419] Russell Bryant * /, include/asterisk/astobj2.h: Merged revisions 81418 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81418 | russell | 2007-08-31 16:27:49 -0500 (Fri, 31 Aug 2007) | 2 lines Remove references to a debugging parameter that does not exist ........ 2007-08-31 19:50 +0000 [r81417] Mark Michelson * /, apps/app_queue.c: Merged revisions 81416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81416 | mmichelson | 2007-08-31 14:48:55 -0500 (Fri, 31 Aug 2007) | 6 lines Fixed broken behavior of a reload on realtime queues. Prior to this patch, if a reload was issued and a realtime queue had callers waiting in it, then the queue would be removed from the queue list, but it would not actually be freed (in fact, a debug message warning about a memory leak would come up). With this patch, reloads do not touch realtime queues at all. ........ 2007-08-31 18:46 +0000 [r81413] Jason Parker * apps/app_dial.c, /: Merged revisions 81412 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #10621) ........ r81412 | qwell | 2007-08-31 13:44:44 -0500 (Fri, 31 Aug 2007) | 4 lines Re-order dial options to be in line with the existing alpha order. Issue 10621, initial patch by junky ........ 2007-08-31 17:43 +0000 [r81411] Philippe Sultan * /, channels/chan_gtalk.c: Merged revisions 81410 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81410 | phsultan | 2007-08-31 19:38:26 +0200 (Fri, 31 Aug 2007) | 3 lines Make the 'gtalk show channels' CLI command available. Closes issue 10548, reported by keepitcool. ........ 2007-08-31 15:58 +0000 [r81408] Kevin P. Fleming * /, codecs/codec_zap.c: Merged revisions 81405 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81405 | kpfleming | 2007-08-31 10:51:45 -0500 (Fri, 31 Aug 2007) | 2 lines add missing "transcoder show" (and deprecated "show transcoder") CLI commands that were in 1.2 but never added to 1.4 ........ 2007-08-31 15:54 +0000 [r81402-81407] Joshua Colp * /, res/res_speech.c: Merged revisions 81406 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81406 | file | 2007-08-31 12:53:16 -0300 (Fri, 31 Aug 2007) | 2 lines Make it the engine's responsible to check for the presence of results. ........ * /, res/res_features.c: Merged revisions 81403 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81403 | file | 2007-08-31 11:38:59 -0300 (Fri, 31 Aug 2007) | 4 lines (closes issue #10618) Reported by: dimas Don't pass through the stopped sounds frame.... just drop it. ........ * /, res/res_features.c: Merged revisions 81401 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81401 | file | 2007-08-30 20:53:41 -0300 (Thu, 30 Aug 2007) | 4 lines (closes issue #10009) Reported by: dimas Don't output a bridge failed warning message if it failed because one of the channels was part of the masquerade process. That is perfectly normal. ........ 2007-08-30 23:52 +0000 [r81400] Tilghman Lesher * channels/chan_zap.c: Add new queryable fields from zaptel to 'zap show status' 2007-08-30 22:08 +0000 [r81398] Mark Michelson * /, apps/app_queue.c: Merged revisions 81397 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81397 | mmichelson | 2007-08-30 17:05:56 -0500 (Thu, 30 Aug 2007) | 7 lines Removing an extraneous (and possibly misleading) log message. Firstly, if the announce file isn't found, the streaming functions will report it. Secondly, not all non-zero returns from play_file mean that the announce file wasn't found. Positive return values simply mean that a digit was pressed (most likely to skip through the announcement). (closes issue #10612, reported and patched by dimas) ........ 2007-08-30 21:25 +0000 [r81394-81396] Joshua Colp * /, channels/chan_sip.c: Merged revisions 81395 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81395 | file | 2007-08-30 18:23:50 -0300 (Thu, 30 Aug 2007) | 6 lines (closes issue #10514) Reported by: casper Patches: chan_sip.c.80129.diff uploaded by casper (license 55) Remove needless check for AUTH_UNKNOWN_DOMAIN. It was impossible for it to ever be that value. ........ * channels/chan_sip.c: (closes issue #10565) Reported by: tootai Make sure the external IP address has the standard SIP port set for when the user does not specify the port in the externip setting. 2007-08-30 21:16 +0000 [r81393] Steve Murphy * main/cdr.c, /: Merged revisions 81392 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81392 | murf | 2007-08-30 15:11:48 -0600 (Thu, 30 Aug 2007) | 1 line via issue 10599, where 'CDR already initialized' messages are being generated. Since all channels will have an init'd CDR attached at creation time, this message is now particularly useless. Removed. ........ 2007-08-30 20:55 +0000 [r81391] Joshua Colp * apps/app_minivm.c: (closes issue #10336) Reported by: junky Patches: minivm_output2.diff uploaded by junky (license 177) Change console output of minivm show stats to be more simple for external parsing. 2007-08-30 20:31 +0000 [r81389-81390] Tilghman Lesher * main/sched.c: A schedule id of 0 is not possible and is used to flag that we want to add a new item * apps/app_readexten.c: Change wording as requested by Kevin 2007-08-30 18:52 +0000 [r81388] Mark Michelson * configs/queues.conf.sample: Added note to sample queues.conf file to line up with most recent change regarding setinterfacevar. MEMBERREALTIME indicates whether a member is realtime. 2007-08-30 17:51 +0000 [r81387] Tilghman Lesher * main/logger.c: Always force reread of the config when we're rotating the log file (closes issue #10598) 2007-08-30 15:40 +0000 [r81384] Russell Bryant * /, channels/h323/ast_h323.cxx: Merged revisions 81383 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81383 | russell | 2007-08-30 10:38:29 -0500 (Thu, 30 Aug 2007) | 3 lines Add missing checks for the PTRACING define. (closes issue #10559, paravoid) ........ 2007-08-30 15:36 +0000 [r81382] Mark Michelson * /, apps/app_queue.c: Merged revisions 81381 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81381 | mmichelson | 2007-08-30 10:35:51 -0500 (Thu, 30 Aug 2007) | 3 lines Changed some manager event messages to reflect whether a queue member is a realtime member or not ........ 2007-08-30 15:34 +0000 [r81380] Russell Bryant * configs/modem.conf.sample (removed), /, configs/enum.conf.sample, configs/extensions.ael.sample: Merged revisions 81379 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81379 | russell | 2007-08-30 10:33:48 -0500 (Thu, 30 Aug 2007) | 3 lines Fix a typo, update a reload command, and remove an unused configuration file. (closes issue #10606, casper) ........ 2007-08-30 15:24 +0000 [r81378] Tilghman Lesher * apps/app_readexten.c (added): Add ReadExten app and VALID_EXTEN function (closes issue #10082) 2007-08-30 14:54 +0000 [r81376] Christian Richter * channels/chan_misdn.c, /: Merged revisions 81373 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81373 | crichter | 2007-08-30 16:43:33 +0200 (Do, 30 Aug 2007) | 1 line Fixed some warnings. ........ 2007-08-30 14:42 +0000 [r81370-81372] Joshua Colp * main/pbx.c, CHANGES: (closes issue #10603) Reported by: jmls Patches: pbx.diff uploaded by jmls (license 141) Add REASON dialplan variable for when an originated call fails and the failed extension is executed. * /, res/res_features.c: Merged revisions 81369 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81369 | file | 2007-08-30 11:23:40 -0300 (Thu, 30 Aug 2007) | 4 lines (issue #10599) Reported by: dimas Handle the -1 control subclass during feature dialing (it indicates to stop sounds). ........ 2007-08-30 08:50 +0000 [r81368] Christian Richter * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged revisions 81367 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81367 | crichter | 2007-08-30 10:31:59 +0200 (Do, 30 Aug 2007) | 11 lines Fixed a severe issue where a misdn_read would lock the channel, but read would not return because it blocks. later chan_misdn would try to queue a frame like a AST_CONTROL_ANSWER which could result in a deadlock situation. misdn_read will now not block forever anymore, and we don't queue the ANSWER frame at all when we already was called with misdn_answer -> answer would be called twice. Also we don't explicitly send a RELEASE_COMPLETE on receiption of a RELEASE anymore, because mISDN does that for us, this resulted in a problem on some switches, which would block our port after some calls for a short while. ........ 2007-08-29 22:05 +0000 [r81365] Mark Michelson * apps/app_queue.c: Added the MEMBERREALTIME variable when using setinterfacevar in queues.conf 2007-08-29 21:55 +0000 [r81364] Joshua Colp * include/asterisk/event.h: Make the event header file work under C++. 2007-08-29 21:30 +0000 [r81363] Steve Murphy * main/config.c: init newer so compile won't complain. 2007-08-29 21:25 +0000 [r81362] Russell Bryant * main/config.c: make trunk build again. murf will have to review this to see if it was the right fix, as it is related to his last change. 2007-08-29 20:55 +0000 [r81361] Steve Murphy * res/res_config_pgsql.c, channels/chan_sip.c, include/asterisk/config.h, channels/chan_iax2.c, channels/iax2-parser.c, res/res_config_sqlite.c, main/config.c, main/channel.c, res/res_config_odbc.c, pbx/pbx_spool.c, main/manager.c, channels/chan_skinny.c, apps/app_minivm.c, main/http.c, utils/extconf.c, apps/app_directory.c, apps/app_parkandannounce.c, apps/app_voicemail.c: This code was in team/murf/bug8684-trunk; it should fix bug 8684 in trunk. I didn't add it to 1.4 yet, because it's not entirely clear to me if this is a bug fix or an enhancement. A lot of files were affected by small changes like ast_variable_new getting an added arg, for the file name the var was defined in; ast_category_new gets added args of filename and lineno; ast_category and ast_variable structures now record file and lineno for each entry; a list of all #include and #execs in a config file (or any of its inclusions are now kept in the ast_config struct; at save time, each entry is put back into its proper file of origin, in order. #include and #exec directives are folded in properly. Headers indicating that the file was generated, are generated also for each included file. Some changes to main/manager.c to take care of file renaming, via the UpdateConfig command. Multiple inclusions of the same file are handled by exploding these into multiple include files, uniquely named. There's probably more, but I can't remember it right now. 2007-08-29 19:41 +0000 [r81353-81356] Russell Bryant * main/event.c: Try to clarify the rules on changing ast_event and ast_event_ie * main/event.c: Fix parenthesis from my last commit * main/event.c: Change pointer aritmetic on void * to char * * main/event.c: there is not actually code that sends these over the network in trunk yet 2007-08-29 16:39 +0000 [r81350] Mark Michelson * /, apps/app_queue.c: Merged revisions 81349 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81349 | mmichelson | 2007-08-29 11:35:29 -0500 (Wed, 29 Aug 2007) | 12 lines This patch, in essence, will correctly pause a realtime queue member and reflect those changes in the realtime engine. (issue #10424, reported by irroot, patch by me) This patch creates a new function called update_realtime_member_field, which is a generic function which will allow any one field of a realtime queue member to be updated. This patch only uses this function to update the paused status of a queue member, but it lays the foundation for persisting the state of a realtime member the same way that static members' state is maintained when using the persistentmembers setting ........ 2007-08-29 16:25 +0000 [r81348] Joshua Colp * main/event.c: Return ast_event_get_ie_raw to using an iterator and fix logic in ast_event_iterator_next. 2007-08-29 16:09 +0000 [r81347] Mark Michelson * /, apps/app_queue.c: Merged revisions 81346 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81346 | mmichelson | 2007-08-29 11:08:09 -0500 (Wed, 29 Aug 2007) | 3 lines Changed some tabs to spaces ........ 2007-08-29 16:07 +0000 [r81344-81345] Joshua Colp * main/event.c: This concludes bringing trunk back to a working state. * include/asterisk/event.h, main/event.c: To keep others happy... revert part of my additions so trunk works. 2007-08-29 15:59 +0000 [r81343] Russell Bryant * /, main/Makefile: Merged revisions 81342 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81342 | russell | 2007-08-29 10:57:29 -0500 (Wed, 29 Aug 2007) | 3 lines If chan_h323 is not being built, don't use g++ to do the final link of Asterisk. (in response to a question on the asterisk-dev list) ........ 2007-08-29 15:57 +0000 [r81341] Mark Michelson * /, apps/app_queue.c: Merged revisions 81340 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81340 | mmichelson | 2007-08-29 10:52:42 -0500 (Wed, 29 Aug 2007) | 8 lines This fix creates a more accurate way of detecting whether realtime members were deleted. (closes issue 10541, reported by Alric, patched by me) The REALLY nice things about this patch is that queue members now have a "realtime" field which will be true if the member is a realtime member. This means we can check this value prior to certain processing if it should ONLY be done for realtime members. ........ 2007-08-29 15:21 +0000 [r81335] Tilghman Lesher * channels/chan_iax2.c: Changed one too many variable settings in issue #9315 (closes issue #10592) 2007-08-29 15:19 +0000 [r81334] Joshua Colp * include/asterisk/event.h, include/asterisk/event_defs.h, main/event.c: Add API calls for iterating through an event. This should allow events to have multiple information elements (while there was nothing preventing it before you could not actually access any except the first one). 2007-08-29 14:19 +0000 [r81333] Mark Michelson * apps/app_meetme.c: Changing a NOTICE to a DEBUG. (closes issue #10591, reported and patched by junky, with small modification by me) 2007-08-29 14:16 +0000 [r81326-81332] Joshua Colp * /, channels/chan_sip.c: Merged revisions 81331 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81331 | file | 2007-08-29 11:13:55 -0300 (Wed, 29 Aug 2007) | 4 lines (closes issue #9690) Reported by: mattv Make rtp timeouts work even if two RTP streams are directly bridged in the RTP stack. ........ * include/asterisk/utils.h: Add inline function for signed linear subtraction. 2007-08-28 21:39 +0000 [r81292] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 81291 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81291 | russell | 2007-08-28 16:38:26 -0500 (Tue, 28 Aug 2007) | 3 lines Change the message about receiving a mini-frame before the first full voice frame to a DEBUG message. ........ 2007-08-28 21:35 +0000 [r81290] Joshua Colp * main/logger.c: Add some read/write locking magic to make logger reload operate again. 2007-08-28 20:03 +0000 [r81277] Tilghman Lesher * main/logger.c, UPGRADE.txt, configs/logger.conf.sample: Support better rotation of log files to be more like system logging (closes issue #10398) 2007-08-28 19:12 +0000 [r81227-81264] Russell Bryant * include/asterisk/audiohook.h: Change the audiohook lock and unlock wrappers to macros instead of inline functions. As inline functions, the lock debug information will show that these are always locked in audiohooks.h instead of the file where the lock was actually acquired. * funcs/func_enum.c, pbx/pbx_dundi.c: Add proper channel locking around the uses of datastore_add and _find. There are still more places in the tree that I have not yet changed if someone wants to go through and find the places they are used without the channel locked. * main/channel.c, funcs/func_volume.c, include/asterisk/channel.h: * Constify the uid field of channel datastores * Convert some spaces to tabs in func_volume * Add a note in channel.h making it clear that none of the datastore API calls lock the channel they are given, so the channel should be locked before calling the functions that take a channel argument. * include/asterisk/app.h, main/app.c, CHANGES, main/asterisk.c, doc/tex/asterisk-conf.tex: (closes issue #7852) Reported by: nic_bellamy Patches: 2006-10-03_svn_44249_voicemail_lockmode_v3.patch uploaded by nic_bellamy (license 213) Add support for configurable file locking methods. The default is "lockfile", which is the old behavior. There is an additional option, "flock", which is intended for use in situations where the lockfile method will not work, such as with SMB/CIFS mounts. * /, configs/indications.conf.sample: Merged revisions 81226 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81226 | russell | 2007-08-28 10:41:15 -0500 (Tue, 28 Aug 2007) | 2 lines Add Russian tones. (closes issue #7953, hanabana) ........ 2007-08-28 14:37 +0000 [r81210] Joshua Colp * res/res_features.c: (closes issue #10579) Reported by: ornati Make sure the called channel during the attended transfer process becomes associated with the calling channel so that the ast_waitfor_* call works properly under epoll. 2007-08-28 14:12 +0000 [r81121-81190] Mark Michelson * /, contrib/scripts/vmail.cgi: Merged revisions 81189 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81189 | mmichelson | 2007-08-28 09:12:14 -0500 (Tue, 28 Aug 2007) | 5 lines Fixes a forwarding problem when using res_config_mysql (closes issue #10573, reported by chrisvaughan, patch suggested by chrisvaughan as well) ........ * /, apps/app_queue.c: Merged revisions 81158 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81158 | mmichelson | 2007-08-27 17:40:19 -0500 (Mon, 27 Aug 2007) | 5 lines Resolve a potential deadlock. In this case, a single queue is locked, then the queue list. In changethread(), the queue list is locked, and then each individual queue is locked. Under the right circumstances, this could deadlock. As such, I have unlocked the individual queue before locking the queue list, and then locked the queue back after the queue list is unlocked. ........ * /, channels/chan_agent.c: Merged revisions 81120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81120 | mmichelson | 2007-08-27 16:08:48 -0500 (Mon, 27 Aug 2007) | 7 lines DTMF begin frames should be ignored so that when an agent acks a call with the '#' key, he doesn't cause a queue's announce file to be interrupted. Also went ahead and did the same for the '*' key and for ending a call. (closes issue #10528, reported by deskhack, patched by me) ........ 2007-08-27 20:55 +0000 [r81118] Tilghman Lesher * apps/app_directed_pickup.c: Enhance Pickup to do native pickupgroup pickup when no arguments are specified (closes issue #10404) 2007-08-27 17:44 +0000 [r81043-81098] Russell Bryant * /, pbx/pbx_dundi.c: This should have been trunk only, I guess. oh well ... it's harmless. Merged revisions 81065 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81065 | russell | 2007-08-27 11:38:33 -0500 (Mon, 27 Aug 2007) | 1 line explicity define a variable as a boolean ........ * /, pbx/pbx_dundi.c: Merged revisions 81074 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81074 | russell | 2007-08-27 12:27:48 -0500 (Mon, 27 Aug 2007) | 3 lines Add a \todo to note that this module leaks most of the memory it allocates on unload and should be fixed (when I'm not in the middle of something else ...). ........ * /, res/res_musiconhold.c: Merged revisions 81042 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81042 | russell | 2007-08-27 11:16:25 -0500 (Mon, 27 Aug 2007) | 11 lines (closes issue #10419) Reported by: mustardman Patches: asterisk-mohposition.diff.txt uploaded by jamesgolovich (license 176) This patch fixes a few problems with music on hold. * Fix issues with starting at the beginning of a file when it shouldn't. * Fix the inuse counter to be decremented even if the class had not been set to be deleted when not in use anymore * Don't arbitrarily limit the number of MOH files to 255 ........ 2007-08-27 15:03 +0000 [r81013] Joshua Colp * /, channels/chan_sip.c: Merged revisions 81012 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81012 | file | 2007-08-27 12:01:59 -0300 (Mon, 27 Aug 2007) | 6 lines (closes issue #10561) Reported by: jesselang Patches: chan_sip-ChannelReload-20080825.patch uploaded by jesselang (license 202) Remove an extra \r\n to make the ChannelReload event conform with every other event. ........ 2007-08-27 14:56 +0000 [r81011] Mark Michelson * /, apps/app_queue.c: Merged revisions 81010 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81010 | mmichelson | 2007-08-27 09:55:44 -0500 (Mon, 27 Aug 2007) | 3 lines Found a case where the queue's membercount is off. It does not take into account dynamic members on a reload. ........ 2007-08-27 13:35 +0000 [r80962-80991] Joshua Colp * channels/chan_sip.c: Remove places that say if no language is specified it will default to english... since on some setups this is untrue. * /, main/rtp.c: Merged revisions 80974 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80974 | file | 2007-08-27 10:20:31 -0300 (Mon, 27 Aug 2007) | 4 lines (closes issue #10562) Reported by: idkpmiller Correct jitter value output in the CLI to be as expected. ........ * configs/sip.conf.sample: (closes issue #10569) Reported by: IgorG Patches: sip_conf-80933-1.patch uploaded by IgorG (license 20) Fix up sip.conf sample configuration. 2007-08-26 18:12 +0000 [r80933] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 80932 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80932 | russell | 2007-08-26 13:11:26 -0500 (Sun, 26 Aug 2007) | 3 lines Remove an extra signal_condition() for the scheduler thread. (closes issue #10564, patch from casper) ........ 2007-08-25 17:55 +0000 [r80821-80898] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 80895 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80895 | russell | 2007-08-25 12:37:39 -0500 (Sat, 25 Aug 2007) | 7 lines Fix some issues with the handling of the scheduler in chan_iax2. Most of the places that scheduled items to be executed by the scheduler thread did not signal the scheduler thread to wake up so that it could recalculate the time until the next action. These changes will make the scheduler thread more responsive and ensure that actions get executed as close to when intended as possible instead of it being possible for very long delays. ........ * pbx/pbx_dundi.c: localize a variable and remove a duplicate error message * apps/app_queue.c: use ast_strlen_zero * /, channels/chan_iax2.c: Merged revisions 80849 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80849 | russell | 2007-08-24 16:22:50 -0500 (Fri, 24 Aug 2007) | 5 lines If dnsmgr is in use, and no DNS servers are available when Asterisk first starts, then don't give up on poking peers. Allow the poke to get rescheduled so that it will work once the dnsmgr is able to resolve the host. (closes issue #10521, patch by jamesgolovich) ........ * /, main/dsp.c: Merged revisions 80820 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80820 | russell | 2007-08-24 15:24:05 -0500 (Fri, 24 Aug 2007) | 7 lines Improve the debouncing logic in the DTMF detector to fix some reliability issues. Previously, this code used a shift register of hits and non-hits. However, if the start of the digit isn't clean, it is possible for the leading edge detector to miss the digit. These changes replace the flawed shift register logic and also does the debouncing on the trailing edge as well. (closes issue #10535, many thanks to softins for the patch) ........ 2007-08-24 20:21 +0000 [r80819] BJ Weschke * apps/app_queue.c: Merged revisions 80818 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80818 | bweschke | 2007-08-24 15:52:06 -0400 (Fri, 24 Aug 2007) | 3 lines A minor correction to the available logic of autofill. If a queue member is paused, they're not really "available" so don't count them as such. Somewhat related to issue #10155 ........ 2007-08-24 19:50 +0000 [r80817] Tilghman Lesher * main/pbx.c: Fix documentation for Set (closes issue #10549) 2007-08-24 19:03 +0000 [r80790] Steve Murphy * main/cdr.c, /: Merged revisions 80789 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80789 | murf | 2007-08-24 12:52:15 -0600 (Fri, 24 Aug 2007) | 1 line From a complaint by jmls, I realize that the message in cdr_disposition is unnecessary. To get failure disposition, just return -1; no use having more than one case do that. ........ 2007-08-24 18:05 +0000 [r80778] Matthew Fredrickson * channels/chan_zap.c: Add VMWI chan_zap support #9909 2007-08-24 15:53 +0000 [r80751] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 80750 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80750 | mmichelson | 2007-08-24 10:51:03 -0500 (Fri, 24 Aug 2007) | 3 lines Fix a possible crash in IMAP voicemail. ........ 2007-08-24 15:42 +0000 [r80748] Steve Murphy * utils/conf2ael.c: fix up the MODULEINFO in conf2ael.c as well 2007-08-24 15:29 +0000 [r80725] Russell Bryant * /, utils/ael_main.c: Merged revisions 80722 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80722 | russell | 2007-08-24 10:28:05 -0500 (Fri, 24 Aug 2007) | 3 lines Tweak the formatting of this MODULEINFO block. I think this would have caused a "*" to get in the menuselect-tree file. ........ 2007-08-24 14:55 +0000 [r80690-80718] Steve Murphy * /, utils/ael_main.c, utils/conf2ael.c: Merged revisions 80717 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80717 | murf | 2007-08-24 08:48:49 -0600 (Fri, 24 Aug 2007) | 1 line This change addresses JerJer's complaint that aelparse builds and installs even if pbx_ael is unchecked in the menuselect stuff. ........ 2007-08-24 11:49 +0000 [r80662] Philippe Sultan * /, channels/chan_gtalk.c: Merged revisions 80661 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80661 | phsultan | 2007-08-24 13:42:46 +0200 (Fri, 24 Aug 2007) | 9 lines Closes issue #10509 Googletalk calls are answered too early, which results in CDRs wrongly stating that a call was ANSWERED when the calling party cancelled a call before before being established. We must not answer the call upon reception of a 'transport-accept' iq packet, but this packet still needs to be acknowledged, otherwise the remote peer would close the call (like in #8970). ........ 2007-08-23 23:37 +0000 [r80649] Steve Murphy * pbx/ael/ael-test/ref.ael-ntest10, res/ael/ael.tab.c, res/ael/ael.y, res/ael/ael.tab.h, pbx/ael/ael-test/ref.ael-test6, pbx/ael/ael-test/ref.ael-test7: an unreported crash I debugged, looked like it was backing up way too far after hitting the syntax error. An inspection of the code revealed that error tokens in lists were not rearranged when the rules were rearranged as part of a code neatening-up process. By moving the error tokens to where they should be, I also reduced the number of shift/reduce conflicts to 3 instead of 8. This introduces subtle differences in error messages, so the regressions had to be updated. 2007-08-23 21:34 +0000 [r80510-80616] Russell Bryant * apps/app_while.c: Use the comma separator in app_while. reported by blitzrage on irc, patched by me * /, res/res_features.c, include/asterisk/features.h: Merged revisions 80573 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80573 | russell | 2007-08-23 15:16:41 -0500 (Thu, 23 Aug 2007) | 5 lines When executing a dynamic feature, don't look it up a second time by digit pattern after we already looked it up by name. This causes broken behavior if there is more than one feature defined with the same digit pattern. (closes issue #10539, reported by bungalow, patch by me) ........ * /, funcs/func_timeout.c: Merged revisions 80547 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80547 | russell | 2007-08-23 14:29:44 -0500 (Thu, 23 Aug 2007) | 3 lines Revert very broken fix for issue #10540 ... none of these values take ms so I don't know what I was thinking ........ * /, funcs/func_timeout.c: Merged revisions 80539 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80539 | russell | 2007-08-23 14:21:53 -0500 (Thu, 23 Aug 2007) | 4 lines Fix func_timeout to take values in floating point so 1.5 actually means 1.5 seconds instead of being rounded. (closes issue #10540, reported by spendergrass, patch by me) ........ * doc/asterisk-mib.txt, res/snmp/agent.c: Fix a typo in the Asterisk MIB and fix astNumChanBridged so it acts as a counter again (closes issue #10118, patch by jeffg) 2007-08-23 17:18 +0000 [r80508] Kevin P. Fleming * channels/chan_zap.c, /: Merged revisions 80501 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80501 | kpfleming | 2007-08-23 12:08:25 -0500 (Thu, 23 Aug 2007) | 2 lines report the actual channel number that was unregistered, instead of assuming that the interface list consists of channels 1 through with no gaps in the sequence ........ 2007-08-23 17:04 +0000 [r80470-80500] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 80499 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80499 | russell | 2007-08-23 12:02:50 -0500 (Thu, 23 Aug 2007) | 3 lines Fix some code where it was possible for a reference to a peer to not get released when it should. Thank you to Marta Carbone for pointing this out! ........ * /, res/res_agi.c: Merged revisions 80469 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80469 | russell | 2007-08-23 10:49:28 -0500 (Thu, 23 Aug 2007) | 2 lines Revert res_agi fix that didn't quite work until we get it right ... ........ 2007-08-23 15:48 +0000 [r80453-80468] Joshua Colp * channels/chan_sip.c: If no default language has been specified print out that it will default to english when using sip show peer or sip show user. * main/minimime/mm.h: Return trunk to a working state by including compat.h in minimime. 2007-08-22 23:26 +0000 [r80428-80429] Jason Parker * main/minimime/mm_util.c, main/minimime/mm_codecs.c, main/minimime/mm_mem.h, main/minimime/mm_base64.c, main/minimime/mm.h: Convert minimime to use the proper uint*_t types, rather than u_int*_t * apps/app_minivm.c: Cast calls to getpid. This was done in 1.4 already, this one was just new 2007-08-22 22:54 +0000 [r80361-80427] Russell Bryant * /, include/asterisk/astobj2.h: Merged revisions 80426 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80426 | russell | 2007-08-22 17:54:03 -0500 (Wed, 22 Aug 2007) | 6 lines Add some more documentation on iterating ao2 containers. The documentation implies that is possible to miss an object or see an object twice while iterating. After looking through the code and talking with mmichelson, I have documented the exact conditions under which this can happen (which are rare and harmless in most cases). ........ * /, main/astobj2.c: Merged revisions 80424 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80424 | russell | 2007-08-22 17:40:27 -0500 (Wed, 22 Aug 2007) | 10 lines When converting this code to use the list macros, I changed it so objects are added to the head of a bucket instead of the tail. However, while looking over code with mmichelson, we noticed that the algorithm used in ao2_iterator_next requires that items are added to the tail. This wouldn't have caused any huge problem, but it wasn't correct. It meant that if an object was added to a container while you were iterating it, and it was added to the same bucket that the current element is in, then the new object would be returned by ao2_iterator_next, and any other objects in the bucket would be bypassed in the traversal. ........ * channels/chan_iax2.c: allow peers and users to go into a hash table * /, channels/chan_sip.c: Merged revisions 80390 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80390 | russell | 2007-08-22 16:00:44 -0500 (Wed, 22 Aug 2007) | 3 lines Don't crash when using realtime in chan_sip without an insecure setting in the database. (closes issue #10348, reported by link55, fixed by me) ........ * channels/chan_iax2.c: Unsubscribe from MWI events in the peer destructor * /, main/Makefile, include/asterisk/astobj2.h (added), include/asterisk/strings.h, channels/chan_iax2.c, main/astobj2.c (added): Merged revisions 80362 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80362 | russell | 2007-08-22 15:21:36 -0500 (Wed, 22 Aug 2007) | 34 lines Merge changes from team/russell/iax_refcount. This set of changes fixes problems with the handling of iax2_user and iax2_peer objects. It was very possible for a thread to still hold a reference to one of these objects while a reload operation tries to delete them. The fix here is to ensure that all references to these objects are tracked so that they can't go away while still in use. To accomplish this, I used the astobj2 reference counted object model. This code has been in one of Luigi Rizzo's branches for a long time and was primarily developed by one of his students, Marta Carbone. I wanted to go ahead and bring this in to 1.4 because there are other problems similar to the ones fixed by these changes, so we might as well go ahead and use the new astobj if we're going to go through all of the work necessary to fix the problems. As a nice side benefit of these changes, peer and user handling got more efficient. Using astobj2 lets us not hold the container lock for peers or users nearly as long while iterating. Also, by changing a define at the top of chan_iax2.c, the objects will be distributed in a hash table, drastically increasing lookup speed in these containers, which will have a very big impact on systems that have a large number of users or peers. The use of the hash table will be made the default in trunk. It is not the default in 1.4 because it changes the behavior slightly. Previously, since peers and users were stored in memory in the same order they were specified in the configuration file, you could influence peer and user matching order based on the order they are specified in the configuration. The hash table does not guarantee any order in the container, so this behavior will be going away. It just means that you have to be a little more careful ensuring that peers and users are matched explicitly and not forcing chan_iax2 to have to guess which user is the right one based on secret, host, and access list settings, instead of simply using the username. If you have any questions, feel free to ask on the asterisk-dev list. ........ * /, res/res_agi.c: Merged revisions 80360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80360 | russell | 2007-08-22 14:53:30 -0500 (Wed, 22 Aug 2007) | 5 lines Juggie in #asterisk-dev was reporting problems where fgets would return without reading the whole line when using fastagi. When this happens, errno was set to EINTR or EAGAIN. This patch accounts for the possibility and lets fgets continue in that case. ........ 2007-08-22 18:54 +0000 [r80303-80331] Jason Parker * Makefile, build_tools/mkpkgconfig, /, build_tools/make_build_h, build_tools/strip_nonapi, build_tools/prep_moduledeps, build_tools/make_buildopts_h: Merged revisions 80330 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80330 | qwell | 2007-08-22 13:53:18 -0500 (Wed, 22 Aug 2007) | 7 lines Fix a few build issues in Solaris (and likely others). Use GREP and ID variables from autoconf. Reported to me in #asterisk-dev I forgot who reported this - sorry. :( ........ * Makefile, /: Merged revisions 80304 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80304 | qwell | 2007-08-22 13:25:34 -0500 (Wed, 22 Aug 2007) | 2 lines Change a syntax that the GNU make in Solaris dislikes. ........ * /, build_tools/make_version: Merged revisions 80302 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80302 | qwell | 2007-08-22 13:06:00 -0500 (Wed, 22 Aug 2007) | 3 lines Fix a bashism (we explicitly request /bin/sh). Remove some oddly placed quotes I found in passing. ........ 2007-08-22 16:27 +0000 [r80258-80262] Russell Bryant * utils/check_expr.c: Ensure that the object code for ast_atomic_fetchadd_int() gets included in the check_expr binary when building with LOW_MEMORY defined. (reported by Brian Capouch on the asterisk-dev list, patch by me) * Makefile, /: Merged revisions 80257 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80257 | russell | 2007-08-22 11:21:58 -0500 (Wed, 22 Aug 2007) | 4 lines Honor the contents of the COPTS variable as custom target CFLAGS. Apparently this is what openwrt does. (reported by Brian Capouch on the asterisk-dev list, patch by me) ........ 2007-08-22 16:16 +0000 [r80256] Joshua Colp * /, main/rtp.c: Merged revisions 80255 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80255 | file | 2007-08-22 13:14:38 -0300 (Wed, 22 Aug 2007) | 4 lines (closes issue #10526) Reported by: sinistermidget Revert commit from issue #10355 and return timestamp skew to 640. ........ 2007-08-22 14:17 +0000 [r80241-80242] Steve Murphy * /: blocking 80167 * /, main/alaw.c: Merged revisions 80166 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80166 | murf | 2007-08-21 10:36:34 -0600 (Tue, 21 Aug 2007) | 1 line This patch solves problem 1 in 8126; it should not slow down the alaw codec, but should prevent signal degradation via multiple trips thru the codec. Fossil estimates the twice thru this codec will prevent fax from working. 4-6 times thru would result hearable, noticeable, voice degradation. ........ 2007-08-21 21:58 +0000 [r80226] Russell Bryant * funcs/func_odbc.c: use ast_atomic_fetchadd_int for incrementing resultcount 2007-08-21 20:55 +0000 [r80217] Steve Murphy * res/ael/pval.c: As per 10472, mvanbaak thought the generated code would look better this way. 2007-08-21 18:49 +0000 [r80184] Russell Bryant * /, channels/chan_sip.c: Merged revisions 80183 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80183 | russell | 2007-08-21 13:42:15 -0500 (Tue, 21 Aug 2007) | 7 lines Don't record SIP dialog history if it's not turned on. Also, put an upper limit on how many history entires will be stored for each SIP dialog. It is currently set to 50, but can be increased if deemed necessary. (closes issue #10421, closes issue #10418, patches suggested by jmoldenhauer, patches updated by me) (Security implications documented in AST-2007-020) ........ 2007-08-21 15:51 +0000 [r80157] Joshua Colp * main/audiohook.c: Minor tweak. Don't manipulate volume of the audio in the buffer if no audio is actually there. 2007-08-21 15:23 +0000 [r80133] Russell Bryant * /, channels/chan_mgcp.c: Merged revisions 80132 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80132 | russell | 2007-08-21 10:22:22 -0500 (Tue, 21 Aug 2007) | 3 lines Don't try to dereference the owner channel when it may not exist (issue #10507, maxper) ........ 2007-08-21 15:04 +0000 [r80131] Jason Parker * /, configs/cdr.conf.sample: Merged revisions 80130 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80130 | qwell | 2007-08-21 10:03:45 -0500 (Tue, 21 Aug 2007) | 7 lines (closes issue #10510) Reported by: casper Patches: cdr.conf.diff uploaded by casper (license 55) Fix a few errors in sample cdr config file. ........ 2007-08-20 22:53 +0000 [r80113] Steve Murphy * build_tools/cflags.xml, main/ulaw.c, codecs/slin_ulaw_ex.h, codecs/ulaw_slin_ex.h, include/asterisk/alaw.h, main/translate.c, include/asterisk/ulaw.h, main/alaw.c: This change set fixes bug 8126 in trunk. It is implemented via compile time options, activated via the menuselect stuff, which defaults to the old way. non-zero sample data added. Translate tables expressed in microseconds instead of milliseconds, with 5-digit data now instead of 3, giving 2 more digits of precision. 2007-08-20 17:37 +0000 [r80075] Steve Murphy * include/asterisk/lock.h, utils/extconf.c: Stephn Davies reports that this will help make things work on 64-bit machines 2007-08-20 16:18 +0000 [r80050] Mark Michelson * /, apps/app_queue.c: Merged revisions 80049 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80049 | mmichelson | 2007-08-20 11:17:43 -0500 (Mon, 20 Aug 2007) | 4 lines Found a pointless ternary if. member->dynamic was set to 1 and has no opportunity to change between then and this line, so "dynamic" will ALWAYS be output. ........ 2007-08-20 16:12 +0000 [r80048] Jason Parker * /, configs/extensions.conf.sample: Merged revisions 80047 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80047 | qwell | 2007-08-20 11:08:49 -0500 (Mon, 20 Aug 2007) | 7 lines (closes issue #10499) Reported by: casper Patches: extensions.conf.sample.diff uploaded by casper (license 55) Update CLI examples in extensions.conf.sample to reflect command changes. ........ 2007-08-20 15:53 +0000 [r80046] Joshua Colp * apps/app_voicemail.c: Remove remnants of last commit so trunk builds again. 2007-08-20 15:37 +0000 [r80045] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 80044 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r80044 | mmichelson | 2007-08-20 10:34:43 -0500 (Mon, 20 Aug 2007) | 5 lines Ukrainian language voicemail support. (closes issue #10458, reported and patched by Oleh) ........ 2007-08-20 15:27 +0000 [r80037] Steve Murphy * utils/pval.c (removed): pval.c should not be in svn, in the utils dir 2007-08-20 15:10 +0000 [r80023-80033] Joshua Colp * utils/pval.c: Bring pval.c in utils up to date with pval.c in res/ael. * channels/chan_zap.c: Fix random segfault issue when loading chan_zap. Trying to access a configuration structure that has already been destroyed is bad, mmmk? 2007-08-20 02:46 +0000 [r79999] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 79998 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79998 | tilghman | 2007-08-19 21:42:49 -0500 (Sun, 19 Aug 2007) | 2 lines Missing curly braces. Oops. (Reported by snuffy via IRC) ........ 2007-08-20 00:54 +0000 [r79988-79990] Joshua Colp * channels/chan_iax2.c: (closes issue #10495) Reported by: stevedavies Make sure context pointer is valid or else chan_iax2 will go kaboom. * utils/Makefile: (closes issue #10496) Reported by: caio1982 Fix building on OSX. * channels/chan_h323.c: Fix building of trunk. I'm doing work on a Sunday night just to avoid watching Snakes on a Plane which my roommate is watching. 2007-08-19 14:17 +0000 [r79980] Tilghman Lesher * utils/Makefile: Add strcompat dependency for check_expr (needed for platforms that don't have strndup) 2007-08-18 23:58 +0000 [r79972] Joshua Colp * configure, configure.ac: Actually check the return value of epoll_create to make sure it works. 2007-08-18 14:34 +0000 [r79940-79949] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 79947 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79947 | tilghman | 2007-08-18 09:30:44 -0500 (Sat, 18 Aug 2007) | 3 lines Don't allocate vmu for messagecount when we could just use the stack instead (closes issue #10490) Also, remove a useless (and leaky) SQLAllocHandle (closes issue #10480) ........ * channels/chan_zap.c, channels/chan_sip.c, channels/chan_h323.c, channels/chan_iax2.c: We weren't properly encapsulating the mtime ignores of config files (closes issue #10488) 2007-08-17 21:19 +0000 [r79915] Mark Michelson * apps/app_voicemail.c: I broke the build. Now I'm fixing it. 2007-08-17 21:04 +0000 [r79913] Russell Bryant * channels/chan_zap.c, /: Merged revisions 79912 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79912 | russell | 2007-08-17 16:01:43 -0500 (Fri, 17 Aug 2007) | 4 lines Avoid a crash in the handling of DTMF based Caller ID. It is valid for ast_read to return NULL in the case that the channel has been hung up. (crash reported by anonymouz666 on IRC in #asterisk-dev) ........ 2007-08-17 19:16 +0000 [r79907] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 79906 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79906 | mmichelson | 2007-08-17 14:14:05 -0500 (Fri, 17 Aug 2007) | 6 lines Patch allows for more seamless transition from file storage voicemail to ODBC storage voicemail. If a retrieval of a greeting from the database fails, but the file is found on the file system, then we go ahead an insert the greeting into the database. The result of this is that people who switch from file storage to ODBC storage do not need to rerecord their voicemail greetings. ........ 2007-08-17 19:13 +0000 [r79903-79905] Jason Parker * /, channels/chan_sip.c, main/utils.c, include/asterisk/strings.h: Merged revisions 79904 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #10430) ........ r79904 | qwell | 2007-08-17 14:12:19 -0500 (Fri, 17 Aug 2007) | 11 lines Don't send a semicolon over the wire in sip notify messages. Caused by fix for issue 9938. I basically took the code that existed before 9938 was fixed, and copied it into a new function - ast_unescape_semicolon There should be very few places this will be needed (pbx_config does NOT need this (see issue 9938 for details)) Issue 10430, patch by me, with help/ideas from murf (thanks murf). ........ * channels/chan_local.c, /: Merged revisions 79902 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #10485) ........ r79902 | qwell | 2007-08-17 12:44:22 -0500 (Fri, 17 Aug 2007) | 4 lines Re-add the setting of callerid name and number. Issue 10485, reported by and fix explained by paradise. ........ 2007-08-17 16:39 +0000 [r79901] Tilghman Lesher * configs/logger.conf.sample: Documentation for %q in logger.conf, as suggested by jtodd (closes issue #10475) 2007-08-17 16:04 +0000 [r79888-79894] Jason Parker * res/res_features.c: Fix Dial arguments in res_features. Closes issue #10484, patch by lunn. * pbx/pbx_dundi.c: Correct the argument separator for a Dial statement in pbx_dundi. Closes issue #10483, patch by lunn 2007-08-17 14:41 +0000 [r79885] Tilghman Lesher * main/config.c: Change this flag... might not otherwise unlock in an OOM situation 2007-08-17 14:14 +0000 [r79861-79862] Russell Bryant * channels/chan_iax2.c: Make use of ast_sched_replace() in some places in chan_iax2 * channels/chan_sip.c, include/asterisk/sched.h, main/sched.c: This commit adds a scheduler API call, ast_sched_replace that can be used in place of a very common construct. I also used it in a number of places in chan_sip. if (id > -1) ast_sched_del(sched, id); id = ast_sched_add(sched, ...); changes to: ast_sched_replace(id, sched, ...); 2007-08-17 13:45 +0000 [r79859-79860] Tilghman Lesher * res/res_config_odbc.c, res/res_config_sqlite.c: store and destroy implementations for sqlite (closes issue #10446) and odbc (closes issue #10447) * res/res_config_pgsql.c, funcs/func_lock.c: store and destroy implementations for realtime pgsql (closes issue #10372) 2007-08-17 13:39 +0000 [r79858] Russell Bryant * /, channels/chan_sip.c: Merged revisions 79857 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79857 | russell | 2007-08-17 08:37:08 -0500 (Fri, 17 Aug 2007) | 5 lines Fix some crashes in chan_sip. This patch changes various places that add items to the scheduler to ensure that they don't overwrite the ID of a previously scheduled item. If there is one, it should be removed. (closes issue #10391, closes issue #10256, probably others, patch by me) ........ 2007-08-17 08:29 +0000 [r79841] Christian Richter * channels/chan_misdn.c, /: Merged revisions 79833 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79833 | crichter | 2007-08-17 10:22:36 +0200 (Fr, 17 Aug 2007) | 1 line sometimes we don't need to signal dtmf tones to asterisk, we just want them to go through as inband. Otherwise they might be generated by the other channel partner and then there is a double tone. ........ 2007-08-17 01:19 +0000 [r79824] Joshua Colp * channels/chan_zap.c: Fix building of chan_zap under development mode without libpri and libss7 installed. 2007-08-16 23:31 +0000 [r79813] Tilghman Lesher * funcs/func_lock.c: Revise dialplan locks to permit multiple locks per channel, but with deadlock avoidance 2007-08-16 22:33 +0000 [r79764-79794] Russell Bryant * /: Merged revisions 79792 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79792 | russell | 2007-08-16 17:32:33 -0500 (Thu, 16 Aug 2007) | 4 lines Fix a little race condition that could cause a crash if two channels had MOH stopped at the same time that were using a class that had been marked for deletion when its use count hits zero. ........ * /, res/res_musiconhold.c: Merged revisions 79778 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79778 | russell | 2007-08-16 17:24:25 -0500 (Thu, 16 Aug 2007) | 14 lines This patch fixes a bug where reloading the module with "module reload" did not delete classes from memory that were no longer in the config. This patch fixes that problem as well as another one. Previously, if you reloaded MOH using the "moh reload" CLI command, which behaved differently than "module reload ...", MOH had to be stopped on every channel and started again immediately. However, there was no way to tell what class was being used, so they would all fall back to the default class. (closes issue #10139) Reported by: blitzrage Patches: asterisk-10139-advanced.diff.txt uploaded by jamesgolovich (license 176) Tested by: jamesgolovich ........ * /, channels/chan_iax2.c: Merged revisions 79756 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79756 | russell | 2007-08-16 16:29:24 -0500 (Thu, 16 Aug 2007) | 11 lines Fix more deadlocks in chan_iax2 that were introduced by making frame handling and scheduling multi-threaded. Unfortunately, we have to do some expensive deadlock avoidance when queueing frames on to the ast_channel owner of the IAX2 pvt struct. This was already handled for regular frames, but ast_queue_hangup and ast_queue_control were still used directly. Making these changes introduced even more places where the IAX2 pvt struct can disappear in the context of a function holding its lock due to calling a function that has to unlock/lock it to avoid deadlocks. I went through and fixed all of these places to account for this possibility. (issue #10362, patch by me) ........ 2007-08-16 21:28 +0000 [r79755] Joshua Colp * /: Fix properties on trunk again. 2007-08-16 21:21 +0000 [r79749] Mark Michelson * /, channels/chan_agent.c: Merged revisions 79748 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79748 | mmichelson | 2007-08-16 16:16:40 -0500 (Thu, 16 Aug 2007) | 8 lines Fixes a problem where agents would get stuck busy due to their wrapuptime being longer than the queue's wrapuptime and ringinuse=no for the queue. (closes issue #10215, reported by Doug, repaired by me) Special thanks to fkasumovic for pointing out the source of the problem and to bweschke for helping to come up with a solution! ........ 2007-08-16 21:09 +0000 [r79747] Tilghman Lesher * main/udptl.c, cdr/cdr_sqlite3_custom.c, /, res/res_features.c, codecs/codec_adpcm.c, apps/app_alarmreceiver.c, cdr/cdr_adaptive_odbc.c, channels/chan_iax2.c, main/config.c, main/loader.c, res/res_smdi.c, channels/chan_skinny.c, main/http.c, apps/app_amd.c, channels/chan_alsa.c, cdr/cdr_odbc.c, cdr/cdr_manager.c, codecs/codec_g722.c, apps/app_privacy.c, codecs/codec_speex.c, channels/chan_agent.c, codecs/codec_g726.c, channels/iax2-provision.c, apps/app_playback.c, channels/iax2-provision.h, channels/chan_misdn.c, res/res_indications.c, pbx/pbx_config.c, main/asterisk.c, res/res_odbc.c, apps/app_voicemail.c, channels/chan_vpb.cc, res/res_snmp.c, apps/app_meetme.c, codecs/codec_gsm.c, res/res_musiconhold.c, channels/chan_gtalk.c, cdr/cdr_pgsql.c, apps/app_followme.c, res/res_jabber.c, cdr/cdr_radius.c, codecs/codec_zap.c, res/res_config_sqlite.c, main/enum.c, channels/misdn_config.c, cdr/cdr_csv.c, main/cdr.c, channels/chan_phone.c, res/res_config_odbc.c, main/manager.c, apps/app_osplookup.c, funcs/func_odbc.c, apps/app_minivm.c, main/logger.c, apps/app_directory.c, apps/app_rpt.c, cdr/cdr_custom.c, channels/chan_mgcp.c, codecs/codec_lpc10.c, res/res_config_pgsql.c, main/dnsmgr.c, channels/chan_zap.c, channels/chan_sip.c, apps/app_festival.c, codecs/codec_alaw.c, res/res_adsi.c, include/asterisk/config.h, apps/app_queue.c, channels/chan_oss.c, main/rtp.c, cdr/cdr_tds.c, channels/chan_jingle.c, channels/misdn/chan_misdn_config.h, channels/chan_h323.c, pbx/pbx_dundi.c, codecs/codec_ulaw.c: Don't reload a configuration file if nothing has changed. 2007-08-16 19:40 +0000 [r79736] Steve Murphy * utils/pval.c, utils/conf2ael.c: Many thanks to mvanbaak for his update to translate hints; I added the -d option for local testing purposes. This is from bug 10472 2007-08-16 18:23 +0000 [r79724-79725] Dwayne M. Hubbard * channels/chan_iax2.c: added counter for iax2 show registry CLI output, closes issue 10461, thanks junky * apps/app_voicemail.c: added counter for voicemail show users, issue 10462, thanks junky 2007-08-16 17:34 +0000 [r79714-79719] Steve Murphy * utils/conf2ael.c: mvanbaak asks: why did you include that twice? Answer: dunno. removed redundant include * utils/extconf.c, utils/conf2ael.c: svn did me dirty for some reason. Left 5 files out of the commit; Tilghman copied them in from the branch, but I had made changes to these. Here they are. 2007-08-16 15:59 +0000 [r79691] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 79690 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79690 | mmichelson | 2007-08-16 10:58:34 -0500 (Thu, 16 Aug 2007) | 5 lines base_encode is not trying to open a log file, so we should not call it a log file in the warning. (related to issue #10452, reported by bcnit) ........ 2007-08-16 15:29 +0000 [r79687-79688] Joshua Colp * pbx/pbx_dundi.c: (closes issue #10467) Reported by: lunn Patches: pbx_dundi.diff uploaded by lunn (license 179) Don't print a warning saying an ethernet interface was found when it indeed was. * utils/conf2ael.c: Make conf2ael build on 64-bit systems. 2007-08-16 09:45 +0000 [r79666] Philippe Sultan * /, res/res_jabber.c: Merged revisions 79665 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79665 | phsultan | 2007-08-16 11:37:10 +0200 (Thu, 16 Aug 2007) | 21 lines A fix for two critical problems detected while working with Daniel McKeehan in issue #10184. Upon priority change, the resource list is not NULL terminated when moving an item to the end of the list. This makes Asterisk endlessy loop whenever it needs to read the list. Jids with different resource and priority values, like in Gmail's and GoogleTalk's jabber clients put that problem in evidence. Upon reception of a 'from' attribute with an empty resource string, Asterisk crashes when trying to access the found->cap pointer if the resource list for the given buddy is not empty. This situation is perfectly valid and must be handled. The Gizmoproject's jabber client put that problem in evidence. Also added a few comments in the code as well as a handle for the capabilities from Gmail's jabber client, which are stored in a caps:c tag rather than the usual c tag. Closes issue #10184. ........ 2007-08-16 09:22 +0000 [r79660] Christian Richter * /, channels/misdn/ie.c: Merged revisions 79642 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79642 | crichter | 2007-08-16 10:21:21 +0200 (Do, 16 Aug 2007) | 1 line 0x80 + protocol is wrong for USERUSER when we want to send IA5 Chars. ........ 2007-08-16 06:52 +0000 [r79638] Olle Johansson * CHANGES: Doc change 2007-08-15 22:53 +0000 [r79634] Jason Parker * res/res_musiconhold.c: Modify the names of functions/variables in res_musiconhold to be useful. Closes issue #10464, patch by caio1982 2007-08-15 21:25 +0000 [r79623] Tilghman Lesher * include/asterisk/pval.h (added), utils/pval.c (added), include/asterisk/extconf.h (added), utils/extconf.c (added), utils/conf2ael.c (added): Missing from murf's last trunk commit, which was why trunk won't compile 2007-08-15 19:34 +0000 [r79611] Joshua Colp * /: Remove properties that appeared from Steve's last branch merge. Automerge has already run so everyone's branches based off of trunk are probably toast by now. 2007-08-15 19:21 +0000 [r79595] Steve Murphy * /, pbx/ael/ael.y (removed), pbx/ael/ael-test/ref.ael-test11, res/Makefile, pbx/ael/ael-test/ref.ael-test14, pbx/ael/ael-test/ref.ael-test15, pbx/ael/ael-test/ref.ael-test16, pbx/ael/ael-test/ref.ael-test19, include/asterisk/ast_expr.h, pbx/ael/ael_lex.c (removed), pbx/pbx_ael.c, pbx/ael/ael.flex (removed), res/ael (added), main/pbx.c, UPGRADE.txt, res/res_ael_share.c (added), pbx/Makefile, CHANGES, utils/Makefile, pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael.tab.c (removed), pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test4, include/asterisk/ael_structs.h, pbx/ael/ael.tab.h (removed), pbx/ael/ael-test/ref.ael-test5, utils/ael_main.c, include/asterisk/pbx.h, pbx/ael/ael-test/ref.ael-test6, pbx/ael/ael-test/ref.ael-test7, utils/check_expr.c: This commit closes bug 7605, and half-closes 7638. The AEL code has been redistributed/repartitioned to allow code re-use both inside and outside of Asterisk. This commit introduces the utils/conf2ael program, and an external config-file reader, for both normal config files, and for extensions.conf (context, exten, prio); It provides an API for programs outside of asterisk to use to play with the dialplan and config files. 2007-08-15 14:42 +0000 [r79558] Joshua Colp * /, main/rtp.c: Merged revisions 79553 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79553 | file | 2007-08-15 11:40:23 -0300 (Wed, 15 Aug 2007) | 6 lines (closes issue #10440) Reported by: irroot (closes issue #10454) Reported by: flo_turc Increase maximum timestamp skew to 120. 20 was apparently far too low. ........ 2007-08-15 14:27 +0000 [r79529] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 79527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79527 | mmichelson | 2007-08-15 09:26:40 -0500 (Wed, 15 Aug 2007) | 5 lines Fixed an error in the Russian language voicemail intro. (issue #10458, reported and patched by Oleh) ........ 2007-08-15 14:20 +0000 [r79524] Joshua Colp * /, channels/chan_sip.c: Merged revisions 79523 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79523 | file | 2007-08-15 11:18:44 -0300 (Wed, 15 Aug 2007) | 6 lines (closes issue #10456) Reported by: irroot Patches: sip_timeout.patch uploaded by irroot (license 52) Change hardcoded timer value to defined value. I'm doing this in 1.4 as well so if it needs to be changed in the future this place would not have been forgotten. ........ 2007-08-15 11:27 +0000 [r79507] Christian Richter * channels/chan_misdn.c, /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: Merged revisions 78936 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78936 | crichter | 2007-08-10 15:24:03 +0200 (Fr, 10 Aug 2007) | 1 line fixed a bug with the useruser information element. We send them now also in the disconnect message. ........ 2007-08-14 18:50 +0000 [r79437-79471] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 79470 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79470 | russell | 2007-08-14 13:49:10 -0500 (Tue, 14 Aug 2007) | 2 lines Fix another spot where an iax2_peer would be leaked if realtime was in use. ........ * /, channels/chan_iax2.c: Merged revisions 79436 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79436 | russell | 2007-08-14 12:31:39 -0500 (Tue, 14 Aug 2007) | 3 lines Fix some memory leaks throughout chan_iax2 related to the use of realtime. I found these while working on iax2_peer object reference tracking. ........ 2007-08-14 15:30 +0000 [r79403] Joshua Colp * /, res/res_features.c: Merged revisions 79397 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79397 | file | 2007-08-14 12:27:13 -0300 (Tue, 14 Aug 2007) | 4 lines (closes issue #10415) Reported by: atis Revert fix for #10327 as it causes more issues then it solves. ........ 2007-08-14 14:32 +0000 [r79392] Steve Murphy * pbx/ael/ael-test/ref.ael-vtest17, /, pbx/ael/ael-test/ref.ael-test11, pbx/ael/ael-test/ref.ael-test14, pbx/ael/ael-test/ael-test5/extensions.ael, pbx/ael/ael-test/ael-test6/extensions.ael, pbx/ael/ael-test/ref.ael-test19, pbx/ael/ael-test/ael-vtest21/extensions.ael, pbx/ael/ael-test/ael-vtest21 (added), pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-test2, pbx/pbx_ael.c, pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test4, utils/ael_main.c, pbx/ael/ael-test/ref.ael-test6, pbx/ael/ael-test/ref.ael-vtest21 (added), pbx/ael/ael-test/ref.ael-vtest13: Merged revisions 79255 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79255 | murf | 2007-08-13 11:49:54 -0600 (Mon, 13 Aug 2007) | 1 line This patch fixes bug 10411. I added a new regression test, some regression test cleanups ........ 2007-08-14 14:17 +0000 [r79379] Joshua Colp * main/channel.c: (closes issue #10427) Reported by: pj Two of the three places ast_waitfor_nandfds could branch off to did not clear outfd and exception. If the calling function did not clear these there was a chance they could get a false positive on testing to see whether they were set. 2007-08-14 13:46 +0000 [r79378] Steve Murphy * main/channel.c, channels/chan_zap.c: Don't ask me why, but waitfordigit will immediately return a 1 on my system, unless the outfd is initialized to -1 before calling the nandfds func 2007-08-13 21:59 +0000 [r79335] Joshua Colp * /, include/asterisk/speech.h, res/res_speech.c, apps/app_speech_utils.c: Merged revisions 79334 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79334 | file | 2007-08-13 18:57:20 -0300 (Mon, 13 Aug 2007) | 2 lines Instead of accepting a single DTMF character accept a full string. ........ 2007-08-13 21:44 +0000 [r79333] Tilghman Lesher * res/res_odbc.c: Only use the sanitysql if it's not zero-len 2007-08-13 20:40 +0000 [r79273-79306] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 79301 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79301 | russell | 2007-08-13 15:37:50 -0500 (Mon, 13 Aug 2007) | 3 lines Don't call find_peer in registry_authrequest with the pvt lock held to avoid a deadlock. ........ * /, channels/chan_iax2.c: Merged revisions 79276 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79276 | russell | 2007-08-13 15:18:30 -0500 (Mon, 13 Aug 2007) | 4 lines Release the pvt lock before calling find_peer in register_verify to avoid a deadlock. Also, remove some unnecessary locking in auth_fail that was only done recursively. ........ * /, channels/chan_iax2.c: Merged revisions 79274 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79274 | russell | 2007-08-13 15:02:57 -0500 (Mon, 13 Aug 2007) | 3 lines Don't call find_peer within update_registry with a pvt lock held. This can cause a deadlock as the code will eventually call find_callno. ........ * /, channels/chan_iax2.c: Merged revisions 79272 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79272 | russell | 2007-08-13 14:27:39 -0500 (Mon, 13 Aug 2007) | 9 lines I am fighting deadlocks in chan_iax2. I have tracked them down to a single core issue. You can not call find_callno() while holding a pvt lock as this function has to lock another (every) other pvt lock. Doing so can lead to a classic deadlock. So, I am tracking down all of the code paths where this can happen and fixing them. The fix I committed earlier today was along the same theme. This patch fixes some code down the path of authenticate_reply. ........ 2007-08-13 15:39 +0000 [r79238] Mark Michelson * CHANGES, apps/app_queue.c: Allow non-realtime queues to have realtime members (issue #10424, reported and patched by irroot) 2007-08-13 15:32 +0000 [r79222] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 79214 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79214 | russell | 2007-08-13 10:28:13 -0500 (Mon, 13 Aug 2007) | 4 lines Fix a potential deadlock in socket_process. check_provisioning can eventually call find_callno. You can't hold a pvt lock while calling find_callno because it goes through and locks every single one looking for a match. ........ 2007-08-13 14:55 +0000 [r79208] Joshua Colp * /, include/asterisk/speech.h, res/res_speech.c, apps/app_speech_utils.c: Merged revisions 79207 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79207 | file | 2007-08-13 11:51:09 -0300 (Mon, 13 Aug 2007) | 2 lines Add an API call to allow the engine to know that DTMF was received. ........ 2007-08-13 14:23 +0000 [r79176] Russell Bryant * main/channel.c, include/asterisk/channel.h: constify the return value of reason2str 2007-08-13 14:22 +0000 [r79175] Joshua Colp * channels/chan_jingle.c, channels/chan_phone.c, channels/chan_local.c, channels/chan_misdn.c, channels/chan_zap.c, /, channels/chan_sip.c, channels/chan_skinny.c, channels/chan_h323.c, channels/chan_gtalk.c, channels/chan_iax2.c, channels/chan_oss.c, channels/chan_mgcp.c: Merged revisions 79174 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79174 | file | 2007-08-13 11:18:04 -0300 (Mon, 13 Aug 2007) | 4 lines (closes issue #10437) Reported by: haklin Don't set the callerid name and number a second time on a newly created channel. ast_channel_alloc itself already sets it and setting it twice would cause a memory leak. ........ 2007-08-11 05:28 +0000 [r79147] Tilghman Lesher * /, res/res_odbc.c: Merged revisions 79142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79142 | tilghman | 2007-08-11 00:23:04 -0500 (Sat, 11 Aug 2007) | 2 lines Ensure the connection gets marked as used at allocation time (closes issue #10429, report and fix by mnicholson) ........ 2007-08-10 21:29 +0000 [r79109] Jason Parker * channels/chan_skinny.c: Use localized softkey labels. Add some information about localization "codes". 2007-08-10 21:03 +0000 [r79100] Steve Murphy * main/channel.c, pbx/pbx_spool.c, include/asterisk/channel.h: Merged revisions 79099 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79099 | murf | 2007-08-10 14:53:43 -0600 (Fri, 10 Aug 2007) | 1 line From a user complaint on #asterisk, I have forced pbx_spool to explain what reason codes mean, when they are logged ........ 2007-08-10 20:48 +0000 [r79098] Russell Bryant * funcs/func_devstate.c: Store custom device states in astdb so that they will persist a restart. As a side benefit, this simplifies the code a bit, too. 2007-08-10 18:37 +0000 [r79074] Joshua Colp * main/dial.c: Bring up to date with poll changes. 2007-08-10 18:35 +0000 [r79045-79068] Steve Murphy * main/cdr.c, /: Merged revisions 79049 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79049 | murf | 2007-08-10 12:25:51 -0600 (Fri, 10 Aug 2007) | 1 line Re bug behavior mentioned in #asterisk, made this tweak to code, to prevent hundreds of log messages from being generated ........ * /: oops. forgot to commit the prop change on . * main/cdr.c: Merged revisions 79044 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79044 | murf | 2007-08-10 11:43:49 -0600 (Fri, 10 Aug 2007) | 1 line This will help debug; from a question asked on #asterisk ........ 2007-08-10 16:24 +0000 [r79005-79027] Russell Bryant * include/asterisk/devicestate.h, apps/app_meetme.c, res/res_features.c, main/devicestate.c, main/event.c, funcs/func_devstate.c: Merge a set of device state improvements from team/russell/events. The way a device state change propagates is kind of silly, in my opinion. A device state provider calls a function that indicates that the state of a device has changed. Then, another thread goes back and calls a callback for the device state provider to find out what the new state is before it can go send it off to whoever cares. I have changed it so that you can include the state that the device has changed to in the first function call from the device state provider. This removes the need to have to call the callback, which locks up critical containers to go find out what the state changed to. This change set changes the "simple" device state providers to use the new method. This includes parking, meetme, and SLA. I have also mostly converted chan_agent in my branch, but still have some more things to think through before presenting the plan for converting channel drivers to ensure all of the right events get generated ... * /, include/asterisk/lock.h: Merged revisions 78995 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78995 | russell | 2007-08-10 10:20:09 -0500 (Fri, 10 Aug 2007) | 4 lines The last set of changes that I made to "core show locks" made it not able to track mutexes unless they were declared using AST_MUTEX_DEFINE_STATIC. Locks initialized with ast_mutex_init() were not tracked. It should work now. ........ 2007-08-10 14:17 +0000 [r78952-78956] Joshua Colp * /, main/file.c: Merged revisions 78955 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78955 | file | 2007-08-10 11:15:53 -0300 (Fri, 10 Aug 2007) | 2 lines Don't bother having the core pass through or emulate begin DTMF frames when in an ast_waitstream. It only cares about the end of DTMF. ........ * /, configs/queues.conf.sample: Merged revisions 78951 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78951 | file | 2007-08-10 10:49:19 -0300 (Fri, 10 Aug 2007) | 4 lines (closes issue #10422) Reported by: bhowell Add note to sample configuration about module load order and how it can cause perfectly good queue members to be marked as invalid. ........ 2007-08-09 23:49 +0000 [r78908] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 78907 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78907 | mmichelson | 2007-08-09 18:47:00 -0500 (Thu, 09 Aug 2007) | 4 lines Improved a bit of logic regarding comma-separated mailboxes in has_voicemail. Also added some braces to some compound if statements since unbraced if statements scare me in general. ........ 2007-08-09 23:32 +0000 [r78906] Steve Murphy * Makefile, /: Merged revisions 78891 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78891 | murf | 2007-08-09 17:10:46 -0600 (Thu, 09 Aug 2007) | 1 line This fixes bug 10416; thanks to mvanbaak for the pretty output ........ 2007-08-09 22:19 +0000 [r78861-78862] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 78859 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78859 | mmichelson | 2007-08-09 16:51:17 -0500 (Thu, 09 Aug 2007) | 9 lines Quite a few changes regarding IMAP storage. 1. instead of using inboxcount as the core message counting function, we use messagecount instead. This makes it possible to count messages in folders besides just INBOX and Old. 2. inboxcount and hasvoicemail now use messagecount as their means of determining return values. 3. Added a copy_message function for IMAP storage. Unfortunately I don't have the means to test it, but it seems like a pretty straightforward function. 4. Removed a #ifndef IMAP_STORAGE and matching #endif from leave_voicemail for a couple of reasons. One, we want to support copying mail to multiple IMAP boxes, and two, IMAP was broken because a STORE macro had been moved into this section of code. ........ 2007-08-09 20:07 +0000 [r78829] Russell Bryant * apps/app_minivm.c: Don't use strncpy for moving a chunk of memory to another that is overlapping. This was found by running Asterisk under valgrind. 2007-08-09 19:35 +0000 [r78718-78824] Russell Bryant * channels/chan_sip.c: When looking up a mailbox, use the default context if not specified as something else * channels/chan_sip.c: Restore the ability to have multiple mailboxes listed for the mailbox option in sip.conf. chan_sip now maintains separate internal MWI subscriptions for each one. * /, apps/app_voicemail.c: Merged revisions 78778 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78778 | russell | 2007-08-09 12:58:31 -0500 (Thu, 09 Aug 2007) | 1 line add a comment to indicate that inboxcount for ODBC_STORAGE needs to be fixed to support multiple mailboxes ........ * /, apps/app_voicemail.c: Merged revisions 78749 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78749 | russell | 2007-08-09 12:24:40 -0500 (Thu, 09 Aug 2007) | 9 lines Fix subscriptions to multiple mailboxes for ODBC_STORAGE. Also, leave a comment for this to be fixed for IMAP_STORAGE, as well. I left IMAP alone since I know MarkM was working on this code right now for another reason. This is broken even worse in trunk, but for a different reason. The fact that the mailbox option supported multiple mailboxes is completely not obvious from the code in the channel drivers. Anyway, I will fix that in another commit ... ........ * channels/chan_zap.c, channels/chan_sip.c, include/asterisk/event_defs.h, channels/chan_iax2.c, channels/chan_mgcp.c, apps/app_voicemail.c: Fix a problem that I had introduced into MWI handling. I had ignored the mailbox context. Now, all related MWI event dealings pay attention to the context as well. * /, apps/app_meetme.c: Merged revisions 78717 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78717 | russell | 2007-08-09 11:12:57 -0500 (Thu, 09 Aug 2007) | 7 lines Fix a problem with the combination of the 'F' option to pass DTMF through a conference and options that use DTMF to activate various features. The problem was that the BEGIN frame would be passed through, but the END frame would get intercepted to activate a feature. Then, the other conference members would hear DTMF for forever, which they didn't seem to like very much. (closes issue #10400, reported by stevefeinstein, fixed by me) ........ 2007-08-08 22:05 +0000 [r78649-78686] Joshua Colp * configure: Regenerate configure script. This actually just updated the revision number... since my last merge changed it to an older number, while it was in fact generated from a much newer revision. * channels/chan_skinny.c: Minor fix for building under dev mode when byteswapping macro header files are not available. * apps/app_dial.c, channels/chan_zap.c, channels/chan_sip.c, include/asterisk/autoconfig.h.in, channels/chan_agent.c, configure.ac, include/asterisk/channel.h, channels/chan_gtalk.c, channels/chan_oss.c, main/rtp.c, main/channel.c, channels/chan_jingle.c, channels/chan_phone.c, channels/chan_misdn.c, channels/chan_skinny.c, configure, channels/chan_features.c, channels/chan_h323.c, channels/chan_alsa.c, channels/chan_nbs.c, channels/chan_mgcp.c: Add support for using epoll instead of poll. This should increase scalability and is done in such a way that we should be able to add support for other poll() replacements. * channels/chan_zap.c: HAVEL_SS7 should be HAVE_SS7. Reported by kwallace. * main/channel.c, include/asterisk/audiohook.h (added), funcs/func_volume.c (added), main/Makefile, main/slinfactory.c, include/asterisk/chanspy.h (removed), include/asterisk/channel.h, main/audiohook.c (added), apps/app_chanspy.c, apps/app_mixmonitor.c, include/asterisk/slinfactory.h: Merge audiohooks branch into trunk. This is a new API for developers to listen and manipulate the audio going through a channel. 2007-08-08 19:30 +0000 [r78648] Jason Parker * /, doc/jabber.txt: Merged revisions 78646 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78646 | qwell | 2007-08-08 14:29:42 -0500 (Wed, 08 Aug 2007) | 2 lines Fix mogs email address. ........ 2007-08-08 19:03 +0000 [r78637] Joshua Colp * channels/chan_iax2.c: Correct spelling. s/threaads/threads/ 2007-08-08 18:34 +0000 [r78590-78635] Mark Michelson * /, apps/app_queue.c: Merged revisions 78575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78575 | mmichelson | 2007-08-08 09:26:36 -0500 (Wed, 08 Aug 2007) | 4 lines Changing a bit of logic so that someone will NEVER exit the queue on timeout unless they have enabled the 'n' option. This commit relates to issue #10320. Thanks to jfitzgibbon for detailing the idea behind this code change. ........ 2007-08-08 13:52 +0000 [r78570] Joshua Colp * /, configs/sip.conf.sample: Merged revisions 78569 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78569 | file | 2007-08-08 10:51:01 -0300 (Wed, 08 Aug 2007) | 4 lines (closes issue #10335) Reported by: adamgundy Update sip.conf to include another scenario where directrtpsetup will fail. ........ 2007-08-07 23:04 +0000 [r78541] Russell Bryant * main/pbx.c, pbx/pbx_spool.c, main/sha1.c, res/res_features.c, res/res_crypto.c, utils/smsq.c, include/asterisk/features.h: Add another big set of doxygen documentation improvements from snuffy. (closes issue #9892) (closes issue #10395) 2007-08-07 22:13 +0000 [r78521] Joshua Colp * main/manager.c, include/asterisk/manager.h: Use the linkedlists.h macros for the manager action list. 2007-08-07 21:00 +0000 [r78489] Russell Bryant * res/res_config_odbc.c, /: Merged revisions 78488 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78488 | russell | 2007-08-07 15:57:54 -0500 (Tue, 07 Aug 2007) | 2 lines Fix the build of this module on 64-bit platforms ........ 2007-08-07 19:44 +0000 [r78451] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 78450 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78450 | mmichelson | 2007-08-07 14:43:57 -0500 (Tue, 07 Aug 2007) | 5 lines The logic behind inboxcount's return value was reversed in has_voicemail and message_count. (closes issue #10401, reported by st1710, patched by me) ........ 2007-08-07 19:36 +0000 [r78442] Tilghman Lesher * /, res/res_odbc.c: Merged revisions 78437 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78437 | tilghman | 2007-08-07 14:34:25 -0500 (Tue, 07 Aug 2007) | 2 lines Don't free the environment handle when the connection fails, because other connections might be depending upon it ........ 2007-08-07 19:14 +0000 [r78417] Tilghman Lesher * res/res_config_odbc.c, /, apps/app_directory.c, apps/app_voicemail.c: Merged revisions 78415 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78415 | tilghman | 2007-08-07 14:09:38 -0500 (Tue, 07 Aug 2007) | 2 lines Reconnection doesn't happen automatically when a DB goes down (fixes issue #9389) ........ 2007-08-07 18:26 +0000 [r78378] Jason Parker * /, channels/chan_skinny.c: Merged revisions 78375 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78375 | qwell | 2007-08-07 13:25:15 -0500 (Tue, 07 Aug 2007) | 3 lines Properly check the capabilities count to avoid a segfault. (ASA-2007-019) ........ 2007-08-07 17:46 +0000 [r78372] Russell Bryant * channels/chan_zap.c, /: Merged revisions 78371 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r78371 | russell | 2007-08-07 12:45:30 -0500 (Tue, 07 Aug 2007) | 12 lines Merged revisions 78370 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r78370 | russell | 2007-08-07 12:44:04 -0500 (Tue, 07 Aug 2007) | 4 lines Revert patch committed for issue #9660. It broke E&M trunks. (closes issue #10360) (closes issue #10364) ........ ................ 2007-08-07 16:17 +0000 [r78346-78347] Joshua Colp * channels/chan_zap.c: Can't forget outsignaling! * channels/chan_zap.c: Just for jsmith... make signaling a valid option that acts like signalling. 2007-08-07 16:04 +0000 [r78342] Russell Bryant * res/res_eventtest.c (removed): Remove some test code from trunk as it doesn't need to be here. I'm just going to keep it with a bunch of other changes i have sitting in a branch. 2007-08-07 15:40 +0000 [r78338] Joshua Colp * main/frame.c: (closes issue #10225) Reported by: klaus3000 Clean up AST_FORMAT_LIST list. It may have mattered in the old days to have undefined entries but these days it does not. 2007-08-06 23:00 +0000 [r78312] Jason Parker * channels/chan_agent.c: Add a TalkingToChan to the response of the "agents" manager action. This is similar to the existing "talking to" that you see what using the "agent show" CLI command. Closes issue #10102 2007-08-06 21:59 +0000 [r78276-78279] Joshua Colp * apps/app_senddtmf.c: Fix bug where a NULL timeout would make things explode if SendDTMF was called with it. * apps/app_dial.c, main/channel.c, include/asterisk/app.h, res/res_features.c, apps/app_test.c, main/app.c, include/asterisk/channel.h, apps/app_senddtmf.c: Extend the ast_senddigit and ast_dtmf_stream API calls to allow the duration of the DTMF digit(s) to be specified and make the SendDTMF application have the capability to use it. * main/channel.c, /: Merged revisions 78275 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78275 | file | 2007-08-06 18:41:13 -0300 (Mon, 06 Aug 2007) | 2 lines Add additional DTMF log messages to help when debugging issues. ........ 2007-08-06 20:45 +0000 [r78243] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 78242 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78242 | russell | 2007-08-06 15:44:09 -0500 (Mon, 06 Aug 2007) | 4 lines Fix an issue where dynamic threads can get free'd, but still exist in the dynamic thread list. (closes issue #10392, patch from Mihai, with credit to his colleague, Pete) ........ 2007-08-06 19:52 +0000 [r78227] Doug Bailey * main/tdd.c, include/asterisk/fskmodem.h, main/callerid.c, main/fskmodem.c: Change the fsk filter used in CID and TDD decode to an integer based implementation 2007-08-06 17:51 +0000 [r78186-78192] Mark Michelson * channels/chan_sip.c: Fixing a compiler warning which warns that a variable may be used unitialized. Thanks to mvanbaak for pointing this out. * /, channels/chan_sip.c, include/asterisk/config.h, main/config.c: Merged revisions 78103 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78103 | mmichelson | 2007-08-03 15:25:22 -0500 (Fri, 03 Aug 2007) | 7 lines Changed the behavior of sip's realtime_peer function to match the corresponding way of matching for non-realtime peers. Now matches are made on both the IP address and port number, or if the insecure setting is set to "port" then just match on the IP address. In order to accomplish this, I also added a new API call, ast_category_root, which returns the first variable of an ast_category struct ........ 2007-08-06 16:51 +0000 [r78185] Russell Bryant * /, include/asterisk/linkedlists.h: Merged revisions 78184 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78184 | russell | 2007-08-06 11:50:54 -0500 (Mon, 06 Aug 2007) | 5 lines Fix the return value of AST_LIST_REMOVE(). This shouldn't be causing any problems, though, because the only code that uses the return value only checks to see if it is NULL. (closes issue #10390, pointed out by mihai) ........ 2007-08-06 16:34 +0000 [r78183] Joshua Colp * /, channels/chan_sip.c: Merged revisions 78182 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78182 | file | 2007-08-06 13:32:44 -0300 (Mon, 06 Aug 2007) | 2 lines It is possible for a transfer to occur before the remote device has our tag in which case they send none in the transfer. In this case we need to not fail the transfer dialog lookup. ........ 2007-08-06 16:31 +0000 [r78179-78181] Jason Parker * /, main/config.c: Merged revisions 78180 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #9938) ........ r78180 | qwell | 2007-08-06 11:30:51 -0500 (Mon, 06 Aug 2007) | 5 lines Fix an issue with using UpdateConfig (manager action) where escaped semicolons in a config would be converted to just semicolons (\; to ;) Issue 9938 ........ * channels/chan_skinny.c, configs/skinny.conf.sample: Implement setvar functionality in chan_skinny Closes issue #10379, patch by mvanbaak. 2007-08-06 15:28 +0000 [r78167-78173] Joshua Colp * /, main/rtp.c: Merged revisions 78172 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78172 | file | 2007-08-06 12:27:24 -0300 (Mon, 06 Aug 2007) | 4 lines (closes issue #10355) Reported by: wdecarne Now that we pass through RTP timestamp information we need to make the allowed timestamp skew considerably less. There are situations where a source may change and due to the timestamp difference the receiver will experience an audio gap since we did not indicate by setting the marker bit that the source changed. ........ * apps/app_externalivr.c: (closes issue #10381) Reported by: yehavi Use the filename we parsed using the standard parsing when launching the application specified to ExternalIVR. * /, configure, configure.ac: Merged revisions 78166 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78166 | file | 2007-08-06 11:18:20 -0300 (Mon, 06 Aug 2007) | 4 lines (closes issue #10383) Reported by: rizzo Include stdlib.h so NULL gets defined for gethostbyname_r checks. ........ 2007-08-05 04:16 +0000 [r78142-78144] Russell Bryant * /, include/asterisk/lock.h: Merged revisions 78143 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78143 | russell | 2007-08-04 23:15:31 -0500 (Sat, 04 Aug 2007) | 2 lines Fix compilation failure when MALLOC_DEBUG is enabled, but DEBUG_THREADS is not ........ * apps/app_exec.c: Make this module build on my mac 2007-08-05 03:42 +0000 [r78140] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 78139 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78139 | tilghman | 2007-08-04 22:29:01 -0500 (Sat, 04 Aug 2007) | 2 lines If peer is not found, the error message is misleading (should be peer not found, not ACL failure) ........ 2007-08-05 03:14 +0000 [r78138] Russell Bryant * include/asterisk/linkedlists.h: Fix building res_crypto on systems that init locks with constructors. The problem was that res_crypto now has a RWLIST named "keys". The macro for defining this list defines a function used as a constructor for the list called "init_keys". However, there was another function called init_keys in this module for a CLI command. The fix is just to prepend the generated functions with underscores. 2007-08-03 20:21 +0000 [r78029-78102] Russell Bryant * /, apps/app_voicemail.c: Merged revisions 78101 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78101 | russell | 2007-08-03 15:14:06 -0500 (Fri, 03 Aug 2007) | 10 lines (closes issue #10194) Reported by: blitzrage Patches: bug0010194 uploaded by vovochka Tested by: blitzrage Fix a problem when you call Voicemail() with multiple mailboxes specified and ODBC_STORAGE is in use. The audio part of the message was only given to the first mailbox specified. ........ * /, main/utils.c, include/asterisk/lock.h, main/astmm.c: Merged revisions 78095 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78095 | russell | 2007-08-03 14:39:49 -0500 (Fri, 03 Aug 2007) | 28 lines Add some improvements to lock debugging. These changes take effect with DEBUG_THREADS enabled and provide the following: * This will keep track of which locks are held by which thread as well as which lock a thread is waiting for in a thread-local data structure. A reference to this structure is available on the stack in the dummy_start() function, which is the common entry point for all threads. This information can be easily retrieved using gdb if you switch to the dummy_start() stack frame of any thread and print the contents of the lock_info variable. * All of the thread-local structures for keeping track of this lock information are also stored in a list so that the information can be dumped to the CLI using the "core show locks" CLI command. This introduces a little bit of a performance hit as it requires additional underlying locking operations inside of every lock/unlock on an ast_mutex. However, the benefits of having this information available at the CLI is huge, especially considering this is only done in DEBUG_THREADS mode. It means that in most cases where we debug deadlocks, we no longer have to request access to the machine to analyze the contents of ast_mutex_t structures. We can now just ask them to get the output of "core show locks", which gives us all of the information we needed in most cases. I also had to make some additional changes to astmm.c to make this work when both MALLOC_DEBUG and DEBUG_THREADS are enabled. I disabled tracking of one of the locks in astmm.c because it gets used inside the replacement memory allocation routines, and the lock tracking code allocates memory. This caused infinite recursion. ........ * /, channels/chan_iax2.c: Merged revisions 78063 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78063 | russell | 2007-08-03 12:01:07 -0500 (Fri, 03 Aug 2007) | 4 lines Only pass through HOLD and UNHOLD control frames when the mohinterpret option is set to "passthrough". This was pointed out by Kevin in the middle of a training session. ........ * /, channels/chan_iax2.c: Merged revisions 78028 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78028 | russell | 2007-08-02 21:04:22 -0500 (Thu, 02 Aug 2007) | 6 lines Don't reuse the timespec that was set to 0 in the previous timedwait as it will just return immediately. Also, fix some logic so the thread's lock isn't unlocked twice in the weird case of dynamic threads getting acquired right after a timeout. (pointed out by SteveK) ........ 2007-08-02 21:54 +0000 [r77994-77997] Jason Parker * /, channels/chan_skinny.c, configs/skinny.conf.sample: Merged revisions 77996 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #9779) ........ r77996 | qwell | 2007-08-02 16:53:39 -0500 (Thu, 02 Aug 2007) | 5 lines Make sure we actually allow 6 chars to be sent. Also make note of the "A" option of date format. Issue 9779, modifications by DEA, wedhorn, and myself. ........ * /, channels/chan_skinny.c: Merged revisions 77993 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #10325) ........ r77993 | qwell | 2007-08-02 15:22:40 -0500 (Thu, 02 Aug 2007) | 5 lines If a device disconnects, the session will go away. If this happens during call setup, we need to give up. Issue 10325. ........ 2007-08-02 19:26 +0000 [r77950] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 77949 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77949 | russell | 2007-08-02 14:25:14 -0500 (Thu, 02 Aug 2007) | 5 lines Fix the case where a dynamic thread times out waiting for something to do during the first time it runs. This shouldn't ever happen, but we should account for it anyway. (pointed out by pete, who works with mihai) ........ 2007-08-02 18:43 +0000 [r77948] Jason Parker * /, channels/chan_skinny.c: Merged revisions 77947 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #10299) ........ r77947 | qwell | 2007-08-02 13:42:36 -0500 (Thu, 02 Aug 2007) | 5 lines Make sure we clear the prompt status message on a hangup. Also rearrange messages to better fit with what a wireshark trace shows it should be. Issue 10299, initial patch and solution by sbisker, modified by me to fit with wireshark trace. ........ 2007-08-02 18:32 +0000 [r77946] Steve Murphy * /, main/fskmodem.c: Merged revisions 77945 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r77945 | murf | 2007-08-02 12:21:40 -0600 (Thu, 02 Aug 2007) | 9 lines Merged revisions 77942 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r77942 | murf | 2007-08-02 11:56:37 -0600 (Thu, 02 Aug 2007) | 1 line This patch hopefully solves 10141; The user is running with it, and it doesn't appear to harm asterisk's operation, and may prevent a crash. I'll store it in 1.2, as we have shut down support on 1.2, but since I developed the patch before support finished, and it might affect 1.4 and trunk, I'm going ahead with it. ........ ................ 2007-08-02 18:05 +0000 [r77940-77944] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 77943 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77943 | russell | 2007-08-02 13:04:43 -0500 (Thu, 02 Aug 2007) | 9 lines Fix another race condition in the handling of dynamic threads. If the dynamic thread timed out waiting for something to do, but was acquired to perform an action immediately afterwords, then wait on the condition again to give the other thread a chance to finish setting up the data for what action this thread should perform. Otherwise, if it immediately continues, it will perform the wrong action. (reported on IRC by mihai, patch by me) (related to issue #10289) ........ * channels/chan_iax2.c: Fix an issue that Simon pointed out to me on IRC. There were cases in the trunk version of find_idle_thread() where the old full frame processing information was not cleared out. This would have caused full frames to get deferred for processing by threads that weren't actually processing frames for that call. Nice catch!! * /, channels/chan_iax2.c: Merged revisions 77939 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77939 | russell | 2007-08-02 11:56:04 -0500 (Thu, 02 Aug 2007) | 4 lines Add another sanity check to vnak_retransmit(). This check ensures that frames that have already been marked for deletion don't get retransmitted. (closes issue #10361, patch from mihai) ........ 2007-08-02 15:16 +0000 [r77891-77895] Jason Parker * /, channels/chan_skinny.c: Merged revisions 77894 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #10358) ........ r77894 | qwell | 2007-08-02 10:15:45 -0500 (Thu, 02 Aug 2007) | 5 lines Make sure that we show the correct extension if dialed from a macro "From: 5555" rather than "From: s" Issue 10358, initial patch by DEA, reworked by me to use S_OR, tested by sbisker ........ * /, channels/chan_skinny.c: Merged revisions 77890 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #10291) ........ r77890 | qwell | 2007-08-01 17:28:56 -0500 (Wed, 01 Aug 2007) | 4 lines Put in some additional debug information for softkey/stimulus messages. Issue 10291, patch by DEA. ........ 2007-08-01 22:24 +0000 [r77889] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 77887 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77887 | russell | 2007-08-01 17:16:17 -0500 (Wed, 01 Aug 2007) | 23 lines Fix some race conditions which have been causing weird problems in chan_iax2. The most notable problem is that people have been seeing storms of VNAK frames being sent due to really old frames mysteriously being in the retransmission queue and never getting removed. It was possible that a dynamic thread got created, but did not acquire its lock before the thread that created it signals it to perform an action. When this happens, the thread will sleep until it hits a timeout, and then get destroyed. So, the action never gets performed and in some cases, means a frame doesn't get transmitted and never gets freed since the scheduler never gets a chance to reschedule transmission. Another less severe race condition is in the handling of a timeout for a dynamic thread. It was possible for it to be acquired to perform at action at the same time that it hit a timeout. When this occurs, whatever action it was acquired for would never get performed. (patch contributed by Mihai and SteveK) (closes issue #10289) (closes issue #10248) (closes issue #10232) (possibly related to issue #10359) ........ 2007-08-01 22:19 +0000 [r77888] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 77886 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77886 | tilghman | 2007-08-01 17:14:47 -0500 (Wed, 01 Aug 2007) | 2 lines Voicemail with ODBC_STORAGE defined does not compile cleanly (missing def) ........ 2007-08-01 21:12 +0000 [r77879-77884] Jason Parker * /, channels/chan_skinny.c: Merged revisions 77883 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77883 | qwell | 2007-08-01 16:08:42 -0500 (Wed, 01 Aug 2007) | 7 lines Fix an issue that caused one-way audio on some newer devices (specifically the 7921), due to sending packets in the wrong order during hangup. Also make sure we clear tones/messages on the correct line/instance. Issue 10291, patch by DEA, tested by sbisker and myself. ........ * apps/app_queue.c, doc/tex/queuelog.tex: Add the Ring time in the CONNECT on the queue_log and on the Manager event AgentConnect Closes issue #10349, patch by eliel 2007-08-01 19:37 +0000 [r77864-77878] Joshua Colp * main/pbx.c, configure, configure.ac, main/asterisk.c: Instead of adding the SOLARIS check to each HAVE_SYSINFO check let's just make the sysinfo autoconf logic a bit pickier about what it considers a usable sysinfo. * main/pbx.c, main/asterisk.c: Solaris does not have a sysinfo like we know of on Linux. * configure, configure.ac: Don't look for /dev/urandom when cross compiling. Just assume it is not available. * /, utils/smsq.c, channels/chan_iax2.c, include/asterisk/threadstorage.h, channels/chan_mgcp.c, apps/app_voicemail.c: Merged revisions 77869 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77869 | file | 2007-08-01 14:56:59 -0300 (Wed, 01 Aug 2007) | 2 lines Add some fixes for building on Solaris. ........ * /, main/utils.c: Merged revisions 77867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77867 | file | 2007-08-01 14:52:11 -0300 (Wed, 01 Aug 2007) | 2 lines Whoops, I meant R_5 not R5. ........ * /, configure, configure.ac: Merged revisions 77865 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77865 | file | 2007-08-01 14:42:52 -0300 (Wed, 01 Aug 2007) | 2 lines And for my last trick... make sure that if gethostbyname_r is exported by a library that it is used. ........ * /, configure, include/asterisk/autoconfig.h.in, configure.ac, main/utils.c: Merged revisions 77863 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77863 | file | 2007-08-01 14:22:35 -0300 (Wed, 01 Aug 2007) | 2 lines Extend autoconf logic to determine which version of gethostbyname_r is on the system. ........ 2007-08-01 15:39 +0000 [r77858] Russell Bryant * apps/app_dial.c, main/autoservice.c, main/pbx.c, apps/app_osplookup.c, channels/chan_local.c, channels/chan_vpb.cc, apps/app_meetme.c, res/res_features.c, apps/app_zapras.c, apps/app_macro.c, pbx/pbx_dundi.c, apps/app_queue.c: Convert code that checks the _softhangup member of ast_channel directory to use the ast_check_hangup() funciton. This function takes scheduled hangups into account. (closes issue #10230, patch by Juggie) 2007-08-01 15:28 +0000 [r77857] Joshua Colp * main/cli.c: Convert CLI helpers list to rwlist. 2007-08-01 14:09 +0000 [r77853-77855] Mark Michelson * /, apps/app_queue.c: Merged revisions 77854 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77854 | mmichelson | 2007-08-01 09:08:57 -0500 (Wed, 01 Aug 2007) | 8 lines Fixes an issue I introduced to queues wherein a queue with joinempty=yes would kick people out of the queue because of erroneously thinking the 'n' option was in use. (closes issue #10320, reported by jfitzgibbon, patched by me, tested by blitzrage and me) Thank you blitzrage for all the testing you've done lately with queues! It's much appreciated! ........ * /, apps/app_queue.c: Merged revisions 77852 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77852 | mmichelson | 2007-08-01 08:59:59 -0500 (Wed, 01 Aug 2007) | 7 lines If a queue uses dynamic realtime members, then the member list should be updated after each attempt to call the queue. This fixes an issue where if a caller calls into a queue where no one is logged in, they would wait forever even if a member logged in at some point. (closes issue #10346, reported by and tested by blitzrage, patched by me) ........ 2007-08-01 04:36 +0000 [r77851] Tilghman Lesher * res/res_agi.c: Twould help if we actually defined ->mod before comparing against it (reported and fixed by Juggie via IRC). 2007-07-31 21:33 +0000 [r77847] Steve Murphy * /, contrib/scripts/ast_grab_core: Merged revisions 77844 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r77844 | murf | 2007-07-31 14:59:10 -0600 (Tue, 31 Jul 2007) | 9 lines Merged revisions 77842 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r77842 | murf | 2007-07-31 13:19:35 -0600 (Tue, 31 Jul 2007) | 1 line This probably isn't super-general, but it's a first stab at using kill -11 to generate a core file instead of gcore. ........ ................ 2007-07-31 18:50 +0000 [r77834-77838] Tilghman Lesher * funcs/func_lock.c, CHANGES: Add some documentation detailing an aspect of dialplan functions, as requested by Russell * funcs/func_lock.c (added), UPGRADE.txt: Add func_lock, which creates dialplan mutexes, and note that the Macro apps are now deprecated. (Closes issue #10264) 2007-07-31 16:21 +0000 [r77833] Joshua Colp * /, include/asterisk/speech.h, res/res_speech.c: Merged revisions 77831 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77831 | file | 2007-07-31 13:17:09 -0300 (Tue, 31 Jul 2007) | 2 lines Add a flag to the speech API that allows an engine to set whether it received results or not. ........ 2007-07-31 15:59 +0000 [r77829] Steve Murphy * channels/chan_sip.c: thanks to Russel, for pointing out that the dialoglist_lock/unlock routines also need to be macros if DETECT_DEADLOCKS is set 2007-07-31 15:54 +0000 [r77828] Kevin P. Fleming * build_tools/cflags.xml, /: Merged revisions 77827 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77827 | kpfleming | 2007-07-31 10:53:42 -0500 (Tue, 31 Jul 2007) | 2 lines DETECT_DEADLOCKS can't be enabled without DEBUG_THREADS or it does nothing ........ 2007-07-31 15:22 +0000 [r77825] Mark Michelson * /, channels/chan_sip.c: Merged revisions 77824 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77824 | mmichelson | 2007-07-31 10:21:22 -0500 (Tue, 31 Jul 2007) | 6 lines This patch makes Asterisk send 100 Trying provisional responses upon receipt of re-invites. This makes it so that if there are two or more Asterisk servers between endpoints, the Asterisk servers will not keep retransmitting the re-invites. (closes issue #10274, reported by cstadlmann, patched by me with approval from file) ........ 2007-07-31 15:01 +0000 [r77819-77821] Kevin P. Fleming * channels/chan_sip.c: there is no use in having functions that have no code in them, and hide the locking info when DEBUG_THREADS is enabled... i could have fixed this to be dependent on DEBUG_THREADS, but it would be just as easy for someone to add their test/debugging code to the macros as it would have been to the functions * channels/chan_sip.c: use a different method for overriding the send_digit_begin pointer, as the old one fails to compile on my 64-bit system with gcc-4.1 and --enable-dev-mode turned on * apps/app_senddtmf.c: umm... let's build with --enable-dev-mode, mmkay? 2007-07-31 03:32 +0000 [r77810] Steve Murphy * channels/chan_sip.c: Discovered in experiments on core files: if you wrap the lock and unlock calls with sip_pvt_lock and sip_pvt_unlock, you lose the tracing info you would normally get via DETECT_DEADLOCKS; so I turn these two functions into macros when DETECT_DEADLOCKS is called. This way, you get meaningful stuff in the file and func slots in the lock_info struct. 2007-07-31 01:10 +0000 [r77808] Tilghman Lesher * apps/app_meetme.c, apps/app_dictate.c, apps/app_record.c, apps/app_authenticate.c, apps/app_sayunixtime.c, apps/app_userevent.c, apps/app_chanisavail.c, apps/app_image.c, apps/app_followme.c, apps/app_controlplayback.c, funcs/func_enum.c, funcs/func_odbc.c, apps/app_minivm.c, res/res_agi.c, apps/app_amd.c, apps/app_url.c, apps/app_directory.c, apps/app_rpt.c, apps/app_parkandannounce.c, apps/app_read.c, funcs/func_timeout.c, apps/app_page.c, apps/app_festival.c, apps/app_privacy.c, apps/app_waitforsilence.c, apps/app_disa.c, apps/app_transfer.c, apps/app_talkdetect.c, apps/app_queue.c, apps/app_playback.c, res/res_monitor.c, apps/app_speech_utils.c, funcs/func_curl.c, funcs/func_channel.c, funcs/func_cdr.c, apps/app_sendtext.c, apps/app_macro.c, apps/app_sms.c, apps/app_senddtmf.c, apps/app_mixmonitor.c, apps/app_chanspy.c, apps/app_stack.c, apps/app_voicemail.c: Mostly cleanup of documentation to substitute the pipe with the comma, but a few other formatting cleanups, too. 2007-07-30 20:42 +0000 [r77801] Joshua Colp * main/dial.c, include/asterisk/dial.h: Add support for call forwarding and timeouts to the dialing API. 2007-07-30 20:36 +0000 [r77797-77800] Russell Bryant * channels/chan_iax2.c: Change another unnecessary use of the increment operator to explicitly set the var to 1 * channels/chan_iax2.c: Explicitly set a variable to 1 instead of using the increment operator. * /, channels/chan_iax2.c: Merged revisions 77794 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77794 | russell | 2007-07-30 15:16:43 -0500 (Mon, 30 Jul 2007) | 8 lines Fix an issue that could potentially cause corruption of the global iax frame queue. In the network_thread() loop, it traverses the list using the AST_LIST_TRAVERSE_SAFE macro. However, to remove an element of the list within this loop, it used AST_LIST_REMOVE, instead of AST_LIST_REMOVE_CURRENT, which I believe could leave some of the internal variables of the SAFE macro invalid. Mihai says that he already made this change in his local copy and it didn't help his VNAK storm issues, but I still think it's wrong. :) ........ 2007-07-30 20:19 +0000 [r77796] Jason Parker * /, main/say.c: Merged revisions 77795 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #10083) ........ r77795 | qwell | 2007-07-30 15:17:08 -0500 (Mon, 30 Jul 2007) | 6 lines Applications like SayAlpha() should not hang up the channel if you request an "unknown" character such as a comma. Instead, skip the character and move on. Issue 10083, initial patch by jsmith, modified by me. ........ 2007-07-30 19:42 +0000 [r77793] Luigi Rizzo * main/channel.c: print formats as 0x%x instead of %d in a warning message. Being bitmasks, it is a lot easier to read this way. 2007-07-30 19:39 +0000 [r77789-77792] Russell Bryant * res/res_agi.c: Fix the return value of ast_agi_fdprintf() to include the result from ast_carefulwrite() * res/res_agi.c: Improve ast_agi_fdprintf() by using the ast_str() API. * Use a thread local ast_str for building the string that will be written out to the console for debug, and to the FD for the AGI itself, instead of allocating a buffer on the heap every time the function is called. * Use the information contained within the ast_str to determine how many bytes need to be written instead of calling strlen(). * main/manager.c: Remove an XXX comment noting that it would be nice for a declaration to be inside of a function. (Yes, it would!) Replace it with a note that explains why it can't be done using the way that the AST_THREADSTORAGE macro is currently defined. * include/asterisk/agi.h, /, res/res_agi.c: Merged revisions 77788 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77788 | russell | 2007-07-30 14:13:31 -0500 (Mon, 30 Jul 2007) | 10 lines (closes issue #10279) Reported by: seanbright Patches: res_agi.carefulwrite.1.4.07252007.patch uploaded by seanbright (license 71) res_agi.carefulwrite.trunk.07252007.patch uploaded by seanbright (license 71) Allow the "agi_network: yes" line to be printed out in the AGI debug output. Also, allow partial writes to be handled when writing out this line just like it is for all of the others. ........ 2007-07-30 19:11 +0000 [r77787] Tilghman Lesher * include/asterisk/agi.h, res/res_agi.c: Cleanup of res_agi, ensuring thread safety (closes issue #10288) 2007-07-30 18:56 +0000 [r77786] Russell Bryant * main/channel.c, /: Merged revisions 77785 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77785 | russell | 2007-07-30 13:55:15 -0500 (Mon, 30 Jul 2007) | 3 lines file and I both committed changes for issue #10301. Remove a duplicated assignment to restore the original value of the previous channel. ........ 2007-07-30 18:45 +0000 [r77784] Tilghman Lesher * /, res/res_agi.c: Merged revisions 77783 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r77783 | tilghman | 2007-07-30 13:43:55 -0500 (Mon, 30 Jul 2007) | 10 lines Merged revisions 77782 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r77782 | tilghman | 2007-07-30 13:40:54 -0500 (Mon, 30 Jul 2007) | 2 lines Revert change in revision 71656, even though it fixed a bug, because many people were depending upon the (broken) behavior. ........ ................ 2007-07-30 17:31 +0000 [r77781] Russell Bryant * main/channel.c, /: Merged revisions 77780 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77780 | russell | 2007-07-30 12:29:43 -0500 (Mon, 30 Jul 2007) | 16 lines (closes issue #10301) Reported by: fnordian Patches: asterisk-1.4.9-channel.c.patch uploaded by fnordian (license 110) Additional changes by me Fix some problems in channel_find_locked() which can cause an infinite loop. The reference to the previous channel is set to NULL in some cases. These changes ensure that the reference to the previous channel gets restored before needing it again. I'm not convinced that the code that is setting it to NULL is really the right thing to do. However, I am making these changes to fix the obvious problem and just leaving an XXX comment that it needs a better explanation that what is there now. ........ 2007-07-30 17:12 +0000 [r77772-77779] Joshua Colp * /, res/res_features.c: Merged revisions 77778 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77778 | file | 2007-07-30 14:11:02 -0300 (Mon, 30 Jul 2007) | 4 lines (closes issue #10327) Reported by: kkiely Instead of directly mucking with the extension/context/priority of the channel we are transferring when it has a PBX simply call ast_async_goto on it. This will ensure that the channel gets handled properly and sent to the right place. ........ * apps/app_followme.c: Minor clean up of app_followme. * main/channel.c, /: Merged revisions 77771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77771 | file | 2007-07-30 12:47:52 -0300 (Mon, 30 Jul 2007) | 6 lines (closes issue #10301) Reported by: fnordian Patches: asterisk-1.4.9-channel.c.patch uploaded by fnordian (license 110) Restore previous behavior where if we failed to lock the channel we wanted we would return to exactly the same point as if we had just reentered the function. ........ 2007-07-30 15:22 +0000 [r77770] Russell Bryant * cdr/cdr_adaptive_odbc.c: Resolve some compiler warnings so that I can build under dev mode 2007-07-30 14:53 +0000 [r77769] Joshua Colp * /, apps/app_macro.c: Merged revisions 77768 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r77768 | file | 2007-07-30 11:51:44 -0300 (Mon, 30 Jul 2007) | 12 lines Merged revisions 77767 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r77767 | file | 2007-07-30 11:50:02 -0300 (Mon, 30 Jul 2007) | 4 lines (closes issue #10334) Reported by: ramonpeek Pass through the return value from macro_exec through the MacroIf application. ........ ................ 2007-07-30 10:55 +0000 [r77616-77766] Luigi Rizzo * channels/chan_sip.c: minor code rearrangements: + place the link field at the beginning of struct sip_pvt, and not somewhere in the middle; + in __sip_reliable_xmit, remove a duplicate assignment, and put the statements in a more logical order (i.e. first copy the payload and associated info, then copy arguments from the caller, then finish initializing the headers...) nothing to backport. * channels/chan_sip.c: rename handle_request() to handle_incoming(), as the former was misleading - the function deals with all incoming packets, be them requests or responses. * channels/chan_sip.c: move some dialog-only flags to proper variables, namely SIP_NOVIDEO, SIP_DIALOG_ANSWEREDELSEWHERE, SIP_PAGE2_NOTEXT, SIP_PAGE2_OUTGOING_CALL These are seldom used so the diff is relatively small. Note that 'OUTGOING_CALL' is dangerously similar to another dialog flag, 'SIP_OUTGOING', so the description will need to clarify the different meaning of the two. Also note that the description of NOTEXT is a bit unclear - does it mean we don't support it, or 'not requested or not supported' ? On passing fix a comment referring to video instead of text. Finally, mark with XXX a possibly misleading debugging message. (maybe the latter is worth backporting). * channels/chan_sip.c: use a function, cli_yesno(), to produce the output Yes or No for CLI lines. This helps maintaining consistency on output, slightly improves readability, and maybe one day will make it easier to translate the output in other languages (though i have a hard time believing that a CLI user who needs 'yes' and 'no' to be translated can actually figure out what he/she is doing!) * channels/chan_sip.c: move the two remaining peer flags to proper variables. * channels/chan_sip.c: move RT_FROMCONTACT to a proper sip_peer field. * channels/chan_sip.c: Move some global 'flags' to individual variables. Start putting these variables in a single struct (called 'sip_cfg' for the time being, but it could as well be 'global' or some other name) so it is easy, when reading the code, to figure out what they are for. The downside of using struct fields instead of individual global variables is that the compiler cannot tell if there are unused fields. But the advantage of not cluttering the namespace and manilpulating all these variables at once certainly overcome the disadvantagess. Nothing to backport, again. * channels/chan_sip.c: minor simplification of a conditional statement * channels/chan_sip.c: build the version of sip_tech with no send_digit_begin at load time instead of duplicating the initializer. This should remove the risk of forgetting fields in the initializer. * channels/chan_sip.c: remove bit position from description of SIP_* flags. use AST_FORMAT_AUDIO_MASK instead of playing with AST_FORMAT_MAX_AUDIO to determine audio formats. There is a dubious use of AST_FORMAT_MAX_AUDIO in sip_request_call() which surely needs fixing, namely: /* mask request with some set of allowed formats. * XXX this needs to be fixed. * The original code uses AST_FORMAT_AUDIO_MASK, but it is * unclear what to use here. We have global_capabilities, which is * configured from sip.conf, and sip_tech.capabilities, which is * hardwired to all audio formats. */ The latter is possibly something to backport when fixed. * channels/chan_sip.c: back on cleaning up the usage of flags. Move together flags used in the same way (e.g. dialog only, dialog-peer, ...) so it will become easier to deal with them in a more systematic way. This is being done in stages so it will be easier to detect breakage, if any should occur. * channels/chan_sip.c: more documentation on internal representation of incoming SIP messages. Remove definitions for now-unused flags, and add references to print routines for other flags. * channels/chan_sip.c: make register_unref() return NULL so it is easy to cleanup the original pointer while calling the function. on passing add some comments on one of the places where it is used, and explain why it is safe there. again, a no-op for practical purposes. * channels/chan_sip.c: add some documentation to auto_congest(), and some dialog_ref/unref (they are a no-op at the moment). Also clean a pointer after freeing memory to avoid dangling references, and write a for() loop in canonical form. In practice, everything in this commit is a no-op. * channels/chan_sip.c: more dialog_ref()/dialog_unref() calls * channels/chan_sip.c: more dialog_ref()/dialog_unref() calls * channels/chan_sip.c: start introducing hooks for reference counts on dialog descriptors. This commit is, for all practical purposes, a no-op, as it only introduces the dialog_ref() and dialog_unref() methods, and uses them in a few places (not all the places where they would be needed). The goal is to start annotating the code with these calls, so the transition to a proper container will be easier. Nothing to backport. * channels/chan_sip.c: remove an unused string * channels/chan_sip.c: simplify a conditional expression using S_OR * channels/chan_sip.c: make use of received= and rport= fields in sip replies. In a nutshell, these fields are used to tell a sip entity the address and port its request came from, and are extremely useful in the presence of NATs, especially with symmetric NATs where STUN is totally ineffective. This patch stores the address and port in the 'ourip' field of the dialog descriptor, so they can be reused in subsequent transactions. As it is, it works well for things like REGISTER requiring authentication, because the second REGISTER request (with auth credentials) will carry the correct address. Maybe it can also be useful, in case of an address change, to do one or both of the following: + propagate the new address to the parent user/peer descriptor so that new dialogs will use the correct address from the beginning. This is trivial to implement, I am just waiting for feedback on this. + re-issue a request in case of an address change. This a lot less trivial, maybe unnecessary, and probably covered by the previous item. I would seriously consider this patch for addition to 1.4 and 1.2. The code is very little intrusive, and it would solve in a correct way the nat traversal problems for which externip/externaddr/stunaddr are only a partial and expensive workaround. 2007-07-27 23:21 +0000 [r77572-77603] Tilghman Lesher * cdr/cdr_adaptive_odbc.c: Some ODBC drivers don't set the CHAR_OCTET_LENGTH field correctly. * Makefile: Target asterisk.pdf stopped building when the build was moved to the doc directory. * /, res/res_odbc.c: Merged revisions 77571 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77571 | tilghman | 2007-07-27 13:15:58 -0500 (Fri, 27 Jul 2007) | 2 lines Missing newline ........ 2007-07-27 17:05 +0000 [r77537-77541] Joshua Colp * /, cdr/cdr_pgsql.c: Merged revisions 77540 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77540 | file | 2007-07-27 14:04:08 -0300 (Fri, 27 Jul 2007) | 6 lines (closes issue #10310) Reported by: prashant_jois Patches: cdr_pgsql.patch uploaded by prashant (license 114) Finish the Postgresql connection after the log messages are printed so we don't access invalid memory. ........ * channels/chan_sip.c: Turn 4 lines of code into 1 line that does the same thing. * /, channels/chan_sip.c: Merged revisions 77536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77536 | file | 2007-07-27 13:27:16 -0300 (Fri, 27 Jul 2007) | 6 lines (closes issue #10323) Reported by: julianjm Patches: chan_sip_device_state_hold_fix.v1.diff.txt uploaded by julianjm (license 99) Clear ONHOLD flag when decrementing the onHold peer count. If we did not do this the count may keep decreasing. ........ 2007-07-27 16:20 +0000 [r77534] Tilghman Lesher * pbx/pbx_config.c: 'dialplan save' shouldn't be converting '|' back to ',' anymore. 2007-07-27 15:46 +0000 [r77520] Steve Murphy * apps/app_dial.c, pbx/pbx_ael.c: These fixes take care of two problems: a complaint in asterisk-dev that goto's aren't working in trunk, a side effect of the move to commas as arg seps in apps and funcs; and a problem I spotted myself with dial's 'e' option, where gotos were off by one, because I forgot to set the AUTOLOOP flag in the peer channel. 2007-07-27 14:31 +0000 [r77491] Mark Michelson * /, channels/chan_sip.c: Merged revisions 77490 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77490 | mmichelson | 2007-07-27 09:30:43 -0500 (Fri, 27 Jul 2007) | 3 lines "re-invite" was misspelled ........ 2007-07-26 23:20 +0000 [r77461] Joshua Colp * main/channel.c, /: Merged revisions 77460 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77460 | file | 2007-07-26 20:19:04 -0300 (Thu, 26 Jul 2007) | 4 lines (closes issue #10302) Reported by: litnialex If a DTMF end frame comes from a channel without a begin and it is going to a technology that only accepts end frames (aka INFO) then use the minimum DTMF duration if one is not in the frame already. ........ 2007-07-26 22:17 +0000 [r77432] Kevin P. Fleming * /, doc/tex/mp3.tex, sounds/Makefile: Merged revisions 77424,77429 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77424 | kpfleming | 2007-07-26 17:14:21 -0500 (Thu, 26 Jul 2007) | 2 lines use new canonical name for download server ........ r77429 | kpfleming | 2007-07-26 17:16:42 -0500 (Thu, 26 Jul 2007) | 2 lines change protocol for downloads as well ........ 2007-07-26 21:24 +0000 [r77411] Russell Bryant * Makefile, /: Merged revisions 77410 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77410 | russell | 2007-07-26 16:23:23 -0500 (Thu, 26 Jul 2007) | 10 lines AST_DEVMODE was defined in trunk, but not in 1.4. When Asterisk is compiled under dev mode, AST_DEVMODE will get defined in buildopts.h. Change 1.4 to define it in the same way that trunk does. Also, revert the change that added this define in the Makefile The advantage to doing it this way is that buildopts.h gets installed when you install Asterisk. Then, when building any out of tree modules, or building asterisk-addons, these modules know which options the rest of Asterisk was built with. ........ 2007-07-26 20:39 +0000 [r77381] Mark Michelson * Makefile, /, main/logger.c: Merged revisions 77380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77380 | mmichelson | 2007-07-26 15:35:17 -0500 (Thu, 26 Jul 2007) | 7 lines Fixes to get ast_backtrace working properly. The AST_DEVMODE macro was never defined so the majority of ast_backtrace never attempted compilation. The makefile now defines AST_DEVMODE if configure was run with --enable-dev-mode. Also, changes were made to acccomodate 64 bit systems in ast_backtrace. Thanks to qwell, kpfleming, and Corydon76 for their roles in allowing me to get this committed ........ 2007-07-26 19:33 +0000 [r77349-77351] Tilghman Lesher * /, main/logger.c: Merged revisions 77350 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77350 | tilghman | 2007-07-26 14:32:17 -0500 (Thu, 26 Jul 2007) | 2 lines Missed one ........ * /, main/logger.c: Merged revisions 77348 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77348 | tilghman | 2007-07-26 14:27:18 -0500 (Thu, 26 Jul 2007) | 2 lines Oops, that builtin define should be all-lowercase. ........ 2007-07-26 18:31 +0000 [r77319] Mark Michelson * /, cdr/cdr_pgsql.c: Merged revisions 77318 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77318 | mmichelson | 2007-07-26 13:30:29 -0500 (Thu, 26 Jul 2007) | 8 lines Two consecutive calls to PQfinish could occur, meaning free gets called on the same variable twice. This patch sets the connection to NULL after calls to PQfinish so that the problem does not occur. Also in this patch, prashant_jois informed me that it is safe to pass a null pointer to PQfinish, so I have removed the check for conn's existence from my_unload_module. (closes issue 10295, reported by junky, patched by me with input from prashant_jois) ........ 2007-07-26 15:49 +0000 [r77268-77299] Russell Bryant * main/udptl.c, res/res_features.c, main/say.c, codecs/codec_adpcm.c, apps/app_alarmreceiver.c, cdr/cdr_adaptive_odbc.c, channels/chan_iax2.c, main/indications.c, main/config.c, main/loader.c, res/res_smdi.c, pbx/pbx_spool.c, channels/chan_skinny.c, apps/app_zapscan.c, apps/app_zapras.c, pbx/pbx_realtime.c, channels/chan_alsa.c, apps/app_amd.c, cdr/cdr_odbc.c, res/res_speech.c, apps/app_dial.c, codecs/codec_g722.c, funcs/func_timeout.c, codecs/codec_speex.c, channels/chan_agent.c, codecs/codec_g726.c, channels/iax2-provision.c, apps/app_db.c, channels/chan_misdn.c, main/srv.c, apps/app_waitforring.c, apps/app_macro.c, apps/app_chanspy.c, apps/app_voicemail.c, channels/chan_vpb.cc, apps/app_meetme.c, res/res_snmp.c, codecs/codec_gsm.c, res/res_musiconhold.c, apps/app_followme.c, codecs/codec_zap.c, res/res_jabber.c, main/channel.c, main/cdr.c, channels/chan_phone.c, main/dial.c, res/res_config_odbc.c, main/manager.c, funcs/func_odbc.c, res/res_agi.c, main/app.c, main/image.c, apps/app_rpt.c, apps/app_parkandannounce.c, channels/chan_mgcp.c, apps/app_adsiprog.c, apps/app_while.c, codecs/codec_lpc10.c, res/res_config_pgsql.c, main/dnsmgr.c, channels/chan_zap.c, apps/app_read.c, channels/chan_sip.c, main/translate.c, codecs/codec_alaw.c, apps/app_waitforsilence.c, res/res_crypto.c, apps/app_queue.c, apps/app_getcpeid.c, channels/chan_oss.c, main/rtp.c, apps/app_flash.c, main/abstract_jb.c, main/file.c, channels/chan_h323.c, codecs/codec_ulaw.c, pbx/pbx_dundi.c, apps/app_sms.c, pbx/pbx_gtkconsole.c: Do a massive conversion for using the ast_verb() macro (closes issue #10277, patches by mvanbaak) Basically, this changes ... if (option_verbose > 2) ast_verbose(VERBOSE_PREFIX_3, "Something\n"); to ... ast_verb(3, "Something\n"); * doc/tex/odbcstorage.tex, doc/tex/hardware.tex, doc/tex/mp3.tex, doc/tex/channelvariables.tex, doc/tex/qos.tex, doc/tex/queues-with-callback-members.tex, doc/tex/realtime.tex, doc/tex/dundi.tex, doc/tex/enum.tex, doc/tex/asterisk-conf.tex, doc/tex/manager.tex, doc/tex/misdn.tex, doc/tex/imapstorage.tex, doc/tex/privacy.tex, LICENSE, doc/tex/app-sms.tex, doc/tex/cdrdriver.tex, doc/tex/asterisk.tex: Merge a big batch of documentation fixes for escaping, marking URLs, places where verbatim text went off the end of the page on the PDF, and various other improvements (closes issue #10307, IgorG) * channels/chan_sip.c: Revert some changes to call abs() on the result of ast_random(). * random() is defined to return a positive result, and now ast_random() will always do so as well * main/utils.c: Ensure that the read from /dev/urandom returns a positive result (closes issue #10308, reported by yehavi, patched by me) 2007-07-26 13:19 +0000 [r77267] Tilghman Lesher * channels/chan_sip.c: Things expecting a positive result from ast_random() should not be surprised (closes #10308) 2007-07-26 13:10 +0000 [r77266] Russell Bryant * main/rtp.c: Add a link to the list of assigned RTP payload types for convenience. 2007-07-26 05:35 +0000 [r77233-77248] Luigi Rizzo * main/rtp.c: document how the RTP marker bit is passed for video frames, and why this does not overwrite useful information. * main/rtp.c: add an entry for h263plus in an empty slot of the rtp types. 2007-07-26 01:33 +0000 [r77217-77218] Steve Murphy * /, pbx/pbx_ael.c: The upgrade of application argument separators to comma has an effect on AEL; I commented out the code that substitutes commas with vertbars, so we can get apps to parse their args correctly. * apps/app_meetme.c: Merged revisions 77191 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77191 | murf | 2007-07-25 16:39:27 -0600 (Wed, 25 Jul 2007) | 1 line This fix solves problem with intense squelch noise when someone joins conf in bug 9430; We repro'd the problem with meetme opts of 'CciMo'; Josh Colp supplied this patch, and I'm applying it. It looks like playing the recorded username will louse up the next thing played into the channel. Josh rearranged the code so as to start things over before playing data directly into the conference. ........ 2007-07-25 22:18 +0000 [r77182] Joshua Colp * /, apps/app_speech_utils.c: Merged revisions 77176 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77176 | file | 2007-07-25 19:16:10 -0300 (Wed, 25 Jul 2007) | 4 lines (closes issue #10303) Reported by: jtodd Add SPEECH_DTMF_TERMINATOR variable so the user can specify the digit to terminate a DTMF string with. If none is specified then no terminator will be used. ........ 2007-07-25 21:58 +0000 [r77156] Luigi Rizzo * channels/chan_iax2.c: silence a warning in ast-devmode on a potentially uninitialized var. At first sight (but the function is very large so i am not 100% sure) the code seems correct, so maybe my compiler is just not smart enough to figure that out at the optimization level it has. Not worthwhile merging to 1.4 i believe. 2007-07-25 21:53 +0000 [r77155] Mark Michelson * main/channel.c, /: Merged revisions 77154 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77154 | mmichelson | 2007-07-25 16:52:47 -0500 (Wed, 25 Jul 2007) | 3 lines chan->emulate_dtmf_duration is an unsigned int, not a signed int, so use %u instead of %d in the format string ........ 2007-07-25 17:16 +0000 [r77072] Joshua Colp * /, configure, acinclude.m4: Merged revisions 77071 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77071 | file | 2007-07-25 14:14:14 -0300 (Wed, 25 Jul 2007) | 2 lines Fix autoconf logic for finding OpenH323 when it is not in the first place searched (/usr/share/openh323). ........ 2007-07-25 14:13 +0000 [r77023-77054] Luigi Rizzo * main/translate.c: change the debug level to 3 for an exceedingly annoying message (3-deep nested loop) * main/rtp.c: Merged revisions 77022 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77022 | rizzo | 2007-07-25 11:34:01 +0200 (Wed, 25 Jul 2007) | 3 lines set the sequence number in a frame for all frame types ........ 2007-07-25 01:06 +0000 [r76985] Russell Bryant * CHANGES: remove a couple of entries that got duplicated and snuck into the SIP section. Also, align the NAT/STUN entry with the others. 2007-07-25 00:34 +0000 [r76984] Steve Murphy * channels/chan_zap.c, /: Merged revisions 76983 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r76983 | murf | 2007-07-24 18:18:32 -0600 (Tue, 24 Jul 2007) | 9 lines Merged revisions 76978 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r76978 | murf | 2007-07-24 18:07:24 -0600 (Tue, 24 Jul 2007) | 1 line this fixes bug 10293, where the error message because defaultzone or loadzone was not defined was confusing ........ ................ 2007-07-24 22:13 +0000 [r76874-76940] Tilghman Lesher * /, include/asterisk/lock.h: Merged revisions 76937 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r76937 | tilghman | 2007-07-24 17:12:43 -0500 (Tue, 24 Jul 2007) | 10 lines Merged revisions 76934 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r76934 | tilghman | 2007-07-24 17:11:33 -0500 (Tue, 24 Jul 2007) | 2 lines Oops, res contains the error code, not errno. I was wondering why a mutex was reporting "No such file or directory"... ........ ................ * build_tools/cflags.xml: Add the flag to trigger an intentional crash on mutex errors * doc/tex/manager.tex, doc/tex/misdn.tex, doc/tex/jitterbuffer.tex, doc/tex/odbcstorage.tex, doc/tex/hardware.tex, doc/tex/privacy.tex, doc/tex/billing.tex, doc/tex/ael.tex, doc/tex/channelvariables.tex, doc/tex/qos.tex, doc/tex/realtime.tex, doc/tex/asterisk.tex, doc/tex/queuelog.tex: Fix escaping and some of the formattting (closes issue #10285) 2007-07-24 17:43 +0000 [r76841-76852] Jason Parker * channels/chan_skinny.c: Revert trivial whitespace change (for testing) * channels/chan_skinny.c: Trivial whitespace change to test comitting... 2007-07-24 17:05 +0000 [r76807] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 76803 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r76803 | qwell | 2007-07-24 11:32:20 -0500 (Tue, 24 Jul 2007) | 3 lines Don't create the Asterisk channel until we are starting the PBX on it. (ASA-2007-018) ........ 2007-07-24 16:42 +0000 [r76804] Mark Michelson * /, apps/app_queue.c: Merged revisions 76801 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r76801 | mmichelson | 2007-07-24 11:26:58 -0500 (Tue, 24 Jul 2007) | 13 lines Added a membercount variable to call_queue struct which keeps track of the number of logged in members in a particular queue. This makes it so that the 'n' option for Queue() can act properly depending on which strategy is used. If the strategy is roundrobin, rrmemory, or ringall, we want to ring each phone once before moving on in the dialplan. However, if any other strategy is used, we will only ring one phone since it cannot be guaranteed that a different phone will ring on subsequent attempts to ring a phone. As a side effect of this, the QUEUE_MEMBER_COUNT dialplan function now just reads the membercount variable instead of traversing through the member list to figure out how many members there are. Special thanks to blitzrage for helping to test this out. (closes issue #10127, reported by bcnit, patched by me, tested by blitzrage) ........ 2007-07-24 16:09 +0000 [r76791] Joshua Colp * sounds/Makefile: Don't download/install the sound packages if already installed. 2007-07-24 15:35 +0000 [r76785] Jason Parker * channels/chan_skinny.c: The chan_skinny Dial() syntax was funky. You had to do Dial(Skinny/line@device) This allows you to just Dial(Skinny/line), as long as line isn't ambiguous. Note that this does not remove or deprecate the "old" syntax, as it's still quite useful - even moreso if shared lines get implemented. Initial patch by me, with some changes and suggestions from wedhorn. (closes issue #10263) 2007-07-24 14:49 +0000 [r76755-76770] Luigi Rizzo * channels/chan_sip.c: two small fixes when using stun (reported by Marta Carbone): + externexpire was not initialized properly; + stunaddr was not handled properly on a sip reload * CHANGES: add documentation on nat/stun support in chan_sip 2007-07-24 02:59 +0000 [r76710-76712] Joshua Colp * main/manager.c: Move manager users list over to an rwlist. * res/res_agi.c: You need to put static in front of a static RWLIST declaration to make it really static... and don't call AST_RWLIST_HEAD_DESTROY on a statically declared list. * main/manager.c: Don't bother calling AST_RWLIST_EMPTY on a list before AST_RWLIST_TRAVERSE, it's just a double check. 2007-07-23 22:41 +0000 [r76707-76709] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 76708 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r76708 | tilghman | 2007-07-23 17:38:06 -0500 (Mon, 23 Jul 2007) | 4 lines It was our stated intention for 1.4 that files created in app_voicemail should depend upon the umask. Unfortunately, mkstemp() creates files with mode 0600, regardless of the umask. This corrects that deficiency. ........ * include/asterisk/agi.h, res/res_agi.c: Enhance AGI with several fixes: - Makes the structures handling external AGI commands a bit more thread-safe - Makes AGI transparently work with both live and hungup channels - DeadAGI is hence no longer necessary and is deprecated - CLI bug fixes - Commands will refuse to run if the channel is dead and the command is nonsensical for dead channels. 2007-07-23 21:42 +0000 [r76706] Joshua Colp * res/res_crypto.c: Clean up res_crypto module. It now uses an rwlist to keep the keys and it should also be thread safe now. 2007-07-23 20:27 +0000 [r76703-76704] Tilghman Lesher * res/res_agi.c, UPGRADE.txt: Missed one conversion to comma delimiter (thanks, Juggie) and add documentation on the change to the Local channel name. * funcs/func_rand.c, apps/app_readfile.c, channels/chan_local.c, apps/app_record.c, funcs/func_env.c, funcs/func_strings.c, funcs/func_vmcount.c, include/asterisk/aes.h, funcs/func_logic.c, apps/app_exec.c, apps/app_controlplayback.c, funcs/func_odbc.c, apps/app_skel.c, apps/app_zapras.c, apps/app_url.c, apps/app_externalivr.c, apps/app_parkandannounce.c, apps/app_dial.c, main/pbx.c, apps/app_page.c, apps/app_softhangup.c, UPGRADE.txt, funcs/func_cut.c, apps/app_talkdetect.c, apps/app_queue.c, funcs/func_realtime.c, include/asterisk/app.h, apps/app_channelredirect.c, apps/app_macro.c, pbx/pbx_config.c, apps/app_verbose.c, apps/app_chanspy.c, funcs/func_callerid.c, apps/app_voicemail.c: Merge the dialplan_aesthetics branch. Most of this patch simply converts applications using old methods of parsing arguments to using the standard macros. However, the big change is that the really old way of specifying application and arguments separated by a comma will no longer work (e.g. NoOp,foo|bar). Instead, the way that has been recommended since long before 1.0 will become the only method available (e.g. NoOp(foo,bar). 2007-07-23 19:00 +0000 [r76657] Jason Parker * /, channels/chan_skinny.c: Merged revisions 76656 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r76656 | qwell | 2007-07-23 13:59:28 -0500 (Mon, 23 Jul 2007) | 3 lines Fix some incorrect softkey labels in messages. Don't try to play dialtone in some unimplemented features. ........ 2007-07-23 18:31 +0000 [r76655] Joshua Colp * /, channels/chan_agent.c: Merged revisions 76654 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r76654 | file | 2007-07-23 15:29:48 -0300 (Mon, 23 Jul 2007) | 12 lines Merged revisions 76653 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r76653 | file | 2007-07-23 15:28:13 -0300 (Mon, 23 Jul 2007) | 4 lines (closes issue #5866) Reported by: tyler Do not force channel format changes when a generator is present. The generator may have changed the formats itself and changing them back would cause issues. ........ ................ 2007-07-23 17:58 +0000 [r76621] Jason Parker * /, channels/chan_skinny.c: Merged revisions 76620 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #10276) ........ r76620 | qwell | 2007-07-23 12:57:53 -0500 (Mon, 23 Jul 2007) | 4 lines Don't try to queue up hold/unhold frames on a non-existent channel. Issue 10276. ........ 2007-07-23 17:49 +0000 [r76619] Joshua Colp * /, apps/app_morsecode.c: Merged revisions 76618 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r76618 | file | 2007-07-23 14:48:51 -0300 (Mon, 23 Jul 2007) | 2 lines Allow app_morsecode to build on PPC Linux by putting the value of the digit char in an int. ........ 2007-07-23 14:45 +0000 [r76564] Luigi Rizzo * channels/chan_sip.c: add two missing entries in the replica of the sip_tech that does not use DTMF BEGIN frames. 1.4 seems correct (it does not have the two fields). However, as this bug shows, the current way of creating the sip_tech replica is too error-prone, one can easily forget to update one of the two entries. Perhaps it would be better to create sip_tech_info expliclty at module load, by doing sip_tech_info = sip_tech; sip_tech_info.send_digit_begin = NULL (in this case, this is something applicable to 1.4 as well). 2007-07-23 14:38 +0000 [r76563] Joshua Colp * /, channels/chan_sip.c: Merged revisions 76561 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r76561 | file | 2007-07-23 11:34:21 -0300 (Mon, 23 Jul 2007) | 14 lines Merged revisions 76560 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r76560 | file | 2007-07-23 11:32:07 -0300 (Mon, 23 Jul 2007) | 6 lines (closes issue #10236) Reported by: homesick Patches: rpid_1.4_75840.patch uploaded by homesick (license 91) Accept Remote Party ID on guest calls. ........ ................ 2007-07-23 14:37 +0000 [r76555-76562] Russell Bryant * channels/chan_sip.c: Mark str2dtmfmode() as currently unused to resolve a compiler warning and allow building under dev mode * include/asterisk.h, res/res_snmp.c, channels/chan_sip.c, res/res_crypto.c, res/res_convert.c, main/devicestate.c, include/jitterbuf.h, res/res_config_sqlite.c, main/enum.c, res/res_monitor.c, include/asterisk/file.h, include/asterisk/doxyref.h, res/res_config_odbc.c, res/res_indications.c, main/asterisk.c, res/res_clioriginate.c: (closes issue #10271) Reported by: snuffy Patches: doxygen-updates.diff uploaded by snuffy (license 35) Another big batch of doxygen documentation updates * CHANGES: note the debug and verbose changes in CHANGES * include/asterisk/logger.h, main/pbx.c, main/logger.c, include/asterisk/options.h, main/asterisk.c, main/cli.c: (closes issue #10192) Reported by: bbryant Patches: 20070720__core_debug_by_file.patch uploaded by bbryant (license 36) (with some modifications by me) Tested by: russell, bbryant This set of changes introduces the ability to set the core debug or verbose levels on a per-file basis. Interestingly enough, in 1.4, you have the ability to set core debug for a single file, but that functionality was accidentally lost in the conversion of the CLI commands to the new format. This patch improves upon what was in 1.4 by letting you set it for more than 1 file, and by also supporting verbose. *** Janitor Project *** This patch also introduces a new macro, ast_verb(), which is similar to ast_debug(). Setting the per file verbose value only works for messages that use this macro. Converting existing uses of ast_verbose() can be done like: if (option_debug > 2) ast_verbose(VERBOSE_PREFIX_3 "Something useful\n"); ... ast_verb(3, "Something useful\n"); 2007-07-23 14:18 +0000 [r76547] Luigi Rizzo * channels/chan_sip.c: introduce two functions, map_x_s() and map_s_x(), to map between integers and strings using a single translation table, and use them in a few places instead of ad-hoc routines that duplicate the table. On passing, note that REFER_CONFIRMED is never used, and add a few comments. Nothing to backport here. 2007-07-23 14:02 +0000 [r76524] Russell Bryant * channels/chan_sip.c: Remove an unused function to resolve a compiler warning 2007-07-23 13:46 +0000 [r76523] Joshua Colp * channels/chan_skinny.c, configure, include/asterisk/autoconfig.h.in, configure.ac: Use autoconf logic to determine byte swapping macro presence. This should now also use other macros if present. 2007-07-23 13:29 +0000 [r76521] Luigi Rizzo * channels/chan_sip.c: move "sip prunte realtime ..." and "sip set debug ... " to NEW_CLI style. 2007-07-23 13:24 +0000 [r76520] Joshua Colp * /, channels/chan_skinny.c: Merged revisions 76519 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r76519 | file | 2007-07-23 10:23:09 -0300 (Mon, 23 Jul 2007) | 6 lines (closes issue #10268) Reported by: mvanbaak Patches: chan_skinny_openbsd.diff uploaded by mvanbaak (license 7) Add another OS that has to use the Macros for byte ordering. ........ 2007-07-23 12:29 +0000 [r76486] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 76485 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r76485 | russell | 2007-07-23 07:25:01 -0500 (Mon, 23 Jul 2007) | 6 lines Use a signed integer for storing the number of bytes in the packet read from the network. Using an unsigned value here made it impossible to handle an error returned from recvfrom(). Furthermore, in the case that recvfrom() did return an error, this would cause a crash due to a heap overflow. (closes issue #10265, reported by and fix suggested by timrobbins) ........ 2007-07-23 03:10 +0000 [r76313-76467] Luigi Rizzo * channels/chan_sip.c: Add some documentation on the sipregistry states and the handling of the sip_register structures. This commit only changes comments and whitespace. * channels/chan_sip.c: add a bit of comments on internal functions. * channels/chan_sip.c: rewrite "sip show {channels|subscriptions}" CLI handler using the new-style cli format. No functional changes, nothing to backport. * channels/chan_sip.c: Make sip_destroy() return NULL so the caller can do things like foo = sip_destroy(foo); and reduce the chance of bugs due to dangling pointers. Also remove a duplicate prototype for the function. nothing to backport. * channels/chan_sip.c: add two comment blocks, one on reusing nonces, and one on the handling of an 'authpeer' local variable. * channels/chan_sip.c: comment and slightly restructure handle_request() in the part that handles responses, so that there is a common exit point. Mark two places where probably we could return -1 instead of 0 to report an error to the caller. (change triggered by investigations on how the 'SIP_PKT_IGNORE' field was used). nothing to backport from this commit * channels/chan_sip.c: remove unused argument from handle_invite_replaces(), and also leftover SIP_PKT_* stuff from the previous commit. * channels/chan_sip.c: Cleanup of flags used in struct sip_request, moving them to individual variables. Apart from SIP_PKT_IGNORE which was used a zillion times, the other two are used seldom. On passing: - move the arrays to the end of struct sip_request, so a (small) buffer overflow is less likely to overwrite the other fields; - note that the 'ignore' argument to handle_invite_replaces() is not used and should be removed (will be done in a separate commit). Nothing to backport in this change. * channels/chan_sip.c: move two per-packet flags to proper variables. * channels/chan_sip.c: minor clarification on the usage of SIP_* flags. Also correct some items that were misclassified. * channels/chan_sip.c: document the way sipdebug works, and implement it through variables and not flags. NOTE: The old behaviour (preserved in this commit) is that if sipdebug is set in the config file, it can only be disabled by reloading the config. I am not sure if this is accidental or voluntary, but it is really unconvenient and I think it should be handled in the same way as other options i.e. consider requests from the config file or the cli (or the command line) to be fully equivalent and act on the same status variable. * channels/chan_sip.c: move the SIP_REALTIME flag to a field in the user/peer structure. * channels/chan_sip.c: Add a note to document how the temporary 'pvt' should be initialized before using it. I am unclear on the details right now so i hope someone can comment more. The obvious (and lazy) approach would be to bzero() all of it (except for the string pool), but isn't that too much work ? Feedback wanted here... 2007-07-21 14:39 +0000 [r76296] Joshua Colp * include/asterisk/utils.h, configure, include/asterisk/autoconfig.h.in, configure.ac, main/utils.c: Add support for using /dev/urandom to get random numbers on systems that support it. 2007-07-21 09:35 +0000 [r76229-76279] Luigi Rizzo * channels/chan_sip.c: whoops... was setting needdestroy on the wrong dialog. (spotted by a diff with my own branch) * channels/chan_sip.c: more two more flags to proper variables: ALREADYGONE and NEEDDESTROY. * channels/chan_sip.c: use explicit variables for things that don't need to be stored in ast_flags. First victim is 'SIP_NO_HISTORY' replaced by a 'do_history' field in the sip_pvt structure. * channels/chan_sip.c: Use ast_str_append() instead of ast_build_string() to construct the sdp messages. Overall the code is slightly more readable (because the string is fully described by a single pointer), and more efficient (because the length is stored explicitly so you don't need to do strlen()). (I have been using this code for almost a year now.) I wish we had infix string operators to do this sort of things! Nothing to backport from this change. 2007-07-21 01:25 +0000 [r76224] Luigi Rizzo * channels/chan_sip.c: We have two 'technology' descriptors for a SIP channel, so define and use a macro to determine whether we are pointing to one of them, so when one goes away (or a new one appears) we don't have to touch all the code. 2007-07-21 01:08 +0000 [r76222] Steve Murphy * apps/app_queue.c: One small documentation update made to accompany 10154, the upgrading of the queue ringing to allow periodic announcments 2007-07-21 01:01 +0000 [r76221] Luigi Rizzo * channels/chan_sip.c, configs/sip.conf.sample: Enhance NAT support as discussed on the -dev list, i.e.: + extensive documentation changes both in sip.conf.sample and in the source; + allow "externip" and "externhost" to include a port number as well; + allow "bindaddr" to have a port number (making bindport unnecessary, even though it is still present for backward compatibility); + introduce the new "stunaddr" parameter to specify an STUN server to be used from the main SIP socket; + extend the "sip show settings" output to show all the above. Internally: + change related data structures from struct in_addr to struct sockaddr_in to store the port numbers as well; + reorganize ast_sip_ouraddrfor() (should also be renamed to sip_ouraddrfor() because it is not a generic API, though it might become so if called with a socket as an additional argument, in which case it can be moved elsewhere). As mentioned in the documentation, media sessions still do not use STUN so the port numbers may still be incorrect when Asterisk is behind a NAT On passing, some of the debugging messages printing media addresses are probably using the wrong values, but this will be checked/fixed in a subsequent commit if needed. Part of the following chunk in the function that handles a "sip reload" is probably needed on previous versions as well, to avoid leaking the memory used for the "localaddr" list: @@ -17244,13 +17274,17 @@ /* Reset IP addresses */ memset(&bindaddr, 0, sizeof(bindaddr)); + memset(&stunaddr, 0, sizeof(stunaddr)); + memset(&internip, 0, sizeof(internip)); + /* Free memory for local network address mask */ + ---> ast_free_ha(localaddr); <----- memset(&localaddr, 0, sizeof(localaddr)); memset(&externip, 0, sizeof(externip)); memset(&default_prefs, 0 , sizeof(default_prefs)); 2007-07-21 00:57 +0000 [r76220] Steve Murphy * apps/app_queue.c: This update was supplied in 10154; to allow announcemnts if the 'r' option (ringing) is provided. 2007-07-20 22:25 +0000 [r76216] Jason Parker * configs/say.conf.sample, apps/app_playback.c: Add support for default "say mode" (whether to use the "old" method or "new" method. "new" method being config file) Add support for autocomplete of "say load" CLI command. Patch by IgorG (closes issue #10243) 2007-07-20 21:41 +0000 [r76213] Steve Murphy * /, sounds/Makefile: Merged revisions 76211 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r76211 | murf | 2007-07-20 15:36:05 -0600 (Fri, 20 Jul 2007) | 1 line This patch from 10249 is worth applying! It prevents downloading sound files if they are already downloaded. Darn Practical, if you ask me ........ 2007-07-20 21:04 +0000 [r76175-76179] Jason Parker * /, channels/chan_skinny.c: Merged revisions 76174 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r76174 | qwell | 2007-07-20 15:32:55 -0500 (Fri, 20 Jul 2007) | 2 lines It's possible for sub->owner to be NULL here if you cancel the call immediately after/during sending a digit. ........ 2007-07-20 18:44 +0000 [r76140] Mark Michelson * /, apps/app_directory.c: Merged revisions 76139 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r76139 | mmichelson | 2007-07-20 13:42:27 -0500 (Fri, 20 Jul 2007) | 6 lines When using users.conf for the entries in the directory, if multiple users had the same last name, only the first user listed would be available in the directory. (closes issue #10200, reported by mrskippy, patched by me) ........ 2007-07-20 18:28 +0000 [r76138] Russell Bryant * main/channel.c, /: Merged revisions 76132 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r76132 | russell | 2007-07-20 13:22:24 -0500 (Fri, 20 Jul 2007) | 6 lines Use the define that specifies the default length of an artificially created DTMF digit in the ast_senddigit() function. The define is set to 100ms by default, which is the same thing that this function was using. But, using the define lets changes take effect in this case, as well as the others where it was already used. ........ 2007-07-20 17:21 +0000 [r76055-76091] Joshua Colp * /, channels/chan_sip.c: Merged revisions 76087 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r76087 | file | 2007-07-20 14:20:09 -0300 (Fri, 20 Jul 2007) | 14 lines Merged revisions 76080 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r76080 | file | 2007-07-20 14:16:48 -0300 (Fri, 20 Jul 2007) | 6 lines (closes issue #10247) Reported by: fkasumovic Patches: chan_sip.patch uploaded by fkasumovic (license #101) Drop any peer realm authentication entries when reloading so multiple entries do not get added to the peer. ........ ................ * /, res/res_convert.c: Merged revisions 76067 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r76067 | file | 2007-07-20 14:10:17 -0300 (Fri, 20 Jul 2007) | 6 lines (closes issue #10246) Reported by: fkasumovic Patches: res_conver.patch uploaded by fkasumovic (license #101) Use the last occurance of . to find the extension, not the first occurance. ........ * channels/chan_sip.c: It is impossible for the externhost variable to not exist, it is however possible for it to be empty. 2007-07-20 15:06 +0000 [r76034-76037] Luigi Rizzo * channels/chan_sip.c: Don't use a field size for the last argument of printf format, because in this case the string is left-aligned and it is not truncated anyways. Omitting the field size prevents the generation of trailing whitespace, which makes the string fit in smaller windows. * channels/chan_sip.c: Extend the 'network settings' section with indication on the localnet settings (requires the change in SVN 76034), and also give an indication on whether/why/how the remapping of addresses in SIP message is done or not. I think this is especially useful for debugging the configuration, as the address remapping depends on a combination of at least 3 parameters (localnet, externhost, externip) and successful DNS lookup. An example of the output of this section is below: Network Settings: --------------------------- SIP address remapping: Enabled using externhost Externhost: foo.dyndns.net Externip: 80.64.128.23:0 Externrefresh: 10 Internal IP: 12.34.56.78:5060 Localnet: 192.168.0.0/255.255.0.0 10.0.0.0/255.0.0.0 I leave to the community the judgement if the above info is a useful addition for 1.4. It is not a bugfix, but it is neither a new feature, only a useful diagnostic tool. Note that I would like to move there also the bindaddress/port information, in the usual addr:port format e.g. Bindaddress: 0.0.0.0:5060 so that network information is all in one place. * include/asterisk/acl.h, main/acl.c: expose struct ast_ha so external code can do things such as printing it (e.g. chan_sip.c in a subsequent commit). Obviously exposing the internals of a data structure is far from ideal (especially in a case like this where the implementation is very inefficient and will need to be changed at some point). On the other hand, it was also unclear what additional APIs should we provide instead, and because exposing the stucture has no impact on source and binary compatibility, this seemed to me the best option at this time. 2007-07-20 01:54 +0000 [r76015] Tilghman Lesher * main/logger.c: Reduce some logging contention by switching several locks over to rwlocks 2007-07-19 23:24 +0000 [r75982-75983] Steve Murphy * apps/app_dial.c, include/asterisk/utils.h, channels/chan_local.c, channels/chan_sip.c, include/asterisk/dundi.h, res/res_features.c, include/asterisk/chanspy.h, include/asterisk/speech.h, channels/iax2-provision.c, include/asterisk/cdr.h, include/asterisk/channel.h, res/res_musiconhold.c, channels/chan_iax2.c, main/rtp.c, channels/iax2-provision.h, main/loader.c, include/asterisk/abstract_jb.h, include/asterisk/features.h, main/channel.c, include/asterisk/app.h, funcs/func_odbc.c, include/asterisk/module.h, include/asterisk/jabber.h, apps/app_minivm.c, main/app.c, pbx/pbx_dundi.c, apps/app_mixmonitor.c, apps/app_voicemail.c: After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort. * apps/app_queue.c: This repairs a 'warning: ISO C90 forbids mixed declarations and code' message that cripples my dev-mode enabled build 2007-07-19 19:02 +0000 [r75977-75979] Mark Michelson * /, apps/app_queue.c: Merged revisions 75978 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r75978 | mmichelson | 2007-07-19 13:59:30 -0500 (Thu, 19 Jul 2007) | 3 lines The diff on this looks pretty big but all I did was remove a pointless if statement (always evaluates true). ........ * /, apps/app_queue.c: Merged revisions 75969 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r75969 | mmichelson | 2007-07-19 11:26:10 -0500 (Thu, 19 Jul 2007) | 10 lines Changes in handling return values of several functions in app_queue. This all started as a fix for issue #10008 but now includes all of the following changes: 1. Simplifying the code to handle positive return values from ast API calls. 2. Removing the background_file function. 3. The fix for issue #10008 (closes issue #10008, reported and patched by dimas) ........ 2007-07-19 15:59 +0000 [r75911-75930] Russell Bryant * res/res_agi.c: (closes issue #10210, reported and patched by juggie) This merges the trunk only part of the patches from this issue. In 1.4, res_agi will issue a warning if you try to use DeadAGI on a channel that is not hung up. Now, in trunk, it just plain won't let you do it. * /, channels/chan_iax2.c: Merged revisions 75928 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r75928 | russell | 2007-07-19 10:53:15 -0500 (Thu, 19 Jul 2007) | 14 lines Merged revisions 75927 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r75927 | russell | 2007-07-19 10:49:42 -0500 (Thu, 19 Jul 2007) | 6 lines When processing full frames, take sequence number wraparound into account when deciding whether or not we need to request retransmissions by sending a VNAK. This code could cause VNAKs to be sent erroneously in some cases, and to not be sent in other cases when it should have been. (closes issue #10237, reported and patched by mihai) ........ ................ * main/acl.c: Remove some debug code that was added in revision 75894, which removed some other debug code. :) 2007-07-19 12:38 +0000 [r75873-75894] Luigi Rizzo * main/acl.c: comment out some terribly expensive debugging code in the body of ast_apply_ha() * channels/chan_sip.c: print more of the network settings (externip, externhost etc.) in the "sip show settings" cli output. I have put these in a separate section, probably even bindaddr and SIP port should go there. There are more things to add here e.g. localnet and so on. * channels/chan_sip.c: document the use of externip, externhost and other nat-related options, as well as the handling of the sip socket. * channels/chan_sip.c: ast_sip_ouraddrfor() never fails, so make it void and remove the code that would never be called. * channels/chan_sip.c: portability fix: use %f instead of %lf when printing double. The l is useless. 2007-07-19 04:45 +0000 [r75841-75857] Tilghman Lesher * channels/misdn/ie.c, channels/misdn/isdn_lib.c: Allow chan_misdn to build in dev-mode * apps/app_rpt.c: Fix trunk where I broke it earlier (for ast_strftime branch) 2007-07-18 23:00 +0000 [r75808] Jason Parker * /, channels/chan_skinny.c: Merged revisions 75807 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r75807 | qwell | 2007-07-18 17:59:18 -0500 (Wed, 18 Jul 2007) | 1 line Need to make sure we set milliseconds and timestamp - pointed out by the recent ast_ time stuff from Tilghman ........ 2007-07-18 22:52 +0000 [r75806] Russell Bryant * channels/chan_iax2.c: I thought I noticed a memory leak earlier when I saw that the contents of this list were not destroyed when the module is unloaded. However, after reading the code related to the use of this list a lot today, I realized that it isn't necessary. So, I have added a comment to explain why it isn't necessary. 2007-07-18 22:40 +0000 [r75805] Tilghman Lesher * channels/chan_iax2.c: Change IAX variables to use datastores (closes issue #9315) 2007-07-18 21:10 +0000 [r75761] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 75759 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r75759 | russell | 2007-07-18 16:09:46 -0500 (Wed, 18 Jul 2007) | 13 lines Merged revisions 75757 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r75757 | russell | 2007-07-18 16:09:13 -0500 (Wed, 18 Jul 2007) | 5 lines When traversing the queue of frames for possible retransmission after receiving a VNAK, handle sequence number wraparound so that all frames that should be retransmitted actually do get retransmitted. (issue #10227, reported and patched by mihai) ........ ................ 2007-07-18 20:43 +0000 [r75750] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 75749 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r75749 | tilghman | 2007-07-18 15:40:18 -0500 (Wed, 18 Jul 2007) | 10 lines Merged revisions 75748 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r75748 | tilghman | 2007-07-18 15:31:36 -0500 (Wed, 18 Jul 2007) | 2 lines Store prior to copy (closes issue #10193) ........ ................ 2007-07-18 20:18 +0000 [r75714-75734] Jason Parker * /, channels/chan_skinny.c: Merged revisions 75732 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r75732 | qwell | 2007-07-18 15:17:27 -0500 (Wed, 18 Jul 2007) | 1 line Umm, why are we transmitting dialtone on cfwdall? ........ * /, channels/chan_skinny.c: Merged revisions 75711 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #9245) ........ r75711 | qwell | 2007-07-18 14:54:32 -0500 (Wed, 18 Jul 2007) | 4 lines Fixes for 7935/7936 conference phones. Issue 9245, patch by slimey. ........ 2007-07-18 19:51 +0000 [r75710] Jason Parker * /, channels/chan_skinny.c: Merged revisions 75707 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #9887) ........ r75707 | qwell | 2007-07-18 14:48:12 -0500 (Wed, 18 Jul 2007) | 4 lines Fix issues with new 79x1 phones. Issue 9887, patches by DEA ........ 2007-07-18 19:50 +0000 [r75709] Russell Bryant * channels/chan_iax2.c: convert some lines indented with spaces to tabs 2007-07-18 19:47 +0000 [r75706] Tilghman Lesher * main/say.c, funcs/func_strings.c, main/utils.c, apps/app_alarmreceiver.c, include/asterisk/localtime.h, cdr/cdr_pgsql.c, cdr/cdr_radius.c, channels/chan_iax2.c, main/loader.c, main/cli.c, cdr/cdr_csv.c, main/cdr.c, channels/chan_phone.c, main/manager.c, channels/chan_skinny.c, cdr/cdr_sqlite.c, apps/app_minivm.c, channels/misdn/ie.c, main/logger.c, main/http.c, main/stdtime/localtime.c, cdr/cdr_odbc.c, apps/app_rpt.c, include/asterisk/options.h, channels/chan_mgcp.c, cdr/cdr_manager.c, main/pbx.c, channels/chan_zap.c, funcs/func_timeout.c, channels/chan_sip.c, channels/chan_agent.c, channels/iax2-parser.c, apps/app_playback.c, cdr/cdr_tds.c, main/callerid.c, res/snmp/agent.c, apps/app_sms.c, include/asterisk/strings.h, main/asterisk.c, apps/app_voicemail.c: Merge in ast_strftime branch, which changes timestamps to be accurate to the microsecond, instead of only to the second 2007-07-18 17:59 +0000 [r75659] Dwayne M. Hubbard * /, apps/app_queue.c: Merged revisions 75658 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r75658 | dhubbard | 2007-07-18 12:56:30 -0500 (Wed, 18 Jul 2007) | 9 lines Merged revisions 75657 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r75657 | dhubbard | 2007-07-18 12:48:33 -0500 (Wed, 18 Jul 2007) | 1 line removed the word 'pissed' from ast_log(...) function call for BE-90 ........ ................ 2007-07-18 15:45 +0000 [r75586-75624] Joshua Colp * /, channels/chan_sip.c: Merged revisions 75623 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r75623 | file | 2007-07-18 12:44:02 -0300 (Wed, 18 Jul 2007) | 2 lines Few more places that needs to check for onhold state. ........ * /, channels/chan_sip.c: Merged revisions 75621 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r75621 | file | 2007-07-18 12:41:06 -0300 (Wed, 18 Jul 2007) | 5 lines (closes issue #10165) Reported by: elandivar It is possible for hold status to exist without call limits set, so we need to ensure update_call_counter is executed regardless. ........ * /, channels/chan_h323.c: Merged revisions 75619 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r75619 | file | 2007-07-18 12:25:45 -0300 (Wed, 18 Jul 2007) | 2 lines Don't bother reloading chan_h323 if it did not load successfully in the first place. This would otherwise cause a crash. ........ * funcs/func_curl.c: Clean up func_curl a bit. 2007-07-18 14:35 +0000 [r75585] Steve Murphy * main/channel.c, channels/chan_sip.c, res/res_features.c, pbx/pbx_dundi.c, main/rtp.c, apps/app_voicemail.c: This corrects the problem with flags and %lld formats on 64-bit machines, where uint64_t is NOT acceptable for %lld, and also works on 32-bit machines. At least, with gcc. 2007-07-18 14:20 +0000 [r75566-75584] Joshua Colp * /, pbx/pbx_dundi.c: Merged revisions 75583 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r75583 | file | 2007-07-18 11:18:53 -0300 (Wed, 18 Jul 2007) | 5 lines (closes issue #10224) Reported by: irroot Record the threadid of each running thread before shutting them down as the thread themselves may change the value. ........ * channels/chan_sip.c, channels/chan_agent.c, pbx/pbx_realtime.c, apps/app_voicemail.c: Minor code tweaks. Variables were being checked wrong in some situations and didn't need to be checked in others. 2007-07-18 12:38 +0000 [r75530] Tilghman Lesher * /, apps/app_meetme.c: Merged revisions 75529 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r75529 | tilghman | 2007-07-18 07:29:41 -0500 (Wed, 18 Jul 2007) | 2 lines Using a freed frame causes crashes (closes issue #9317) ........ 2007-07-17 21:52 +0000 [r75505] Steve Murphy * pbx/pbx_ael.c: Spotted this bug today myself, trying to reproduce a BE bug. Use a vert bar instead of a comma, when calling RAND. 2007-07-17 20:58 +0000 [r75446-75451] Russell Bryant * /, channels/chan_skinny.c: Merged revisions 75450 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r75450 | russell | 2007-07-17 15:57:56 -0500 (Tue, 17 Jul 2007) | 11 lines Merged revisions 75449 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r75449 | russell | 2007-07-17 15:57:09 -0500 (Tue, 17 Jul 2007) | 3 lines Properly check for the length in the skinny packet to prevent an invalid memcpy. (ASA-2007-016) ........ ................ * channels/iax2-parser.h, /, channels/chan_iax2.c, channels/iax2-parser.c: Merged revisions 75445 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r75445 | russell | 2007-07-17 15:48:21 -0500 (Tue, 17 Jul 2007) | 13 lines Merged revisions 75444 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r75444 | russell | 2007-07-17 15:45:27 -0500 (Tue, 17 Jul 2007) | 5 lines Ensure that when encoding the contents of an ast_frame into an iax_frame, that the size of the destination buffer is known in the iax_frame so that code won't write past the end of the allocated buffer when sending outgoing frames. (ASA-2007-014) ........ ................ 2007-07-17 20:42 +0000 [r75438-75442] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 75441 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r75441 | russell | 2007-07-17 15:42:12 -0500 (Tue, 17 Jul 2007) | 12 lines Merged revisions 75440 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r75440 | russell | 2007-07-17 15:41:41 -0500 (Tue, 17 Jul 2007) | 4 lines After parsing information elements in IAX frames, set the data length to zero, so that code later on does not think it has data to copy. (ASA-2007-015) ........ ................ 2007-07-17 20:05 +0000 [r75406] Mark Michelson * apps/app_dial.c, /: Merged revisions 75405 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r75405 | mmichelson | 2007-07-17 15:03:48 -0500 (Tue, 17 Jul 2007) | 6 lines Fixing an error I made earlier. ast_fileexists can return -1 on failure, so I need to be sure that we only enter the if statement if it is successful. Related to my fix to issue #10186 ........ 2007-07-17 20:01 +0000 [r75402-75404] Russell Bryant * main/pbx.c, /: Merged revisions 75403 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r75403 | russell | 2007-07-17 15:01:12 -0500 (Tue, 17 Jul 2007) | 12 lines (closes issue #10209) Reported by: juggie Patches: 10209-trunk-2.patch uploaded by juggie Tested by: juggie, blitzrage In ast_pbx_run(), mark a channel as hung up after an application returned -1, or when it runs out of extensions to execute. This is so that code can detect that this channel has been hung up for things like making sure DeadAGI is used on actual dead channels, and is beneficial for other things, like making sure someone doesn't try to start spying on a channel that is about to go away. ........ * /, res/res_agi.c: Merged revisions 75401 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r75401 | russell | 2007-07-17 14:45:07 -0500 (Tue, 17 Jul 2007) | 3 lines Remove a duplicated newline character in AGI debug output. (closes issue #10207, patch by seanbright) ........ 2007-07-17 19:40 +0000 [r75400] Steve Murphy * apps/app_dial.c, include/asterisk/utils.h, channels/chan_local.c, channels/chan_sip.c, include/asterisk/dundi.h, res/res_features.c, include/asterisk/chanspy.h, include/asterisk/speech.h, channels/iax2-provision.c, include/asterisk/cdr.h, include/asterisk/channel.h, res/res_musiconhold.c, channels/chan_iax2.c, main/rtp.c, channels/iax2-provision.h, main/loader.c, include/asterisk/features.h, include/asterisk/abstract_jb.h, main/channel.c, funcs/func_odbc.c, include/asterisk/module.h, include/asterisk/jabber.h, apps/app_minivm.c, utils/ael_main.c, pbx/pbx_dundi.c, apps/app_mixmonitor.c, utils/check_expr.c, apps/app_voicemail.c: via 10206, I have added an option (e) to Dial to allow the h exten to get run on peer. Had to upgrade ast_flag stuff to 64 bits to do this. 2007-07-17 14:48 +0000 [r75381] Joshua Colp * include/asterisk/config.h: Make trunk build once again. 2007-07-17 14:32 +0000 [r75365-75379] Luigi Rizzo * include/asterisk/config.h, main/config.c: Introduce ast_parse_arg() , a generic function to parse strings in a consistent way. This is meant to replace the custom code which is repeated all over the place in the various files when parsing config files, CLI entries and other string information. Right now the code supports parsing int32, uint32 and sockaddr_in with optional default values and bound checks. It contains minimal error checking, but that can be easily extended as the need arises. Being a new API i am introducing this only in trunk, though I believe that once the interface has been ironed out it might become a worthwhile addition to 1.4 as well - basically, the first time we will need to fix a piece of argument parsing code, we might as well bring in this change and use the new API instead. * apps/app_minivm.c: Initialize a variable to avoid a warning when the compiler (and/or the optimization level) may think it is used uninitialized. The code was indeed correct, but unfortunately the result of some compiler checks such as -Wunused and -Wuninitialized depends heavily on the optimization level. 2007-07-17 12:01 +0000 [r75351] Jason Parker * apps/app_dial.c: Fix an incorrect parenthesization (TODO: Find a better word) in app_dial Pointed out by Fanzhou Zhao Closes issue #10216 2007-07-16 20:58 +0000 [r75307] Kevin P. Fleming * /, main/dns.c: Merged revisions 75306 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r75306 | kpfleming | 2007-07-16 15:53:24 -0500 (Mon, 16 Jul 2007) | 11 lines Merged revisions 75304 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r75304 | kpfleming | 2007-07-16 15:46:58 -0500 (Mon, 16 Jul 2007) | 3 lines provide proper copyright/license attribution for this structure that was copied from a BSD-licensed header file long, long ago... ........ ................ 2007-07-16 18:38 +0000 [r75255-75260] Joshua Colp * main/pbx.c, include/asterisk/pbx.h: Change the function name slightly... just for kpfleming! * configure, include/asterisk/autoconfig.h.in, configure.ac: Add in check for the GCC attribute deprecated. It may be used soon! * funcs/func_enum.c, funcs/func_rand.c, main/pbx.c, funcs/func_curl.c, funcs/func_version.c, funcs/func_cut.c, funcs/func_vmcount.c, include/asterisk/pbx.h, funcs/func_realtime.c: For my next trick I will make it so dialplan functions no longer need to call ast_module_user_add and ast_module_user_remove. These are now called in the ast_func_read and ast_func_write functions outside of the module. 2007-07-16 18:18 +0000 [r75254] Mark Michelson * apps/app_dial.c, /: Merged revisions 75253 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r75253 | mmichelson | 2007-07-16 13:16:15 -0500 (Mon, 16 Jul 2007) | 8 lines Restoring functionality from 1.2 wherein Retrydial will not exit if there is no announce file specified. This change makes it so that if there is no announce file specified, the application will continue until finished (or caller hangs up). If a bogus announce file is specified, then a warning message will be printed saying that the file could not be found, but execution will still continue. (closes issue #10186, reported by jon, patched by me) ........ 2007-07-16 15:57 +0000 [r75183-75227] Joshua Colp * apps/app_verbose.c: I found this sillyness when I did my ast_module_user conversion. Return immediately if no data was passed to the Verbose application. * apps/app_readfile.c, apps/app_record.c, apps/app_sayunixtime.c, apps/app_test.c, apps/app_alarmreceiver.c, apps/app_image.c, apps/app_chanisavail.c, apps/app_ices.c, apps/app_exec.c, apps/app_skel.c, apps/app_zapscan.c, apps/app_dumpchan.c, apps/app_zapras.c, apps/app_amd.c, apps/app_url.c, apps/app_externalivr.c, apps/app_milliwatt.c, apps/app_dial.c, main/pbx.c, apps/app_page.c, apps/app_privacy.c, apps/app_echo.c, apps/app_softhangup.c, apps/app_disa.c, apps/app_morsecode.c, apps/app_talkdetect.c, apps/app_transfer.c, apps/app_db.c, apps/app_playback.c, apps/app_speech_utils.c, apps/app_zapbarge.c, apps/app_waitforring.c, apps/app_sendtext.c, apps/app_macro.c, apps/app_zapateller.c, apps/app_chanspy.c, apps/app_mixmonitor.c, apps/app_cdr.c, apps/app_voicemail.c, apps/app_meetme.c, apps/app_dictate.c, apps/app_authenticate.c, apps/app_userevent.c, apps/app_followme.c, apps/app_controlplayback.c, apps/app_osplookup.c, apps/app_setcallerid.c, apps/app_minivm.c, apps/app_mp3.c, apps/app_directory.c, apps/app_rpt.c, apps/app_ivrdemo.c, apps/app_parkandannounce.c, apps/app_adsiprog.c, apps/app_while.c, apps/app_nbscat.c, apps/app_read.c, apps/app_festival.c, apps/app_system.c, apps/app_getcpeid.c, apps/app_queue.c, apps/app_channelredirect.c, apps/app_forkcdr.c, apps/app_flash.c, apps/app_directed_pickup.c, apps/app_sms.c, include/asterisk/pbx.h, apps/app_senddtmf.c, apps/app_stack.c, apps/app_verbose.c: Applications no longer need to call ast_module_user_add and ast_module_user_remove. This is now taken care of in the pbx_exec function outside of the application. * apps/app_readfile.c, res/res_features.c, apps/app_record.c, apps/app_sayunixtime.c, apps/app_test.c, apps/app_alarmreceiver.c, apps/app_image.c, apps/app_chanisavail.c, apps/app_ices.c, apps/app_exec.c, apps/app_zapscan.c, apps/app_dumpchan.c, apps/app_zapras.c, apps/app_amd.c, apps/app_url.c, apps/app_externalivr.c, apps/app_milliwatt.c, apps/app_dial.c, apps/app_page.c, apps/app_privacy.c, apps/app_echo.c, apps/app_softhangup.c, apps/app_disa.c, apps/app_morsecode.c, funcs/func_cut.c, apps/app_talkdetect.c, apps/app_transfer.c, apps/app_playback.c, apps/app_speech_utils.c, funcs/func_curl.c, apps/app_zapbarge.c, apps/app_waitforring.c, apps/app_sendtext.c, apps/app_macro.c, apps/app_zapateller.c, apps/app_mixmonitor.c, apps/app_chanspy.c, apps/app_cdr.c, apps/app_voicemail.c, apps/app_meetme.c, apps/app_authenticate.c, apps/app_userevent.c, funcs/func_vmcount.c, apps/app_followme.c, funcs/func_enum.c, res/res_config_odbc.c, apps/app_setcallerid.c, apps/app_osplookup.c, apps/app_minivm.c, res/res_agi.c, apps/app_mp3.c, res/res_realtime.c, apps/app_rpt.c, apps/app_ivrdemo.c, apps/app_parkandannounce.c, apps/app_adsiprog.c, apps/app_while.c, apps/app_nbscat.c, res/res_config_pgsql.c, apps/app_read.c, apps/app_festival.c, apps/app_waitforsilence.c, apps/app_system.c, apps/app_queue.c, apps/app_getcpeid.c, funcs/func_realtime.c, apps/app_forkcdr.c, apps/app_channelredirect.c, apps/app_flash.c, funcs/func_blacklist.c, apps/app_sms.c, apps/app_senddtmf.c, apps/app_stack.c, apps/app_verbose.c: It is no longer required for each module that deals with a channel to call ast_module_user_hangup_all in it's unload function. The loader will automatically perform this action for it. 2007-07-16 02:51 +0000 [r75163-75164] Russell Bryant * include/asterisk/devicestate.h, include/asterisk/dundi.h, include/asterisk/enum.h, include/asterisk/config.h, include/asterisk/io.h, include/asterisk/cli.h, include/asterisk/channel.h, include/asterisk/cdr.h, include/asterisk/manager.h, include/asterisk/tdd.h, include/asterisk/abstract_jb.h, include/asterisk/file.h, include/asterisk/res_odbc.h, include/asterisk/adsi.h, include/asterisk/crypto.h, include/asterisk/doxyref.h, include/asterisk/image.h, include/asterisk/musiconhold.h, include/asterisk/jabber.h, include/asterisk/linkedlists.h, include/asterisk/module.h, include/asterisk/strings.h, include/asterisk/pbx.h, include/asterisk/frame.h, include/asterisk/say.h, include/asterisk/translate.h: Merge a bunch of doxygen updates to header files. This includes changes to use the \retval tag for documenting return values, fixing various warnings when generating the documentation, and various other things. (closes issue #10203, snuffy) * funcs/func_iconv.c: Cast the 2nd argument to iconv() to a void *, as some systems define it as a (const char *), while others define it as (char *). This is done to suppress compiler warnings about it. 2007-07-13 20:37 +0000 [r75109] Russell Bryant * /: Merged revisions 75108 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r75108 | russell | 2007-07-13 15:36:16 -0500 (Fri, 13 Jul 2007) | 11 lines Merged revisions 75107 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r75107 | russell | 2007-07-13 15:35:22 -0500 (Fri, 13 Jul 2007) | 3 lines Fix a couple potential minor memory leaks. load_moh_classes() could return without destroying the loaded configuration. ........ ................ 2007-07-13 20:16 +0000 [r75082] Mark Michelson * /, apps/app_chanspy.c: Merged revisions 75078 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r75078 | mmichelson | 2007-07-13 15:15:30 -0500 (Fri, 13 Jul 2007) | 13 lines Merged revisions 75066 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r75066 | mmichelson | 2007-07-13 15:10:39 -0500 (Fri, 13 Jul 2007) | 5 lines Fixed an issue where chanspy flags were uninitialized if no options were passed. What triggered this investigation was an IRC chat where some people's quiet flags were set while others' weren't even though none of them had specified the q option. ........ ................ 2007-07-13 20:15 +0000 [r75054-75077] Russell Bryant * main/rtp.c: resolve a compiler warning * /, res/res_musiconhold.c: Merged revisions 75067 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r75067 | russell | 2007-07-13 15:10:40 -0500 (Fri, 13 Jul 2007) | 14 lines Merged revisions 75059 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r75059 | russell | 2007-07-13 15:07:21 -0500 (Fri, 13 Jul 2007) | 6 lines Ensure that adding a user to the list of users of a specific music on hold class is not done at the same time as any of the other operations on this list to prevent list corruption. Using the global moh_data lock for this is not ideal, but it is what is used to protect these lists everywhere else in the module, and I am only changing what is necessary to fix the bug. ........ ................ * channels/chan_zap.c, /: Merged revisions 75053 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r75053 | russell | 2007-07-13 14:11:26 -0500 (Fri, 13 Jul 2007) | 20 lines Merged revisions 75052 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r75052 | russell | 2007-07-13 14:10:00 -0500 (Fri, 13 Jul 2007) | 12 lines (closes issue #9660) Reported by: mmacvicar Patches submitted by: bbryant, russell Tested by: mmacvicar, marco, arcivanov, jmhunter, explidous When using a TDM400P (and probably other analog cards) there was a chance that you could hang up and pick the phone back up where it has been long enough to be not considered a flash hook, but too soon such that the device reports that it is busy and the person on the phone will only hear silence. This patch makes chan_zap more tolerant of this and gives the device a couple of seconds to succeed so the person on the phone happily gets their dialtone. ........ ................ 2007-07-13 16:22 +0000 [r75034] Luigi Rizzo * include/asterisk/rtp.h, main/rtp.c: Small improvement to the STUN support so it can be used by sockets other than RTP ones. The main change is a new API function in main/rtp.c (see there for a description) int ast_stun_request(int s, struct sockaddr_in *dst, const char *username, struct sockaddr_in *answer) which can be used to send an STUN request on a socket, and optionally wait for a reply and store the STUN_MAPPED_ADDRESS into the 'answer' argument (obviously, the version that waits for a reply is blocking, but this is no different from DNS resolutions). Internally there are minor modifications to let stun_handle_packet() be somewhat configurable on how to parse the body of responses. At the moment i am not committing any change to the clients, but adding STUN client support is extremely simple, e.g. chan_sip.c could do something like this: + add a variable to store the stun server address; static struct sockaddr_in stunaddr = { 0, }; /*!< stun server address */ + add code to parse a config file of the form "stunaddr=my.stun.server.org:3478" (not shown for brevity); + right after binding the main sip socket, talk to the stun server to determine the externally visible address if (stunaddr.sin_addr.s_addr != 0) ast_stun_request(sipsock, &stunaddr, NULL, &externip); so now 'externip' is set with the externally visible address. so it is really trivial. Similarly ast_stun_request could be called when creating the RTP socket (possibly adding a struct sockaddr_in field in the struct ast_rtp to store the externalip). 2007-07-12 23:02 +0000 [r74999] Mark Michelson * /, channels/chan_agent.c: Merged revisions 74997 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ ........ 2007-07-12 20:46 +0000 [r74956] Steve Murphy * /, channels/chan_sip.c: Merged revisions 74955 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r74955 | murf | 2007-07-12 14:42:08 -0600 (Thu, 12 Jul 2007) | 1 line This patch resolves 10143; thanks to irroot for the patch; looked acceptable. Let the community decide if it messes things up ........ 2007-07-12 19:19 +0000 [r74891-74923] Joshua Colp * main/channel.c, /: Merged revisions 74922 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r74922 | file | 2007-07-12 16:17:59 -0300 (Thu, 12 Jul 2007) | 2 lines Whoops... didn't want this to be returned to 0 each iteration. ........ * main/channel.c, /: Merged revisions 74888 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r74888 | file | 2007-07-12 14:16:28 -0300 (Thu, 12 Jul 2007) | 2 lines When waiting for a digit ensure that a begin frame was received with it, not just an end frame. (issue #10084 reported by rushowr) ........ 2007-07-12 16:54 +0000 [r74865-74867] Jason Parker * /, channels/chan_skinny.c: Merged revisions 74866 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r74866 | qwell | 2007-07-12 11:53:35 -0500 (Thu, 12 Jul 2007) | 1 line It helps if I actually add this stuff for the 7921 too - otherwise it won't actually do much of anything. ........ * /, channels/chan_skinny.c: Merged revisions 74864 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r74864 | qwell | 2007-07-12 11:48:49 -0500 (Thu, 12 Jul 2007) | 1 line Add device ID for 7921 wireless skinny phone ........ 2007-07-12 16:21 +0000 [r74850] Luigi Rizzo * main/rtp.c: more cleanup, this time to stun_handle_packet(). Among other things: + mark a potentially dangerous write-past-end-of-buffer + localize some variables in the block generating stun replies. As before, not ready yet for a merge to 1.4 2007-07-12 15:55 +0000 [r74816] Joshua Colp * /, res/res_musiconhold.c: Merged revisions 74815 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r74815 | file | 2007-07-12 12:53:55 -0300 (Thu, 12 Jul 2007) | 10 lines Merged revisions 74814 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r74814 | file | 2007-07-12 12:51:24 -0300 (Thu, 12 Jul 2007) | 2 lines Only print out a warning for situations where it is actually helpful. (issue #10187 reported by denke) ........ ................ 2007-07-12 15:42 +0000 [r74813] Luigi Rizzo * main/rtp.c: a little bit of code cleanup to rtp.c, mostly to function ast_rtp_new_with_bindaddr(): 1. add comments to the logic of the main loop; 2. use a common exit point on failure so the cleanup is done only in one place; 3. handle failures in rtp_socket() in the main loop of the function; No functional changes except for #3 above, so it is not yet worthwhile merging this and other changes to 1.4 Once the cleanup work on this file will be complete (which among other things should include some extensions to the stun support) it might be a good thing to push all the changes to 1.4 2007-07-11 23:05 +0000 [r74769] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 74767 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r74767 | russell | 2007-07-11 17:57:07 -0500 (Wed, 11 Jul 2007) | 13 lines Merged revisions 74766 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r74766 | russell | 2007-07-11 17:53:26 -0500 (Wed, 11 Jul 2007) | 5 lines The function make_trunk() can fail and return -1 instead of a valid new call number. Fix the uses of this function to handle this instead of treating it as the new call number. This would cause a deadlock and memory corruption. (possible cause of issue #9614 and others, patch by me) ........ ................ 2007-07-11 21:15 +0000 [r74726] Mark Michelson * /, channels/chan_agent.c: Merged revisions 74722 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r74722 | mmichelson | 2007-07-11 16:14:09 -0500 (Wed, 11 Jul 2007) | 13 lines Merged revisions 74719 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r74719 | mmichelson | 2007-07-11 16:12:30 -0500 (Wed, 11 Jul 2007) | 5 lines The cli command "agent logoff Agent/x soft" did not work...at all. Now it does. (closes issue #10178, reported and patched by makoto, with slight modification for 1.4 and trunk by me) ........ ................ 2007-07-11 21:09 +0000 [r74703-74713] Joshua Colp * res/res_agi.c: Code cleanup of res_agi * res/res_smdi.c: Code cleanup of res_smdi * pbx/pbx_spool.c: Clean up pbx_spool. So many nested if statements... * main/udptl.c, include/asterisk/udptl.h: Use linkedlist macros for UDPTL protocol list. 2007-07-11 18:35 +0000 [r74658] Russell Bryant * res/res_config_odbc.c: Merged revisions 74657 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r74657 | russell | 2007-07-11 13:34:51 -0500 (Wed, 11 Jul 2007) | 12 lines Merged revisions 74656 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r74656 | russell | 2007-07-11 13:33:23 -0500 (Wed, 11 Jul 2007) | 4 lines Make sure that the ESCAPE immediately follows the condition that uses LIKE. This fixes realtime extensions with ODBC. (closes issue #10175, reported by stuarth, patch by me) ........ ................ 2007-07-11 18:21 +0000 [r74636-74648] Steve Murphy * Makefile, /: Merged revisions 74642 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r74642 | murf | 2007-07-11 12:18:42 -0600 (Wed, 11 Jul 2007) | 1 line This fixes 10172, where the entire man8 dir gets removed during an uninstall of asterisk ........ * /: blocking 74628 from trunk... only applied to 1.4 2007-07-11 17:34 +0000 [r74575-74616] Joshua Colp * include/asterisk/speech.h, res/res_speech.c, apps/app_speech_utils.c: Use the linkedlists.h AST_LIST_NEXT macro for modifying the list of results. * channels/chan_phone.c, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 74572 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r74572 | file | 2007-07-11 14:03:08 -0300 (Wed, 11 Jul 2007) | 2 lines Instead of figuring out kernel versions that have compiler.h and not... let's just use autoconf to check for it's presence. (issue #10174 reported by francesco_r) ........ 2007-07-11 16:24 +0000 [r74571] Luigi Rizzo * main/rtp.c: add a bit of documentation on what the stun code in rtp.c does (which is very little, at the moment). Eventually, when the functionality is extended, the changes can be merged back to 1.4. At the moment this is pointless. Note, this change is whitespace only. 2007-07-11 16:19 +0000 [r74516-74570] Joshua Colp * include/asterisk/speech.h, res/res_speech.c, apps/app_speech_utils.c: Allow the native formats of a channel to influence the audio that is going to the engine. The best format will try to be chosen with an ultimate fallback to signed linear if possible. * res/res_speech.c: Can't forget to remember what format is in use for writing. * include/asterisk/speech.h, res/res_speech.c: Change the speech API to allow passing the format through to the engine. * channels/misdn/isdn_lib_intern.h: Change header a bit to get rid of a doxygen parse error. (issue #10177 reported by snuffy) * channels/chan_phone.c, /: Merged revisions 74515 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r74515 | file | 2007-07-11 11:09:13 -0300 (Wed, 11 Jul 2007) | 2 lines Only check if we need to do a SIGMA based tone generation if we have a card. (issue #10179 reported by mikowhy) ........ 2007-07-10 23:34 +0000 [r74477] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 74476 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r74476 | mmichelson | 2007-07-10 18:32:52 -0500 (Tue, 10 Jul 2007) | 5 lines Forwarding a message with IMAP storage was storing the message in the sender's box instead of the forwarded mailbox. (closes issue #10138, reported and patched by jaroth) ........ 2007-07-10 20:02 +0000 [r74375-74429] Jason Parker * /, apps/app_queue.c: Merged revisions 74428 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #10158) ................ r74428 | qwell | 2007-07-10 14:58:53 -0500 (Tue, 10 Jul 2007) | 14 lines Merged revisions 74427 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r74427 | qwell | 2007-07-10 14:57:20 -0500 (Tue, 10 Jul 2007) | 6 lines Fix an issue where it was possible to have a service level of over 100% Between the time recalc_holdtime and update_queue was called, it was possible that the call could have been hungup. Move both additions to the same place, so this won't happen. Issue 10158, initial patch by makoto, modified by me. ........ ................ * /, main/dns.c: Merged revisions 74388 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r74388 | qwell | 2007-07-10 14:10:36 -0500 (Tue, 10 Jul 2007) | 4 lines Don't use #if to check if something is defined - use #ifdef instead. Pointed out by kpfleming ........ * /, channels/chan_agent.c: Merged revisions 74379 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #10169) ................ r74379 | qwell | 2007-07-10 14:06:24 -0500 (Tue, 10 Jul 2007) | 12 lines Merged revisions 74376 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r74376 | qwell | 2007-07-10 14:03:45 -0500 (Tue, 10 Jul 2007) | 4 lines Fix an issue with wrapuptime not working when using AgentLogin. Issue 10169, patch by makoto, with a minor mod by me to not re-break issue 9618 ........ ................ * /, configure, include/asterisk/autoconfig.h.in, configure.ac, main/dns.c: Merged revisions 74374 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #10133) ................ r74374 | qwell | 2007-07-10 13:39:30 -0500 (Tue, 10 Jul 2007) | 13 lines Merged revisions 74373 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r74373 | qwell | 2007-07-10 13:37:23 -0500 (Tue, 10 Jul 2007) | 5 lines Use res_ndestroy on systems that have it. Otherwise, use res_nclose. This prevents a memleak on NetBSD - and possibly others. Issue 10133, patch by me, reported and tested by scw ........ ................ 2007-07-10 16:01 +0000 [r74324] Russell Bryant * /, res/res_musiconhold.c: Merged revisions 74323 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r74323 | russell | 2007-07-10 11:00:11 -0500 (Tue, 10 Jul 2007) | 1 line fix an uninitialized variable ........ 2007-07-10 15:41 +0000 [r74318-74319] Jason Parker * /: svn revert != svn resolved Fix merged property... * apps/app_voicemail.c: Merged revisions 74317 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes issue #10170) ................ r74317 | qwell | 2007-07-10 10:38:32 -0500 (Tue, 10 Jul 2007) | 12 lines Merged revisions 74316 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r74316 | qwell | 2007-07-10 10:37:54 -0500 (Tue, 10 Jul 2007) | 4 lines Fix a small typo in description in of Voicemail() application. Issue 10170, patch by casper. ........ ................ 2007-07-10 15:32 +0000 [r74315] Russell Bryant * res/res_config_odbc.c, /: Merged revisions 74314 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r74314 | russell | 2007-07-10 10:31:41 -0500 (Tue, 10 Jul 2007) | 11 lines Merged revisions 74313 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r74313 | russell | 2007-07-10 10:30:20 -0500 (Tue, 10 Jul 2007) | 3 lines Only use ESCAPE when LIKE is used. (issue #10075, this part reported by jmls on IRC, patch by me) ........ ................ 2007-07-10 15:07 +0000 [r74272] Jason Parker * channels/chan_agent.c, include/asterisk/monitor.h, apps/app_queue.c, res/res_monitor.c: Fix building that was broken by recent monitor.h changes. Thanks Russell for pointing this out (and pointing out what I probably did to prevent gcc from fixing it - don't ctrl-C builds) 2007-07-10 14:51 +0000 [r74263-74266] Joshua Colp * /, main/app.c: Merged revisions 74265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r74265 | file | 2007-07-10 11:50:00 -0300 (Tue, 10 Jul 2007) | 10 lines Merged revisions 74264 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r74264 | file | 2007-07-10 11:48:00 -0300 (Tue, 10 Jul 2007) | 2 lines Ensure the group information category exists before trying to do a string comparison with it. (issue #10171 reported by mlegas) ........ ................ 2007-07-09 21:32 +0000 [r74212] Russell Bryant * /, configure, configure.ac: Merged revisions 74211 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r74211 | russell | 2007-07-09 16:31:30 -0500 (Mon, 09 Jul 2007) | 5 lines Update the configure script to check for a required function that is not present in the 1.2 version of libpri. This will prevent the configure script from thinking that it has compatible libpri support for Asterisk 1.4, when it actually does not because the installed version is from 1.2. ........ 2007-07-09 20:58 +0000 [r74164] Jason Parker * include/asterisk/monitor.h, res/res_monitor.c: (closes issue #7596) Reported by: julien23 Patches submitted by: julien23 Add the ability to disable recording the input or output streams in res_monitor. 2007-07-09 20:54 +0000 [r74163] Russell Bryant * /, res/res_musiconhold.c: Merged revisions 74162 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r74162 | russell | 2007-07-09 15:53:46 -0500 (Mon, 09 Jul 2007) | 9 lines (closes issue #10123) Reported by: blitzrage Patches submitted by: juggie, qwell, me Tested by: blitzrage When trying to find a music on hold class to use, try all of the options, instead of only the first one that is set. Also, change the MusicOnHold applications to not hang up on the channel when a class can not be found. ........ 2007-07-09 20:21 +0000 [r74160] Jason Parker * channels/chan_zap.c, /: Merged revisions 74159 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 Closes issue #9186 ................ r74159 | qwell | 2007-07-09 15:19:28 -0500 (Mon, 09 Jul 2007) | 16 lines Merged revisions 74158 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r74158 | qwell | 2007-07-09 15:18:15 -0500 (Mon, 09 Jul 2007) | 8 lines Several chan_zap options were not working on reload because they were arbitrarily disallowed when reloading some/most PRI options (such as signalling) was disallowed. Options such as polarityonanswerdelay and answeronpolarityswitch can safely be changed on a reload. This corrects that behavior. Issue 9186, patch by tzafrir. ........ ................ 2007-07-09 18:58 +0000 [r74125] Russell Bryant * channels/chan_agent.c: remove an unused variable 2007-07-09 18:43 +0000 [r74121-74123] Mark Michelson * /: Merged revisions 74122 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r74122 | mmichelson | 2007-07-09 13:38:28 -0500 (Mon, 09 Jul 2007) | 3 lines Forgot to get rid of an extraneous debug message. ........ * /, apps/app_queue.c: Merged revisions 74120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r74120 | mmichelson | 2007-07-09 13:32:50 -0500 (Mon, 09 Jul 2007) | 6 lines The n option for Queue should make the queue exit immediately after failure to reach any members and should not be dependent on the timeout value passed to Queue (closes issue #10127, reported by bcnit, repaired by me) ........ 2007-07-09 16:35 +0000 [r74084] Russell Bryant * apps/app_queue.c: Add Queue and DestinationChannel headers to the AgentCalled manager event to be more like the rest of the events in this module. (closes issue #10114, patch by kwakwaversal) 2007-07-09 15:34 +0000 [r74083] Joshua Colp * /, channels/chan_skinny.c: Merged revisions 74082 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r74082 | file | 2007-07-09 12:32:43 -0300 (Mon, 09 Jul 2007) | 2 lines Only destroy the scheduler context if it was allocated. (issue #10124 reported by gzero) ........ 2007-07-09 14:58 +0000 [r74048] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 74047 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r74047 | mmichelson | 2007-07-09 09:57:41 -0500 (Mon, 09 Jul 2007) | 4 lines Fixed a logic error in leave_voicemail. Pass the mailbox instead of the context to inbox_count when the context is "default." (closes issue #10135, reported by yannj, repaired by me) ........ 2007-07-09 14:50 +0000 [r74044-74046] Joshua Colp * /, channels/chan_skinny.c, pbx/pbx_dundi.c: Merged revisions 74045 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r74045 | file | 2007-07-09 11:49:05 -0300 (Mon, 09 Jul 2007) | 2 lines Few minor thread synchronization tweaks. (issue #10124 reported by gzero) ........ * /, configure, acinclude.m4: Merged revisions 74043 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r74043 | file | 2007-07-09 11:34:33 -0300 (Mon, 09 Jul 2007) | 2 lines Use AC_CHECK_HEADER to check for ptlib/openh323 to allow for cross compiling. (issue #9675 reported by zandbelt) ........ 2007-07-09 08:30 +0000 [r74024-74025] Olle Johansson * CHANGES: Update with new features * apps/app_dial.c, channels/chan_local.c, channels/chan_sip.c, include/asterisk/channel.h: Implementation of a feature that will disable "missed calls" counters on SIP phones. If the call is answered by another phone, other phones won't display the call as "missed". You can also add an option to the dial command so that you can have a "followme" scenario and not count the calls as "missed" when you cancel the call. Thanks to Ramon and Frank for feedback on this feature. 2007-07-09 04:09 +0000 [r73994] Tilghman Lesher * include/asterisk/app.h, /, channels/chan_sip.c, main/ast_expr2f.c, include/asterisk/channel.h, funcs/func_devstate.c, apps/app_voicemail.c: Merged revisions 73985 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r73985 | tilghman | 2007-07-08 23:03:20 -0500 (Sun, 08 Jul 2007) | 2 lines Doxygen formatting fixes; fixes errors while 'make progdocs'. (Closes issue #10104) ........ 2007-07-09 03:14 +0000 [r73931-73983] Joshua Colp * main/cdr.c, /: Merged revisions 73980 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r73980 | file | 2007-07-09 00:13:19 -0300 (Mon, 09 Jul 2007) | 2 lines Give Agent channel names priority when doing CDR merging. (issue #10011 reported by krtorio) ........ * res/res_features.c: Use linkedlist macros for parking. * main/manager.c: Make sure the idText variable is empty, and put it in the right place for the manager ack packet. (issue #10152 reported by srt) * /, pbx/pbx_config.c: Merged revisions 73930 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r73930 | file | 2007-07-08 22:13:57 -0300 (Sun, 08 Jul 2007) | 2 lines Add a few sanity checks when writing out the dialplan. (issue #10157 reported by dome) ........ 2007-07-08 21:01 +0000 [r73911] Tilghman Lesher * configure, include/asterisk/autoconfig.h.in, main/ast_expr2.h, main/ast_expr2.y, configure.ac, main/ast_expr2.c: Restore EXP2 and LOG2 functions, by providing mathematical identify functions, when the underlying C functions are not available. 2007-07-08 13:22 +0000 [r73886] Russell Bryant * res/res_features.c: ast_exists_extension() does not return an ast_device_state, so change this function to explicitly check for the int return value. Also, make a few other minor changes such as removing a variable. 2007-07-08 09:49 +0000 [r73850] Olle Johansson * /, channels/chan_sip.c: Merged revisions 73849 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r73849 | oej | 2007-07-08 11:47:31 +0200 (Sun, 08 Jul 2007) | 2 lines While tracking down a bug, I need some more history. Dumphistory is very useful, indeed. ........ 2007-07-07 16:44 +0000 [r73821] Steve Murphy * configure, include/asterisk/autoconfig.h.in, main/ast_expr2.y, configure.ac, bootstrap.sh, main/ast_expr2.c: These changes fix 10145 and 10150, a prob with BSD and exp2/log2 not existing, as well as the bootstrap needing a small upgrade for openbsd. Many thanks to mvanbaak 2007-07-06 23:05 +0000 [r73771] Russell Bryant * /, channels/chan_sip.c: Merged revisions 73769 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r73769 | russell | 2007-07-06 18:02:58 -0500 (Fri, 06 Jul 2007) | 12 lines Merged revisions 73768 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r73768 | russell | 2007-07-06 18:01:22 -0500 (Fri, 06 Jul 2007) | 4 lines If a sip_pvt struct has already registered an extension state callback, remove the old one before adding a new one. If this isn't done, Asterisk will crash. (issue #10120) ........ ................ 2007-07-06 16:39 +0000 [r73728] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 73727 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r73727 | mmichelson | 2007-07-06 11:36:17 -0500 (Fri, 06 Jul 2007) | 8 lines Fixing a rare case which causes voicemail to crash when compiled with IMAP storage. inboxcount has the possibility of finding an "interactive" vm_state when no persistent "non-interactive" vm_state exists for that mailbox. If this should happen when someone attempts to leave a message, it results in a crash. This patch, along with my commit in revision 72670 fix issue 10053, reported by jaroth. closes issue #10053 ........ 2007-07-06 16:30 +0000 [r73726] Kevin P. Fleming * main/minimime/mimeparser.yy.c, main/minimime/mimeparser.h, main/minimime/mimeparser.tab.c, main/minimime/mimeparser.y, main/minimime/Makefile, main/minimime/mimeparser.l, main/minimime/mimeparser.tab.h, main/minimime/mm_parse.c: eliminate another batch of compiler warnings (and a bug, although in code we aren't using)... note that this required manually editing the lexer output code (generated by flex), so some of them will come back if the lexer is rebuilt 2007-07-06 16:14 +0000 [r73680-73701] Russell Bryant * res/res_config_odbc.c, /: Merged revisions 73696 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r73696 | russell | 2007-07-06 11:12:51 -0500 (Fri, 06 Jul 2007) | 16 lines Merged revisions 73684 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r73684 | russell | 2007-07-06 11:06:27 -0500 (Fri, 06 Jul 2007) | 8 lines (closes issue #10075) Reported by: apsaras Patches submitted by: Corydon76 Tested by: apsaras Fix a problem with MSSQL 2005 by explicitly stating that '\' is being used as an escape character. ........ ................ * /, channels/chan_sip.c: Merged revisions 73679 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r73679 | russell | 2007-07-06 10:57:25 -0500 (Fri, 06 Jul 2007) | 15 lines Merged revisions 73678 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r73678 | russell | 2007-07-06 10:55:41 -0500 (Fri, 06 Jul 2007) | 7 lines (closes issue #10125) Reported by: makoto Patches submitted by: makoto This fixes a crash in chan_sip that happens when the bindaddr setting is not valid on Asterisk startup, gets fixed, and then a reload gets issued. ........ ................ 2007-07-06 15:47 +0000 [r73677] Kevin P. Fleming * channels/busy.h (added), channels/ringtone.h (added), channels/Makefile, channels: it really seems pointless to run gentone to create these header files every time we build Asterisk... 2007-07-06 15:28 +0000 [r73676] Mark Michelson * /, channels/chan_agent.c: Merged revisions 73675 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r73675 | mmichelson | 2007-07-06 10:27:28 -0500 (Fri, 06 Jul 2007) | 13 lines Merged revisions 73674 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r73674 | mmichelson | 2007-07-06 10:26:40 -0500 (Fri, 06 Jul 2007) | 5 lines Fixed a bug wherein agents get stuck busy. (issue 9618, reported by jiddings, patched by moi) closes issue #9618 ........ ................ 2007-07-06 03:48 +0000 [r73557-73633] Russell Bryant * CHANGES: Redistribute a lot of the items that were in the Misc. section * CHANGES: note TLS support for manager and HTTP in CHANGES * CREDITS: Philippe was listed twice * /, BUGS: Merged revisions 73629 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r73629 | russell | 2007-07-05 22:34:46 -0500 (Thu, 05 Jul 2007) | 1 line fix a little spelling error ........ * /, channels/chan_sip.c: Merged revisions 73598 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r73598 | russell | 2007-07-05 18:59:22 -0500 (Thu, 05 Jul 2007) | 3 lines Fix a crash in chan_sip. Don't try to stop the monitor thread if it was never started. (closes issue #10124, reported by gzero, fixed by me) ........ * /, channels/chan_iax2.c: Merged revisions 73555 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r73555 | russell | 2007-07-05 18:05:33 -0500 (Thu, 05 Jul 2007) | 3 lines copy from the correct buffer when deferring a full frame (related to issue #9937) ........ 2007-07-05 22:48 +0000 [r73553] Kevin P. Fleming * main/minimime/mm_contenttype.c, main/minimime/mm_envelope.c, main/minimime/mm_mimepart.c, main/minimime/mm_param.c, main/minimime/mm_context.c, main/minimime/mm_mimeutil.c: comment out some code that is not used and does not have prototypes 2007-07-05 22:32 +0000 [r73552] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 73551 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r73551 | russell | 2007-07-05 17:31:31 -0500 (Thu, 05 Jul 2007) | 6 lines * Store the call number that a thread is processing without the full frame bit set to ease debugging * When deferring a full frame for processing, stick it into the queue for the thread that is processing frames for that call, not the one that read the current frame and is about to go back into the idle list (related to issue #9937) ........ 2007-07-05 22:29 +0000 [r73550] Kevin P. Fleming * /, channels/chan_sip.c: Merged revisions 73548 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r73548 | kpfleming | 2007-07-05 17:20:44 -0500 (Thu, 05 Jul 2007) | 10 lines Merged revisions 73547 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r73547 | kpfleming | 2007-07-05 17:11:51 -0500 (Thu, 05 Jul 2007) | 2 lines we shouldn't allow G.723.1 endpoints to use VAD, just like we don't support it for G.729 ........ ................ 2007-07-05 22:23 +0000 [r73549] Jason Parker * apps/app_queue.c: Add the ability to play an announcement to queue caller just before bridging Issue 7479, patch by tristan_mahe. 2007-07-05 20:52 +0000 [r73513-73514] Russell Bryant * main/ast_expr2.y, main/ast_expr2.c: resolve a compiler warning so i can build in dev mode * /, res/res_features.c: Merged revisions 73512 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r73512 | russell | 2007-07-05 15:50:08 -0500 (Thu, 05 Jul 2007) | 5 lines Pass HOLD and UNHOLD frames to the other channel when they are returned from a native bridge function. This fixes a problem where when two zap channels are natively bridged and one does a flash hook, the other channel did not receive music on hold. (Reported to me directly by Doug Bailey at Digium) ........ 2007-07-05 19:20 +0000 [r73468] Joshua Colp * /, channels/chan_sip.c: Merged revisions 73467 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r73467 | file | 2007-07-05 16:18:02 -0300 (Thu, 05 Jul 2007) | 10 lines Merged revisions 73466 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r73466 | file | 2007-07-05 16:15:18 -0300 (Thu, 05 Jul 2007) | 2 lines Copy language information to the dialog structure when calling a peer for situations where a PBX may be started on the dialed channel. (issue #10121 reported by clegall_proformatique) ........ ................ 2007-07-05 18:15 +0000 [r73449] Steve Murphy * main/pbx.c, utils/expr2.testinput, main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2f.c, include/asterisk/ast_expr.h, pbx/pbx_ael.c, UPGRADE.txt, doc/tex/channelvariables.tex, utils/ael_main.c, main/ast_expr2.fl, main/ast_expr2.c, utils/check_expr.c: In regards to changes for 9508, expr2 system choking on floating point numbers, I'm adding this update to round out (no pun intended) and make this FP-capable version of the Expr2 stuff interoperate better with previous integer-only usage, by providing Functions syntax, with 20 builtin functions for floating pt to integer conversions, and some general floating point math routines that might commonly be used also. Along with this, I made it so if a function was not a builtin, it will try and find it in the ast_custom_function list, and if found, execute it and collect the results. Thus, you can call system functions like CDR(), CHANNEL(), etc, from within $\[..\] exprs, without having to wrap them in $\{...\} (curly brace) notation. Did a valgrind on the standalone and made sure there's no mem leaks. Looks good. Updated the docs, too. 2007-07-05 17:21 +0000 [r73432] Tilghman Lesher * apps/app_voicemail.c: Remove directory creation of directories we've never used. 2007-07-05 16:05 +0000 [r73402] Mark Michelson * /, apps/app_queue.c: Merged revisions 73400 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r73400 | mmichelson | 2007-07-05 10:59:41 -0500 (Thu, 05 Jul 2007) | 5 lines Correcting a minor CLI bug I found. When issuing the queue show command, if you type queue show and then press tab, you can continue pressing tab and it will keep auto-completing queue names even though only 1 queue can be used as an argument. ........ 2007-07-05 15:29 +0000 [r73399] Russell Bryant * channels/chan_vpb.cc, /, channels/Makefile: Merged revisions 73398 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r73398 | russell | 2007-07-05 10:28:27 -0500 (Thu, 05 Jul 2007) | 2 lines Make this module build for me in dev-mode ........ 2007-07-05 14:22 +0000 [r73317-73359] Joshua Colp * main/channel.c, /, apps/app_chanspy.c: Merged revisions 73355 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r73355 | file | 2007-07-05 11:21:44 -0300 (Thu, 05 Jul 2007) | 10 lines Merged revisions 73349 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r73349 | file | 2007-07-05 11:19:14 -0300 (Thu, 05 Jul 2007) | 2 lines Tweak spy locking. (issue #9951 reported by welles) ........ ................ * channels/chan_local.c, /: Merged revisions 73319 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r73319 | file | 2007-07-05 10:27:40 -0300 (Thu, 05 Jul 2007) | 10 lines Merged revisions 73318 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r73318 | file | 2007-07-05 10:26:02 -0300 (Thu, 05 Jul 2007) | 2 lines Actually check to make sure a PBX was started on one of the Local channels instead of blindly assuming it was. (issue #10112 reported by makoto) ........ ................ * /, apps/app_queue.c: Merged revisions 73316 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r73316 | file | 2007-07-05 10:22:13 -0300 (Thu, 05 Jul 2007) | 10 lines Merged revisions 73315 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r73315 | file | 2007-07-05 10:19:17 -0300 (Thu, 05 Jul 2007) | 2 lines Reset ServicelevelPerf variable back to 0 if we are unable to calculate it each time... otherwise we will get previous values. (issue #10117 reported by noriyuki) ........ ................ 2007-07-05 07:45 +0000 [r73209-73298] Christian Richter * channels/chan_misdn.c, channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample, channels/misdn_config.c: added general Jitterbuffer Implementation. #9960 * /, channels/misdn/isdn_lib.c: Merged revisions 73253 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r73253 | crichter | 2007-07-04 16:53:48 +0200 (Mi, 04 Jul 2007) | 9 lines Merged revisions 73252 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r73252 | crichter | 2007-07-04 16:50:58 +0200 (Mi, 04 Jul 2007) | 1 line bchannel configurations like echocancel and volume control, need to be setuped on inbound calls too. ........ ................ * channels/chan_misdn.c, /: Merged revisions 73208 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r73208 | crichter | 2007-07-04 10:27:44 +0200 (Mi, 04 Jul 2007) | 9 lines Merged revisions 73207 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r73207 | crichter | 2007-07-04 10:20:54 +0200 (Mi, 04 Jul 2007) | 1 line bad bug in overlapdial case, we called start_pbx multiple times, because the state wasn't changed.. ........ ................ 2007-07-03 22:17 +0000 [r73191] Steve Murphy * /: blocking 73143 (revert of 9508 bug fix for 1.4) -- don't want it backed out of trunk, too 2007-07-03 21:44 +0000 [r73144-73175] Jason Parker * apps/app_voicemail.c: mkstemp doesn't specify a file mode, so we should chmod it to VOICEMAIL_FILE_MODE Taken from a larger patch by ltd - the rest of which is no longer necessary in trunk. Closes issue #9231 * apps/app_meetme.c: Fix a build warning, and potential issue if option p is not set at all. * apps/app_meetme.c: Add support for changing the exit key from # to any DTMF. This does not break existing configs - the arguments to p are optional. Issue 8827, initial patch by junky, mostly rewritten by fw to re-use option p, further modified by me. 2007-07-03 18:25 +0000 [r73127] Russell Bryant * apps/app_queue.c: Fix up the device state processing thread in app_queue so that it's not possible for there to be entries in the queue and the thread is just sleeping (Thanks to mmichelson for bringing the problem to my attention) 2007-07-03 12:40 +0000 [r73054] Tilghman Lesher * apps/app_dial.c, /: Merged revisions 73053 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r73053 | tilghman | 2007-07-03 07:38:53 -0500 (Tue, 03 Jul 2007) | 10 lines Merged revisions 73052 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r73052 | tilghman | 2007-07-03 07:34:14 -0500 (Tue, 03 Jul 2007) | 2 lines RetryDial should accept a 0 argument, but it does not, because atoi does not distinguish between 0 and error (closes issue #10106) ........ ................ 2007-07-03 08:22 +0000 [r73006] Christian Richter * channels/chan_misdn.c, /: Merged revisions 73005 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r73005 | crichter | 2007-07-03 10:17:06 +0200 (Di, 03 Jul 2007) | 9 lines Merged revisions 73004 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r73004 | crichter | 2007-07-03 10:04:35 +0200 (Di, 03 Jul 2007) | 1 line fixed issue, that misdn_l2l1_check could only be called from mISDN Source channels.. #9449 ........ ................ 2007-07-03 05:21 +0000 [r73003] Tilghman Lesher * apps/app_voicemail.c: Typo (closes issue 10105) 2007-07-03 02:51 +0000 [r72987] Jason Parker * res/res_jabber.c: Correct an issue where the wrong type was being used to start sasl. Pointed out by and patch provided by mog. 2007-07-02 23:02 +0000 [r72982-72986] Russell Bryant * main/pbx.c, doc/tex/ast_funcdocs.tex (removed), main/manager.c, doc/tex/ast_cli_commands.tex (removed), res/res_agi.c, doc/tex/ast_appdocs.tex (removed), doc/tex/asterisk.tex, doc/tex/ast_manager_actiondocs.tex (removed), doc/tex/ast_agi_commands.tex (removed), main/cli.c: After some discussion on the asterisk-dev list, we determined that this approach for extracting application, function, manager, and agi documentation is the wrong one to take. The most severe problem is that the output depends on which modules are loaded as well as compile time options, which both determine which parts are available. * doc/jitterbuffer.tex (removed), doc/extensions.tex (removed), doc/tex/ast_cli_commands.tex (added), doc/tex/ast_appdocs.tex (added), doc/tex/realtime.tex (added), doc/qos.tex (removed), doc/queues-with-callback-members.tex (removed), doc/tex/dundi.tex (added), doc/ajam.tex (removed), doc/tex/cliprompt.tex (added), doc/misdn.tex (removed), doc/manager.tex (removed), doc/tex/chaniax.tex (added), doc/sla.tex (removed), doc/billing.tex (removed), doc/tex/app-sms.tex (added), build_tools/prep_tarball, doc/tex/ices.tex (added), doc/localchannel.tex (removed), doc/cdrdriver.tex (removed), doc/tex/asterisk.tex (added), doc/tex/queuelog.tex (added), doc/freetds.tex (removed), doc/odbcstorage.tex (removed), doc/tex/hardware.tex (added), doc/tex/mp3.tex (added), doc/tex (added), doc/channelvariables.tex (removed), doc/ael.tex (removed), doc/enum.tex (removed), doc/tex/configuration.tex (added), doc/security.tex (removed), doc/tex/asterisk-conf.tex (added), Makefile, doc/imapstorage.tex (removed), doc/tex/ast_funcdocs.tex (added), doc/privacy.tex (removed), doc/tex/ast_manager_actiondocs.tex (added), doc/ast_agi_commands.tex (removed), doc/tex/jitterbuffer.tex (added), doc/ast_cli_commands.tex (removed), doc/tex/extensions.tex (added), doc/ast_appdocs.tex (removed), doc/tex/queues-with-callback-members.tex (added), doc/tex/qos.tex (added), doc/realtime.tex (removed), doc/dundi.tex (removed), doc/tex/ajam.tex (added), doc/cliprompt.tex (removed), doc/tex/manager.tex (added), doc/tex/misdn.tex (added), doc/chaniax.tex (removed), doc/tex/README.txt (added), doc/tex/sla.tex (added), doc/app-sms.tex (removed), doc/tex/billing.tex (added), doc/ices.tex (removed), doc/tex/localchannel.tex (added), doc/tex/cdrdriver.tex (added), doc/asterisk.tex (removed), doc/queuelog.tex (removed), doc/tex/odbcstorage.tex (added), doc/tex/freetds.tex (added), doc/hardware.tex (removed), doc/mp3.tex (removed), doc/tex/channelvariables.tex (added), doc/tex/ael.tex (added), doc/tex/enum.tex (added), doc/configuration.tex (removed), doc/tex/security.tex (added), doc/asterisk-conf.tex (removed), doc/tex/imapstorage.tex (added), doc/ast_funcdocs.tex (removed), doc/tex/privacy.tex (added), doc/tex/Makefile (added), doc/ast_manager_actiondocs.tex (removed), doc/tex/ast_agi_commands.tex (added): * Move LaTeX docs into a tex/ subdirectory of the doc/ dir * Add a Makefile in doc/tex/ for generating PDF and HTML * Add a README.txt file to doc/tex/ to document which tools are used and what web sites to visit for getting them. * Update build_tools/prep_tarball to put the proper Asterisk version string in the automatically generated PDF for release tarballs 2007-07-02 21:50 +0000 [r72940] Steve Murphy * utils/expr2.testinput, /, main/Makefile, main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2f.c, UPGRADE.txt, main/ast_expr2.fl, main/ast_expr2.c: Merged revisions 72933 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r72933 | murf | 2007-07-02 14:16:31 -0600 (Mon, 02 Jul 2007) | 1 line support for floating point numbers added to ast_expr2 $\[...\] exprs. Fixes bug 9508, where the expr code fails with fp numbers. The MATH function returns fp numbers by default, so this fix is considered necessary. ........ 2007-07-02 20:45 +0000 [r72937-72939] Russell Bryant * res/res_agi.c, doc/ast_agi_commands.tex: Fix up the AGI doc dump CLI command and update the AGI commands tex file to not include a bunch of empty entries. * doc/ast_cli_commands.tex (added), doc/asterisk.tex: Add CLI commands to the docs * main/cli.c: Add a CLI command to output docs on CLI commands to a file 2007-07-02 20:35 +0000 [r72935-72936] Joshua Colp * channels/chan_iax2.c: Yet another Solaris tweak... * res/res_limit.c: Fix building under Solaris. 2007-07-02 19:31 +0000 [r72920-72932] Russell Bryant * doc/asterisk.tex, doc/ast_agi_commands.tex (added): Add AGI commands to the documentation * res/res_agi.c: Add a CLI command to export the AGI command docs * res/res_agi.c: Add a note that the AGI commands array is not handled in a thread-safe way * doc/asterisk.tex, doc/ast_manager_actiondocs.tex (added): Update the documentation to include a manager action reference * main/manager.c: Add a CLI command to dump the built-in manager action documentation * main/manager.c, /: Merged revisions 72926 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r72926 | russell | 2007-07-02 13:18:46 -0500 (Mon, 02 Jul 2007) | 3 lines Remove a bogus comment and add proper locking to the handler function for the CLI command to show information on manager actions. ........ * doc/ast_funcdocs.tex (added), doc/asterisk.tex: update documentation to include dialplan functions * main/pbx.c: Add "core dump funcdocs" CLI command * main/pbx.c: change the "core dump appdocs" CLI command to use the new API for creating CLI commands * doc/ast_appdocs.tex: update application documentation dump 2007-07-02 14:39 +0000 [r72889] Joshua Colp * main/channel.c, /: Merged revisions 72888 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r72888 | file | 2007-07-02 11:32:59 -0300 (Mon, 02 Jul 2007) | 2 lines Added additional DTMF debug messages for when emulation occurs. ........ 2007-07-02 09:34 +0000 [r72867-72869] Christian Richter * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged revisions 72852 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r72852 | crichter | 2007-07-02 10:41:08 +0200 (Mo, 02 Jul 2007) | 9 lines Merged revisions 72585 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r72585 | crichter | 2007-06-29 15:08:26 +0200 (Fr, 29 Jun 2007) | 1 line check if the bchannel stack id is already used, if so don't use it a second time. Also added a release_chan lock, so that the same chan_list object cannot be freed twice. chan_misdn does not crash anymore on heavy load with these changes. ........ ................ * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c: Merged revisions 72851 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r72851 | crichter | 2007-07-02 10:27:19 +0200 (Mo, 02 Jul 2007) | 9 lines Merged revisions 72099 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r72099 | crichter | 2007-06-27 15:22:37 +0200 (Mi, 27 Jun 2007) | 1 line simplified generation for dummy bchannels, also we mark them as dummies, so they are not used later as real-bchannels, optimized the RESTART mechanisms, we block a channel now on cause:44, and send out a RESTART automatically, then on reception of RESTART_ACKNOWLEDGE we unblock the channel again. ........ ................ * channels/misdn/isdn_lib.h, /, channels/misdn/isdn_lib.c: Merged revisions 72850 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r72850 | crichter | 2007-07-02 10:14:43 +0200 (Mo, 02 Jul 2007) | 9 lines Merged revisions 72087 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r72087 | crichter | 2007-06-27 11:26:53 +0200 (Mi, 27 Jun 2007) | 1 line simplified channel finding and locking a lot. removed unnecessary #ifdefed areas. ........ ................ 2007-07-01 23:53 +0000 [r72807] Russell Bryant * pbx/pbx_spool.c, /: Merged revisions 72806 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r72806 | russell | 2007-07-01 18:52:45 -0500 (Sun, 01 Jul 2007) | 13 lines Merged revisions 72805 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r72805 | russell | 2007-07-01 18:51:34 -0500 (Sun, 01 Jul 2007) | 5 lines When appending lines to call files to keep track of retries, write a leading newline just in case the original call file did not have a newline at the end. This fix is in response to a problem I saw reported on the asterisk-users mailing list. ........ ................ 2007-06-30 16:53 +0000 [r72767] Russell Bryant * /, configure, configure.ac: Merged revisions 72766 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r72766 | russell | 2007-06-30 11:50:40 -0500 (Sat, 30 Jun 2007) | 3 lines Tweak the configure script so that error output isn't spewed to the console when searching for GTK2 libs, and they aren't found. ........ 2007-06-29 21:37 +0000 [r72741] Jason Parker * channels/chan_skinny.c, configs/skinny.conf.sample: Add support for regcontext and regexten to chan_skinny Issue 9762, patch by mvanbaak. 2007-06-29 21:24 +0000 [r72738] Russell Bryant * configure, include/asterisk/autoconfig.h.in, configure.ac, main/http.c: Fix my recent change for sending large files via the http server. This code *must* write the file to the FILE *, and not the raw fd. Otherwise, it breaks TLS support. Thanks to rizzo for catching this! 2007-06-29 21:14 +0000 [r72727] Luigi Rizzo * main/minimime/Makefile: As the comment in the code says: Use weaker error checking because we have some automatically generated files. However just mask out -Werror, because other warnings below: -Wundef -Wstrict-prototypes -Wmissing-declarations -Wmissing-prototypes may actually be important and spot out real bugs. 2007-06-29 20:56 +0000 [r72701-72706] Russell Bryant * /, formats/format_pcm.c: Merged revisions 72705 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r72705 | russell | 2007-06-29 15:56:18 -0500 (Fri, 29 Jun 2007) | 1 line give format_pcm a more concise destription ........ * include/asterisk/http.h, main/manager.c, configure, include/asterisk/autoconfig.h.in, configure.ac, main/http.c: Merge changes from team/russell/http_filetxfer Handle transferring large files from the built-in http server. Previously, the code attempted to malloc a block as large as the file itself. Now it uses the sendfile() system call so that the file isn't copied into userspace at all if it is available. Otherwise, it just uses a read/write of small chunks at a time. 2007-06-29 20:33 +0000 [r72700] Luigi Rizzo * main/Makefile: Make sure that we properly recurse in subdirectories to check dependencies for libraries. Because these targets (e.g. minimime/libmmime.a) are real ones, declaring them .PHONY would cause them to be rebuilt every time (see e.g. SVN 64355). As a workaround I am using the following CHECK_SUBDIR target: CHECK_SUBDIR: # do nothing, just make sure that we recurse in the subdir/ minimime/libmmime.a: CHECK_SUBDIR @cd minimime && $(MAKE) libmmime.a which seems to do a better job than .PHONY (probably because .PHONY forces the rebuild even if the recursive make does not think it is necessary). If this turns out to be the correct approach, we can then merge it back into 1.4 2007-06-29 20:02 +0000 [r72670] Mark Michelson * apps/app_voicemail.c: Found a grievous logical error in get_vm_state_by_imapuser. The imapuser being passed in was never getting compared to imapusers of any of the vm_states in the vmstates list. I also found some places in the code where I used my typical brace style and changed it to match the typical Asterisk brace style. 2007-06-29 19:09 +0000 [r72666] Luigi Rizzo * /: 72665 not applicable to trunk 2007-06-29 04:56 +0000 [r72555-72557] Tilghman Lesher * main/manager.c, /: Merged revisions 72556 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r72556 | tilghman | 2007-06-28 23:47:11 -0500 (Thu, 28 Jun 2007) | 2 lines Issue 10055 - Change memory allocation to use the heap for a command, since the output has the potential to overflow the stack (as it did here) ........ 2007-06-28 21:31 +0000 [r72539] Jason Parker * Makefile, configure, configure.ac, makeopts.in: Apparently some builds of gcc don't have declaration-after-statement. This checks for it in configure, and only uses it if it's available. If it's wrong, somebody please yell at me and tell me why. 2007-06-28 20:52 +0000 [r72524] Dwayne M. Hubbard * funcs/func_math.c: Added AND, OR, and XOR bitwise operations to MATH for issue 9891, thanks jcmoore 2007-06-28 19:41 +0000 [r72492] Tilghman Lesher * res/res_config_pgsql.c, res/res_config_odbc.c, include/asterisk/strings.h: Remove the ill-advised ast_restrdupa API call and related structures 2007-06-28 19:35 +0000 [r72490-72491] Jason Parker * channels/chan_sip.c: Fix building with -Wdeclaration-after-statement, here too * res/res_jabber.c: Fix building with -Wdeclaration-after-statement 2007-06-28 19:07 +0000 [r72452-72466] Luigi Rizzo * /: 72462 is not applicable to trunk * res/res_features.c, apps/app_sms.c: move variable declarations to the beginning of a block. Not applicable to previous branches. * channels/chan_skinny.c: move variable declarations to the beginning of the block * apps/app_minivm.c: move variable declarations to the beginning of a block. Not applicable to previous branches * /: 72453 was already applied to trunk some time ago * Makefile: Add -Wdeclaration-after-statement to AST_DEVMODE to detect declarations in the middle of a block. Approved by: Russel, Kevin The fallout will be fixed in separate commits. I am doing this only on trunk only for the time being, because 1.4 still requires a bit more polishing to give a clean compile (at least on FreeBSD). 2007-06-28 16:35 +0000 [r72437] Matthew Fredrickson * channels/chan_zap.c: Fix bug where point code gets corrupted on CPG 2007-06-27 23:30 +0000 [r72384] Brett Bryant * /, main/asterisk.c: Merged revisions 72383 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r72383 | bbryant | 2007-06-27 18:29:14 -0500 (Wed, 27 Jun 2007) | 11 lines Merged revisions 72373 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r72373 | bbryant | 2007-06-27 18:22:13 -0500 (Wed, 27 Jun 2007) | 3 lines Reinstating patch. This actually fixes the problem, however I was running a development branch without it and mistakenly thought it wasn't fixed. Fixes issue #10010, and #9654: 100% CPU usage caused by an asterisk console losing it's controlling terminal. ........ ................ 2007-06-27 23:26 +0000 [r72354-72382] Joshua Colp * /, apps/app_mixmonitor.c: Merged revisions 72381 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r72381 | file | 2007-06-27 19:25:12 -0400 (Wed, 27 Jun 2007) | 10 lines Merged revisions 72378 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r72378 | file | 2007-06-27 19:24:01 -0400 (Wed, 27 Jun 2007) | 2 lines Update documentation to clarify variable usage with MixMonitor. (issue #9494 reported by netoguy) ........ ................ * channels/chan_jingle.c: Silly jingle... * channels/chan_sip.c, CHANGES: Add SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables when a transfer takes place. (issue #8378 reported by jcovert) 2007-06-27 23:04 +0000 [r72337] Brett Bryant * /, main/asterisk.c: Merged revisions 72335 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r72335 | bbryant | 2007-06-27 18:03:01 -0500 (Wed, 27 Jun 2007) | 10 lines Merged revisions 72333 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r72333 | bbryant | 2007-06-27 17:58:53 -0500 (Wed, 27 Jun 2007) | 2 lines Reverted changes for earlier revisions 72259 to 72261. Issue #9654, #10010 ........ ................ 2007-06-27 22:58 +0000 [r72330-72332] Joshua Colp * /, channels/chan_gtalk.c: Merged revisions 72331 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r72331 | file | 2007-06-27 18:58:02 -0400 (Wed, 27 Jun 2007) | 2 lines Make payload IDs for iLBC/Speex match to our list. Since these are dynamic payloads the other side shouldn't care. (issue #9426 reported by irroot) ........ * /, apps/app_queue.c: Merged revisions 72328 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r72328 | file | 2007-06-27 18:45:49 -0400 (Wed, 27 Jun 2007) | 10 lines Merged revisions 72327 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r72327 | file | 2007-06-27 18:43:11 -0400 (Wed, 27 Jun 2007) | 2 lines Fix issue where queue log events might be missing. (issue #7765 reported by mtryfoss) ........ ................ 2007-06-27 22:47 +0000 [r72329] Mark Michelson * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: Added ability to customize which buttons control forward, reverse, pause, and stop during message playback. (closes issue 9474, reported and patched by jaroth with modifications by me) 2007-06-27 22:27 +0000 [r72325-72326] Jason Parker * main/cli.c: Fix a segfault when trying to tab complete the "core show uptime" command. Reported in #asterisk-dev on IRC by jcmoore, fixed by me. * main/say.c: Add support for Thai language in say.c Issue 9417, patch by dome, with some cleanup done by me. 2007-06-27 21:44 +0000 [r72304] Matthew Fredrickson * channels/chan_zap.c: Let's NOT create a deadlock scenario here 2007-06-27 21:09 +0000 [r72274] Russell Bryant * /, pbx/pbx_config.c: Merged revisions 72272 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r72272 | russell | 2007-06-27 16:08:34 -0500 (Wed, 27 Jun 2007) | 13 lines Merged revisions 72267 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r72267 | russell | 2007-06-27 16:06:45 -0500 (Wed, 27 Jun 2007) | 5 lines Fix a minor issue with parsing the priority number. You could have as much whitespace as you want around a numeric priority, but you couldn't have any whitespace around a special priority like "n" or "hint". (issue #10039, reported by mitheloc, fixed by me) ........ ................ 2007-06-27 20:47 +0000 [r72261] Brett Bryant * /, main/asterisk.c: Merged revisions 72260 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r72260 | bbryant | 2007-06-27 15:46:12 -0500 (Wed, 27 Jun 2007) | 12 lines Merged revisions 72259 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r72259 | bbryant | 2007-06-27 15:43:53 -0500 (Wed, 27 Jun 2007) | 4 lines Fixes 100% load when controlling terminal disappears. Issue #9654, #10010 ........ ................ 2007-06-27 20:26 +0000 [r72233-72258] Joshua Colp * main/channel.c, /: Merged revisions 72257 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r72257 | file | 2007-06-27 16:25:24 -0400 (Wed, 27 Jun 2007) | 10 lines Merged revisions 72256 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r72256 | file | 2007-06-27 16:23:24 -0400 (Wed, 27 Jun 2007) | 2 lines I may possibly get shot for doing this... but... defer CDR processing until after the channel has been dealt with. This should eliminate all of the issues with channels going funky (SIP/PRI) when you are posting CDRs to a database that is either slow or unavailable and do not want to enable batching. ........ ................ * /: Fix up properties. * main/logger.c: Fix -T option. (issue #10073 reported by xylome) 2007-06-27 19:50 +0000 [r72232] Mark Michelson * /, configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: Adding feature to support the storage and retrieval of voicemail greetings using IMAP storage. This feature may be turned on by adding imapgreetings=yes to the general section of voicemail.conf voicemail.conf.sample has details on the options added. As a result, IMAP storage now has RETRIEVE and DISPOSE macros defined. In addition to the IMAP greeting changes, I also have added an enum for the voicemail folders and so now the code should be easier to understand and maintain when it comes to this area. 2007-06-27 19:13 +0000 [r72207] Kevin P. Fleming * channels/chan_zap.c, /: Merged revisions 72205 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r72205 | kpfleming | 2007-06-27 14:13:21 -0500 (Wed, 27 Jun 2007) | 2 lines use the proper type for storing group number bits so that if someone specifies 'group=42' it will actually work instead of being silently ignored ........ 2007-06-27 18:37 +0000 [r72183] Jason Parker * /, apps/app_voicemail.c: Merged revisions 72182 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r72182 | qwell | 2007-06-27 13:36:56 -0500 (Wed, 27 Jun 2007) | 4 lines Fix another problem in voicemail with missing symbols. Issue 10074, patch by kryptolus, extended to include #if 0'd blocks (just in case) ........ 2007-06-27 17:34 +0000 [r72149] Joshua Colp * main/channel.c, /: Merged revisions 72148 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r72148 | file | 2007-06-27 13:31:50 -0400 (Wed, 27 Jun 2007) | 2 lines Make the ast_read_noaudio API call behave better under circumstances where DTMF emulation was happening and a generator was setup. (issue #10065 reported by stevefeinstein) ........ 2007-06-27 17:14 +0000 [r72134] Jason Parker * /, channels/chan_gtalk.c: Merged revisions 72125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r72125 | qwell | 2007-06-27 12:10:32 -0500 (Wed, 27 Jun 2007) | 4 lines Don't modify a variable that we don't want modified. Make a copy of it instead. Issue 10029, patch by phsultan with slight modifications by me (to remove needless casts). Note: chan_jingle in trunk does not appear to have the same bug. ........ 2007-06-27 16:38 +0000 [r72113] Russell Bryant * /, main/rtp.c: Merged revisions 72112 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r72112 | russell | 2007-06-27 11:34:24 -0500 (Wed, 27 Jun 2007) | 3 lines Only output debug information related to RTCP timestamps when RTCP debug is turned on (issue #10066, patch by me) ........ 2007-06-27 08:08 +0000 [r72052] Christian Richter * /, channels/misdn/isdn_lib.c: Merged revisions 72042 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r72042 | crichter | 2007-06-27 09:58:06 +0200 (Mi, 27 Jun 2007) | 13 lines Merged revisions 72040-72041 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r72040 | crichter | 2007-06-27 09:49:27 +0200 (Mi, 27 Jun 2007) | 1 line for inbound TE calls, we setup the bchannel when we get the CONNECT_ACKNOWLEDGE, to make sure mISDN has everything ready. removed some #if 0 areas which weren't used anymore. ........ r72041 | crichter | 2007-06-27 09:54:30 +0200 (Mi, 27 Jun 2007) | 1 line isdn_lib.c didn't compile ........ ................ 2007-06-27 01:00 +0000 [r71988-72007] Joshua Colp * /, pbx/pbx_dundi.c: Merged revisions 72006 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r72006 | file | 2007-06-26 20:58:35 -0400 (Tue, 26 Jun 2007) | 2 lines Make unloading of pbx_dundi actually work. ........ * channels/chan_sip.c, funcs/func_channel.c, CHANGES: Add rtpdest option to SIP CHANNEL() dialplan function to return the IP address and port that RTP (be it audio/video/text) is going to. 2007-06-26 23:03 +0000 [r71952-71954] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 71953 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r71953 | mmichelson | 2007-06-26 18:02:09 -0500 (Tue, 26 Jun 2007) | 4 lines Removing a pointless line. This variable was already set earlier and between then and this line, there is no way that the values on the right side of the assignment could have changed. ........ * apps/app_voicemail.c: The variable msgnum was being overwritten if IMAP storage was enabled. Put necessary #ifndef's around the line which would overwrite. 2007-06-26 20:36 +0000 [r71916] Jason Parker * /, main/rtp.c: Merged revisions 71915 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r71915 | qwell | 2007-06-26 15:36:09 -0500 (Tue, 26 Jun 2007) | 4 lines Don't dereference a pointer that may be NULL here. Issue 10017. ........ 2007-06-26 20:34 +0000 [r71883-71914] Mark Michelson * apps/app_record.c: Create directory if it does not exist. (Closes issue 10061, Reported and patched by eliel) * /, apps/app_voicemail.c: Merged revisions 71877 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r71877 | mmichelson | 2007-06-26 14:00:05 -0500 (Tue, 26 Jun 2007) | 11 lines A few changes, the ultimate goal of which is to keep better track of the number of messages that a mailbox currently has. A description of the changes: 1. Changed the "updated" field of the vm_state struct to act more as a binary semaphore than a counting semaphore, since its current implementation made the inboxcount function not work properly. This change falls in line with a change made by UPenn with their IMAP setup and helps to sync our changes with theirs. 2. Eliminated some redundant calls to get_vm_state_by_mailbox inside leave_voicemail 3. Use the play_folder variable to keep track of the number of old and new messages in a mailbox as the messages are deleted 4. Added an increment to the number of new messages that was not there previously in the leave_voicemail function ........ 2007-06-26 16:39 +0000 [r71830] Jason Parker * res/res_jabber.c: Simplify some code in res_jabber relating to SASL support. Issue 9988, patch by phsultan. 2007-06-26 15:50 +0000 [r71797] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 71796 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r71796 | mmichelson | 2007-06-26 10:47:31 -0500 (Tue, 26 Jun 2007) | 5 lines Fixing bug where the authuser was mistakenly pulled from the mailbox string instead of the IMAP user. (closes issue 10054, reported and patched by jaroth) ........ 2007-06-26 12:30 +0000 [r71752] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 71751 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r71751 | tilghman | 2007-06-26 07:27:47 -0500 (Tue, 26 Jun 2007) | 10 lines Merged revisions 71750 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r71750 | tilghman | 2007-06-26 07:25:58 -0500 (Tue, 26 Jun 2007) | 2 lines Issue 10062 - Trying to move a message without selecting one first results in memory corruption ........ ................ 2007-06-26 00:10 +0000 [r71721-71732] Mark Michelson * configure, configure.ac: Fixes a problem where Asterisk would not compile if IMAP_STORAGE was enabled. Needed to add a space between file name and options. * apps/app_voicemail.c: In my commit earlier today, I accidentally left a prototype that isn't defined. This gets rid of that prototype. 2007-06-25 19:20 +0000 [r71688] Russell Bryant * doc/imapstorage.tex, configure, configure.ac, apps/app_voicemail.c: Allow compilation off app_voicemail with IMAP_STORAE against a system installed version of the c-client library. (issue #10047, jcollie) 2007-06-25 18:20 +0000 [r71658] Tilghman Lesher * /, res/res_agi.c: Merged revisions 71657 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r71657 | tilghman | 2007-06-25 13:14:59 -0500 (Mon, 25 Jun 2007) | 10 lines Merged revisions 71656 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r71656 | tilghman | 2007-06-25 13:12:37 -0500 (Mon, 25 Jun 2007) | 2 lines Issue 10035 - handle_exec returns a result inconsistent with all of the other AGI commands ........ ................ 2007-06-25 16:43 +0000 [r71637] Steve Murphy * main/cdr.c: Luigi's suggestion to move the llfrom decl was a good one. Done. 2007-06-25 16:13 +0000 [r71630] Mark Michelson * apps/app_voicemail.c: Using inboxcount instead of countmessages. 2007-06-25 15:35 +0000 [r71577-71613] Joshua Colp * channels/chan_sip.c: Tweak CLI command completion and some help text. (issue #10049 reported by IgorG) * /, channels/chan_h323.c: Merged revisions 71576 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r71576 | file | 2007-06-25 10:13:45 -0400 (Mon, 25 Jun 2007) | 2 lines Build a peer as well when hash323 is enabled in users.conf (issue #9599 reported by asagage) ........ 2007-06-25 13:42 +0000 [r71557] Russell Bryant * main/say.c, main/rtp.c, main/sched.c: Convert so more logging to ast_debug (issue #10045, dimas) 2007-06-25 13:04 +0000 [r71521-71525] Joshua Colp * /, channels/chan_agent.c: Merged revisions 71522 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r71522 | file | 2007-06-25 09:03:03 -0400 (Mon, 25 Jun 2007) | 2 lines Minor tweak for queueing up the unhold frame... this will teach me to do bugs while half asleep. (issue #10046 reported by dimas) ........ * res/res_agi.c: Minor header inclusion tweak for new usage of stat() 2007-06-25 12:40 +0000 [r71520] Russell Bryant * doc/asterisk-mib.txt, /: Merged revisions 71519 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r71519 | russell | 2007-06-25 07:40:06 -0500 (Mon, 25 Jun 2007) | 2 lines Fix a typo in the Asterisk mib. (issue #10048, Matti) ........ 2007-06-25 09:46 +0000 [r71475-71500] Christian Richter * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged revisions 71214 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r71214 | crichter | 2007-06-23 00:44:42 +0200 (Sa, 23 Jun 2007) | 9 lines Merged revisions 70341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r70341 | crichter | 2007-06-20 17:29:09 +0200 (Mi, 20 Jun 2007) | 1 line fixed a bug that was introduced by copy and paste in the last commit ..bchannels weren't cleaned properly. ........ ................ * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged revisions 71123 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r71123 | crichter | 2007-06-22 17:38:08 +0200 (Fr, 22 Jun 2007) | 9 lines Merged revisions 70672 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r70672 | crichter | 2007-06-21 15:11:29 +0200 (Do, 21 Jun 2007) | 1 line we activate the bchannels in TE mode on incoming calls only when we want to connect the call. ........ ................ * /, channels/misdn/isdn_lib.c: Merged revisions 71122 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r71122 | crichter | 2007-06-22 17:34:31 +0200 (Fr, 22 Jun 2007) | 9 lines Merged revisions 70342 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r70342 | crichter | 2007-06-20 17:42:39 +0200 (Mi, 20 Jun 2007) | 1 line forgot one place .. ........ ................ * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged revisions 71121 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r71121 | crichter | 2007-06-22 17:32:54 +0200 (Fr, 22 Jun 2007) | 9 lines Merged revisions 70311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r70311 | crichter | 2007-06-20 16:47:59 +0200 (Mi, 20 Jun 2007) | 1 line on receiption of cause:44 we mark the channel as in use and inform the user about the situation, we need to test the RESTART stuff then. Also shuffled the empty_chan_in_stack function after the bchannel cleaning functions, to avoid race conditions. ........ ................ * channels/chan_misdn.c, /: Merged revisions 71120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r71120 | crichter | 2007-06-22 17:30:08 +0200 (Fr, 22 Jun 2007) | 9 lines Merged revisions 69887 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r69887 | crichter | 2007-06-19 15:23:04 +0200 (Di, 19 Jun 2007) | 1 line when we send out a SETUP, but get no response, we should cleanup everything after reception of a hangup. ........ ................ * /, channels/misdn/isdn_msg_parser.c: Merged revisions 71118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r71118 | crichter | 2007-06-22 17:27:53 +0200 (Fr, 22 Jun 2007) | 9 lines Merged revisions 69053 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r69053 | crichter | 2007-06-13 11:55:54 +0200 (Mi, 13 Jun 2007) | 1 line restart indicator 0x80 is correct, at least that's what libpri does. ........ ................ * channels/chan_misdn.c, /: Merged revisions 71106 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r71106 | crichter | 2007-06-22 17:22:06 +0200 (Fr, 22 Jun 2007) | 9 lines Merged revisions 68887 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r68887 | crichter | 2007-06-12 10:35:22 +0200 (Di, 12 Jun 2007) | 1 line if the bridged partner is mISDN too we should not send dtmf tones, they are transmitted inband always ........ ................ * channels/chan_misdn.c, /: Merged revisions 71096 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r71096 | crichter | 2007-06-22 17:17:04 +0200 (Fr, 22 Jun 2007) | 9 lines Merged revisions 68874 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r68874 | crichter | 2007-06-12 09:48:52 +0200 (Di, 12 Jun 2007) | 1 line if we have already some digits, we just stop the tones. ........ ................ 2007-06-25 01:11 +0000 [r71413-71434] Joshua Colp * /, channels/chan_sip.c: Merged revisions 71430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r71430 | file | 2007-06-24 21:10:06 -0400 (Sun, 24 Jun 2007) | 10 lines Merged revisions 71414 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r71414 | file | 2007-06-24 21:02:49 -0400 (Sun, 24 Jun 2007) | 2 lines Ignore other URIs after the first in a 300 Multiple Choice response. (issue #10041 reported by homesick) ........ ................ * main/cdr.c, /: Merged revisions 71422 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r71422 | file | 2007-06-24 21:07:31 -0400 (Sun, 24 Jun 2007) | 2 lines Fix it so 1.4 actually compiles on my box. ........ * /, channels/chan_agent.c: Merged revisions 71412 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r71412 | file | 2007-06-24 20:49:21 -0400 (Sun, 24 Jun 2007) | 2 lines Check to make sure the channel pointer is present before queueing up an unhold frame on it. (issue #10046 reported by dimas) ........ 2007-06-24 20:17 +0000 [r71338-71372] Russell Bryant * /, build_tools/prep_tarball: Merged revisions 71371 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r71371 | russell | 2007-06-24 15:16:32 -0500 (Sun, 24 Jun 2007) | 3 lines Include the menuselect-tree file in tarballs to make builds from tarballs a little bit faster ........ * /, main/asterisk.c: Merged revisions 71362 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r71362 | russell | 2007-06-24 15:06:31 -0500 (Sun, 24 Jun 2007) | 10 lines Merged revisions 71358 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r71358 | russell | 2007-06-24 15:04:21 -0500 (Sun, 24 Jun 2007) | 2 lines Revert the patch from issue 9654 due to an unexpected side effect ........ ................ * main/udptl.c, apps/app_meetme.c, main/say.c, main/translate.c, main/jitterbuf.c, apps/app_test.c, main/rtp.c, main/loader.c, main/io.c, main/manager.c, apps/app_skel.c, apps/app_minivm.c, main/logger.c, main/http.c, apps/app_rpt.c, main/sched.c: Conversions to ast_debug() (issue #9984, patches from eliel and dimas) 2007-06-24 17:51 +0000 [r71268-71292] Tilghman Lesher * /, res/res_features.c: Merged revisions 71291 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r71291 | tilghman | 2007-06-24 12:50:24 -0500 (Sun, 24 Jun 2007) | 2 lines Issue 10044 - chan->cdr is NULL here, so peer->cdr is what we really wanted to use ........ * main/manager.c, /, main/db.c: Merged revisions 71289 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r71289 | tilghman | 2007-06-24 12:39:34 -0500 (Sun, 24 Jun 2007) | 10 lines Merged revisions 71288 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r71288 | tilghman | 2007-06-24 12:32:21 -0500 (Sun, 24 Jun 2007) | 2 lines Issue 10043 - There is a legitimate need to be able to set variables to the empty string. ........ ................ * apps/app_mixmonitor.c: Issue 9970 - Ensure directory exists before trying to write an output file 2007-06-23 03:32 +0000 [r71231] Steve Murphy * main/cdr.c, /, res/res_features.c: Merged revisions 71230 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r71230 | murf | 2007-06-22 21:29:48 -0600 (Fri, 22 Jun 2007) | 1 line This patch is meant to fix 8433; where clid and src are lost via bridging. ........ 2007-06-22 19:53 +0000 [r71190] Tilghman Lesher * apps/app_sms.c: Code cleanups 2007-06-22 16:19 +0000 [r71146-71158] Joshua Colp * res/res_agi.c: Use stat to determine whether the file exists or not. (issue #10038 reported by Mike Anikienko) * main/rtp.c: Behold the magic of casting! 2007-06-22 15:15 +0000 [r71093] Steve Murphy * main/cdr.c, /, main/rtp.c: Merged revisions 71063 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r71063 | murf | 2007-06-22 08:10:24 -0600 (Fri, 22 Jun 2007) | 1 line My conditions for merging amaflags info was naive; DOCUMENTATION is the default, although null is possible; theft of user-settable fields is not good. Just copy them, leave them alone. This is for bug 10016. (plus a small fix to rtp, to elim a compiler warning (dev mode)) ........ 2007-06-22 15:03 +0000 [r71069] Jason Parker * /, res/res_agi.c, main/file.c, apps/app_speech_utils.c: Merged revisions 71068 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r71068 | qwell | 2007-06-22 10:00:30 -0500 (Fri, 22 Jun 2007) | 12 lines Merged revisions 71065 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r71065 | qwell | 2007-06-22 09:52:18 -0500 (Fri, 22 Jun 2007) | 4 lines Fix a few silly usages of ast_playstream() - it only ever returns 0... Issue 10035 ........ ................ 2007-06-22 14:56 +0000 [r71067] Brett Bryant * /, main/asterisk.c: Merged revisions 71066 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r71066 | bbryant | 2007-06-22 09:53:08 -0500 (Fri, 22 Jun 2007) | 18 lines Merged revisions 71064 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r71064 | bbryant | 2007-06-22 09:39:34 -0500 (Fri, 22 Jun 2007) | 10 lines Fixed infinite loop when controlling terminal was lost and return value of input function wasn't checked for errors. This would cause 100% cpu to be taken up. (closes issue #9654, issue #10010) Reported by: mnicholson, and eserra Idea for the patch from mnicholson, patched by me ........ ................ 2007-06-22 04:35 +0000 [r71040] Tilghman Lesher * apps/app_dial.c, include/asterisk/utils.h, pbx/pbx_spool.c, apps/app_dictate.c, apps/app_minivm.c, apps/app_test.c, main/logger.c, main/utils.c, apps/app_sms.c, res/res_monitor.c, apps/app_voicemail.c: Issue 9990 - New API ast_mkdir, which creates parent directories as necessary (and is faster than an outcall to mkdir -p) 2007-06-22 04:13 +0000 [r71024] Jason Parker * build_tools/cflags.xml, main/asterisk.c: Nothing to see here. 2007-06-22 03:15 +0000 [r71004] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 71003 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r71003 | russell | 2007-06-21 22:14:41 -0500 (Thu, 21 Jun 2007) | 3 lines Fix a small typo which ... well ... completely broke chan_iax2. oops! (issue #9937, patch by me) ........ 2007-06-21 23:07 +0000 [r70961] Jason Parker * main/manager.c, configs/manager.conf.sample, include/asterisk/manager.h, main/rtp.c: Add manager events for RTCP statistics. Also adds a new "reporting" permission for manager, since it can be incredibly spammy. This permission was discussed on the -dev mailing list some months back. Issue 8613, patch by johann8384, with some minor changes by me. 2007-06-21 22:41 +0000 [r70951] Steve Murphy * main/cdr.c, /: Merged revisions 70949 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r70949 | murf | 2007-06-21 16:34:41 -0600 (Thu, 21 Jun 2007) | 9 lines Merged revisions 70948 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r70948 | murf | 2007-06-21 16:29:50 -0600 (Thu, 21 Jun 2007) | 1 line This little fix is in response to bug 10016, but may not cure it. The code is wrong, clearly. In a situation where you set the CDR's amaflags, and then ForkCDR, and then set the new CDR's amaflags to some other value, you will see that all CDRs have had their amaflags changed. This is not good. So I fixed it. ........ ................ 2007-06-21 21:41 +0000 [r70900] Joshua Colp * /, apps/app_voicemail.c: Merged revisions 70899 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r70899 | file | 2007-06-21 17:40:19 -0400 (Thu, 21 Jun 2007) | 10 lines Merged revisions 70898 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r70898 | file | 2007-06-21 17:37:55 -0400 (Thu, 21 Jun 2007) | 2 lines Don't explode if the gain option is specified without a value. (issue #9274 reported by mfarver) ........ ................ 2007-06-21 21:16 +0000 [r70877-70887] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 70883 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r70883 | russell | 2007-06-21 16:14:53 -0500 (Thu, 21 Jun 2007) | 3 lines Put the thread reading from the socket back in the idle list if it deferred the processing of a full frame to another thread ........ * /, channels/chan_iax2.c: Merged revisions 70866 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r70866 | russell | 2007-06-21 16:07:04 -0500 (Thu, 21 Jun 2007) | 5 lines If a full frame is received while one of the iax2 threads is in the middle of handling a full frame for the same call, queue it up for processing by that same thread later instead of dropping it. (issue #9937, patch by me) ........ 2007-06-21 20:28 +0000 [r70857] Steve Murphy * /, cdr/cdr_custom.c: Merged revisions 70841 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r70841 | murf | 2007-06-21 14:19:36 -0600 (Thu, 21 Jun 2007) | 9 lines Merged revisions 70804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r70804 | murf | 2007-06-21 13:13:17 -0600 (Thu, 21 Jun 2007) | 1 line it was pointed out that the cdr_custom config load could get a lock, and under certain circumstances, would never release it. I also noted that the situation where more than one mapping spec was warned about, but did not ignore further mappings as it had promised. I think I have fixed both situations. ........ ................ 2007-06-21 19:54 +0000 [r70809] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 70808 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r70808 | mmichelson | 2007-06-21 14:49:44 -0500 (Thu, 21 Jun 2007) | 4 lines When volgain is used don't leave a temporary file behind. (Closes Issue 8514, Reported and patched by ulogic, code reviewed by Jason Parker) ........ 2007-06-21 19:08 +0000 [r70794] Kevin P. Fleming * build_tools/make_buildopts_h: when we are building modules that other modules depend on, create preprocessor defines (in buildopts.h) marking that those modules were built 2007-06-21 18:40 +0000 [r70783] Russell Bryant * apps/app_meetme.c: Merge changes from team/russell/sla_reload * Add support for the reload of sla.conf (closes issue #9481, patch by me) 2007-06-21 18:03 +0000 [r70769] Matthew Fredrickson * channels/chan_zap.c: Remove deprecated function call 2007-06-21 15:58 +0000 [r70729-70731] Joshua Colp * res/res_agi.c: Expand AGISTATUS variable to include NOTFOUND which is set when the AGI file could not be found. (issue #9285 reported by srdjan) * /, main/rtp.c: Merged revisions 70727 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r70727 | file | 2007-06-21 11:22:39 -0400 (Thu, 21 Jun 2007) | 2 lines Do not Packet2Packet bridge if packetization settings do not allow it. (issue #9117 reported by phsultan) ........ 2007-06-21 15:23 +0000 [r70728] Russell Bryant * /, apps/app_meetme.c: Merged revisions 70726 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r70726 | russell | 2007-06-21 10:21:16 -0500 (Thu, 21 Jun 2007) | 2 lines Remove a couple of duplicate unlocks ........ 2007-06-21 14:00 +0000 [r70678] Joshua Colp * /, apps/app_voicemail.c: Merged revisions 70677 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r70677 | file | 2007-06-21 09:58:36 -0400 (Thu, 21 Jun 2007) | 2 lines Fix building with ODBC storage enabled. (issue #10025 reported by denisgalvao) ........ 2007-06-21 13:18 +0000 [r70676] Steve Murphy * main/cdr.c, /: Merged revisions 70656 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r70656 | murf | 2007-06-21 07:00:39 -0600 (Thu, 21 Jun 2007) | 1 line Via complaints aired in asterisk-users, I submit these changes, which allow cdr updates to see macro context/exten, whether hung up or not ........ 2007-06-20 23:33 +0000 [r70613] Jason Parker * /, cdr/cdr_pgsql.c: Merged revisions 70612 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r70612 | qwell | 2007-06-20 18:32:39 -0500 (Wed, 20 Jun 2007) | 4 lines Fix some potential memory leaks in cdr_pgsql. Issue 10020, patch by me, with credit to prashant_jois for pointing out the problem. ........ 2007-06-20 23:31 +0000 [r70611] Mark Michelson * apps/app_voicemail.c: Removed an extraneous debug message I'd left in my previous commit 2007-06-20 23:31 +0000 [r70610] Tilghman Lesher * main/pbx.c, apps/app_queue.c: Fix trunk brokenness; also, optimize application registration 2007-06-20 23:26 +0000 [r70607] Steve Murphy * apps/app_dial.c, main/pbx.c, apps/app_queue.c: Cleaning up a small disaster I created earlier 2007-06-20 22:55 +0000 [r70555-70561] Jason Parker * /, cdr/cdr_pgsql.c: Merged revisions 70560 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r70560 | qwell | 2007-06-20 17:55:21 -0500 (Wed, 20 Jun 2007) | 1 line Fix a stupid mistake in my last cdr_pgsql race condition fix ........ * /, cdr/cdr_pgsql.c: Merged revisions 70554 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r70554 | qwell | 2007-06-20 17:31:35 -0500 (Wed, 20 Jun 2007) | 4 lines Fix a race condition in cdr_pgsql that can occur when reloading the module. Issue 10022, patch by me, with credit to prashant_jois for finding the bug. ........ 2007-06-20 22:24 +0000 [r70553] Joshua Colp * /, channels/chan_sip.c: Merged revisions 70552 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r70552 | file | 2007-06-20 18:22:20 -0400 (Wed, 20 Jun 2007) | 10 lines Merged revisions 70551 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r70551 | file | 2007-06-20 18:20:16 -0400 (Wed, 20 Jun 2007) | 2 lines Don't overwrite the configured username setting upon a REGISTER. (issue #8565 reported by jsmith) ........ ................ 2007-06-20 21:38 +0000 [r70531] Steve Murphy * apps/app_dial.c, apps/app_queue.c: As per 9228, now app_queue should have the proper machinery to do gosubs. 2007-06-20 21:31 +0000 [r70530] Mark Michelson * apps/app_voicemail.c: Main fix: Fixing a bug which caused VoiceMailMain to always report that you had 0 messages when using IMAP storage. Secondary fixes: adding locks to list access in several places Big thanks to Russell Bryant for helping out with this. 2007-06-20 20:54 +0000 [r70493-70495] Jason Parker * /, channels/chan_skinny.c: Merged revisions 70494 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r70494 | qwell | 2007-06-20 15:53:16 -0500 (Wed, 20 Jun 2007) | 4 lines Make sure we clear the previously dialed number if it did not exist. Issue 9958. ........ * main/http.c: Revert the change made in revision 45474, since this causes other issues. Issue 10021. 2007-06-20 20:10 +0000 [r70461] Steve Murphy * pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/ael/ael_lex.c, pbx/pbx_ael.c, doc/ael.tex, include/asterisk/ael_structs.h, pbx/ael/ael.tab.h, CHANGES, pbx/ael/ael.flex: This finishes the changes for making Macro args LOCAL to the call, and allowing users to declare local variables. 2007-06-20 19:30 +0000 [r70446] Tilghman Lesher * apps/app_dial.c, /: Merged revisions 70445 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r70445 | tilghman | 2007-06-20 14:29:23 -0500 (Wed, 20 Jun 2007) | 10 lines Merged revisions 70444 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r70444 | tilghman | 2007-06-20 14:25:54 -0500 (Wed, 20 Jun 2007) | 2 lines Issue 9997 - Timelimit times out the wrong channel ........ ................ 2007-06-20 18:48 +0000 [r70398] Russell Bryant * channels/chan_zap.c, /: Merged revisions 70397 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r70397 | russell | 2007-06-20 13:46:49 -0500 (Wed, 20 Jun 2007) | 13 lines Merged revisions 70396 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r70396 | russell | 2007-06-20 13:45:38 -0500 (Wed, 20 Jun 2007) | 5 lines Fix a problem where an established call would not be properly disconnected when a PRI disconnect is received depending on which cause code was received. (issue #9588, original patch by softins, updated patch from jtexter3, and some additional feedback from mhardeman) ........ ................ 2007-06-20 17:55 +0000 [r70361] Joshua Colp * main/frame.c, /, main/rtp.c: Merged revisions 70360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r70360 | file | 2007-06-20 13:52:57 -0400 (Wed, 20 Jun 2007) | 2 lines Put the speex packetization values back in but disable it when setting up the smoother. ........ 2007-06-20 17:35 +0000 [r70358] Tilghman Lesher * apps/app_dial.c, pbx/pbx_ael.c: Merge work to make U(...) option work for Dial 2007-06-20 14:33 +0000 [r70310] Olle Johansson * channels/chan_zap.c: Show TDD status in "zap show channels" Inspired by work at Omnitor in Sweden 2007-06-20 13:00 +0000 [r70253-70291] Tilghman Lesher * apps/app_stack.c: Oops, shouldn't have taken that last shortcut (also add some checks) * apps/app_stack.c: Another method of doing local variables, hopefully a little closer to what codefreeze had in mind * apps/app_stack.c: Local variables for codefreeze 2007-06-20 02:13 +0000 [r70234] Russell Bryant * /, contrib/scripts/ast_grab_core: Merged revisions 70164 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r70164 | russell | 2007-06-19 19:03:22 -0500 (Tue, 19 Jun 2007) | 2 lines don't delete the backtrace in ast_grab_core ........ 2007-06-20 00:26 +0000 [r70199] Joshua Colp * main/frame.c, /: Merged revisions 70198 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r70198 | file | 2007-06-19 20:24:36 -0400 (Tue, 19 Jun 2007) | 2 lines Don't do packetization/smoother stuff with speex, it doesn't work. ........ 2007-06-19 23:38 +0000 [r70122-70162] Steve Murphy * CHANGES: Added a little verbage to CHANGES * apps/app_dial.c, apps/app_queue.c, apps/app_rpt.c: Via bug9228, no way to create macros via AEL, and some of the apps allow you to call macros..., I modded the apps that allow macro calls to allow gosubs calls also, to make them AEL compliant. * UPGRADE.txt, CHANGES: Moved those comments from UPGRADE.txt to CHANGES. Ooops. * UPGRADE.txt: Some UPGRADE.txt comments to cover some enhancements added today. * configs/cdr_manager.conf.sample, cdr/cdr_manager.c: This enhancement provided via bug 9993, a patch to upgrade cdr_manager to have cdr_custom capabilities. Many thanks to eserra for this contribution 2007-06-19 19:15 +0000 [r70088] Russell Bryant * channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged revisions 70084 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r70084 | russell | 2007-06-19 14:13:45 -0500 (Tue, 19 Jun 2007) | 3 lines Only attempt to queue a hangup on the owner channel if it actually exists. (issue #9795, patch from zandbelt) ........ 2007-06-19 18:31 +0000 [r70063] Steve Murphy * main/channel.c, /: Merged revisions 70062 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r70062 | murf | 2007-06-19 12:23:23 -0600 (Tue, 19 Jun 2007) | 9 lines Merged revisions 70053 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r70053 | murf | 2007-06-19 12:07:59 -0600 (Tue, 19 Jun 2007) | 1 line This fixes 9246, where channel variables are not available in the 'h' exten, on a 'ZOMBIE' channel. The fix is to consolidate the channel variables during a masquerade, and then copy the merged variables back onto the clone, so the zombie has the same vars that the 'original' has. ........ ................ 2007-06-19 17:09 +0000 [r70006] Joshua Colp * /, main/rtp.c: Merged revisions 70003 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r70003 | file | 2007-06-19 13:07:40 -0400 (Tue, 19 Jun 2007) | 10 lines Merged revisions 69992 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r69992 | file | 2007-06-19 13:00:58 -0400 (Tue, 19 Jun 2007) | 2 lines Handle the CC field in the RTP header. (issue #9384 reported by DoodleHu) ........ ................ 2007-06-19 17:07 +0000 [r70001] Steve Murphy * include/asterisk/callerid.h, channels/chan_zap.c, doc/India-CID.txt (added), configs/zapata.conf.sample: These changes were submitted via bug 6683, to allow CID detection in India, with carriers that do Polarity/DTMF CID signalling. 2007-06-19 16:25 +0000 [r69988] Joshua Colp * main/channel.c, /: Merged revisions 69987 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r69987 | file | 2007-06-19 12:24:31 -0400 (Tue, 19 Jun 2007) | 10 lines Merged revisions 69986 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r69986 | file | 2007-06-19 12:21:29 -0400 (Tue, 19 Jun 2007) | 2 lines Update BRIDGEPEER variable if set to the new channel name when a masquerade happens. (issue #9699 reported by dimas) ........ ................ 2007-06-19 15:27 +0000 [r69945] Russell Bryant * /, channels/chan_sip.c: Merged revisions 69944 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r69944 | russell | 2007-06-19 10:22:36 -0500 (Tue, 19 Jun 2007) | 10 lines Fix a crash that could occur when handing device state changes. When the state of a device changes, the device state thread tells the extension state handling code that it changed. Then, the extension state code calls the callback in chan_sip so that it can update subscriptions to that extension. A pointer to a sip_pvt structure is passed to this function as the call which needs a NOTIFY sent. However, there was no locking done to ensure that the pvt struct didn't disappear during this process. (issue #9946, reported by tdonahue, patch by me, patch updated to trunk to use the sip_pvt lock wrappers by eliel) ........ 2007-06-19 15:14 +0000 [r69943] Matthew Fredrickson * channels/chan_zap.c, configs/zapata.conf.sample: Add support for setting nature of address, presentation, and other related SS7 number options (#10000) 2007-06-19 13:56 +0000 [r69850-69896] Joshua Colp * /, apps/app_meetme.c: Merged revisions 69895 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r69895 | file | 2007-06-19 09:55:25 -0400 (Tue, 19 Jun 2007) | 10 lines Merged revisions 69894 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r69894 | file | 2007-06-19 09:54:03 -0400 (Tue, 19 Jun 2007) | 2 lines Perform an extra hangup check just in case. (issue #9589 reported by bcnit) ........ ................ * /, res/res_features.c: Merged revisions 69847 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r69847 | file | 2007-06-19 09:00:57 -0400 (Tue, 19 Jun 2007) | 10 lines Merged revisions 69846 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r69846 | file | 2007-06-19 08:57:55 -0400 (Tue, 19 Jun 2007) | 2 lines Add parked call extension AFTER the parking slot has been announced, otherwise two threads will try to handle the same channel and it will go kaboom. (issue #9191 reported by japple) ........ ................ 2007-06-18 23:28 +0000 [r69808-69809] Mark Michelson * apps/app_voicemail.c: Undoing my last commit. I misread the code before. * apps/app_voicemail.c: Cleaned up a section where there were two consecutive identical if statements. Combined the bodies of the two into one if. I blame svn merging for this. 2007-06-18 22:23 +0000 [r69807] Brett Bryant * apps/app_queue.c: Fixed issue where 'stop gracfeully' was hanging ... 2007-06-18 21:58 +0000 [r69806] Joshua Colp * /, main/callerid.c: Merged revisions 69805 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r69805 | file | 2007-06-18 17:57:10 -0400 (Mon, 18 Jun 2007) | 2 lines Fix for building on PowerPC under Linux. ........ 2007-06-18 19:52 +0000 [r69797] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 69796 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r69796 | tilghman | 2007-06-18 14:48:17 -0500 (Mon, 18 Jun 2007) | 2 lines Issue 10005 - Segfault with missing arguments, plus fix a missing define for SIP INFO channels ........ 2007-06-18 19:02 +0000 [r69779-69795] Joshua Colp * /, channels/chan_sip.c: Merged revisions 69794 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r69794 | file | 2007-06-18 15:00:50 -0400 (Mon, 18 Jun 2007) | 2 lines Don't count RTP timeout when involved in a T38 fax session. (issue #9222 reported by ivoc) ........ * /, channels/chan_sip.c: Merged revisions 69775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r69775 | file | 2007-06-18 14:18:12 -0400 (Mon, 18 Jun 2007) | 10 lines Merged revisions 69765 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r69765 | file | 2007-06-18 14:13:03 -0400 (Mon, 18 Jun 2007) | 2 lines Set the peer name on the dialog to the one configured in sip.conf and NOT the username to be used for authentication attempts. (issue #9967 reported by achauvin) ........ ................ 2007-06-18 17:50 +0000 [r69745-69746] Tilghman Lesher * /, contrib/scripts/safe_asterisk: Merged revisions 69744 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r69744 | tilghman | 2007-06-18 12:46:40 -0500 (Mon, 18 Jun 2007) | 10 lines Merged revisions 69743 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r69743 | tilghman | 2007-06-18 12:45:15 -0500 (Mon, 18 Jun 2007) | 2 lines Issue 9998 - Remove SIG prefix, since it's not supported by ksh ........ ................ * apps/app_rpt.c: Janitor for ast_localtime 2007-06-18 16:56 +0000 [r69705-69709] Joshua Colp * main/dnsmgr.c, /: Merged revisions 69708 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r69708 | file | 2007-06-18 12:51:36 -0400 (Mon, 18 Jun 2007) | 2 lines Remember the DNS lookup done when dnsmgr is called for the first time so that it does not needlessly spit out changed messages when the host really didn't change. ........ * main/cdr.c, main/dnsmgr.c, main/asterisk.c: Few more rwlist conversions... why not. 2007-06-18 16:35 +0000 [r69691-69703] Russell Bryant * res/res_config_odbc.c, /, build_tools/menuselect-deps.in, configure, funcs/func_odbc.c, include/asterisk/autoconfig.h.in, configure.ac, cdr/cdr_odbc.c, res/res_odbc.c, apps/app_voicemail.c: Merged revisions 69702 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r69702 | russell | 2007-06-18 11:35:02 -0500 (Mon, 18 Jun 2007) | 6 lines To prevent 92138749238754 more reports of "I have unixodbc installed, but still can't build *_odbc.so!", check for ltdl directly, instead of just listing it as another library to include in the unixodbc check in the configure script. This also makes ltdl show up as a dependency in menuselect so people know what to go install. (related to issue #9989, patch by me) ........ * /, build_tools/prep_moduledeps: Merged revisions 69689 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r69689 | russell | 2007-06-18 11:15:12 -0500 (Mon, 18 Jun 2007) | 5 lines Change the use of "echo -e" to "printf". On systems where /bin/sh is not bash, most of the lines in menuselect-tree were getting a "-e" at the beginning of every line. I'm surprised nobody noticed this, but I think the XML parser was being very nice and ignoring them. ........ 2007-06-18 16:06 +0000 [r69663-69672] Joshua Colp * /, channels/chan_sip.c: Merged revisions 69668 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r69668 | file | 2007-06-18 12:04:55 -0400 (Mon, 18 Jun 2007) | 2 lines Don't defer the BYE till later on a transfer when the transfer itself goes kaboom and has no hope of working. ........ * /, channels/chan_sip.c: Merged revisions 69661 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r69661 | file | 2007-06-18 11:46:32 -0400 (Mon, 18 Jun 2007) | 2 lines Few minor transfer tweaks. We can't unlock something we never locked, and better handle a specific scenario with doing an attended transfer between two non-bridged calls. ........ 2007-06-18 15:46 +0000 [r69662] Russell Bryant * Makefile, /: Merged revisions 69660 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r69660 | russell | 2007-06-18 10:46:14 -0500 (Mon, 18 Jun 2007) | 2 lines Tweak paths for BSD systems (issue #10001, stuarth) ........ 2007-06-18 13:57 +0000 [r69626] Joshua Colp * /, channels/chan_sip.c: Merged revisions 69625 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r69625 | file | 2007-06-18 09:55:00 -0400 (Mon, 18 Jun 2007) | 2 lines Fix issue where it would be possible for the negotiated codecs to get set back to nothing. (issue #9992 reported by yehavi) ........ 2007-06-15 20:21 +0000 [r69583] Russell Bryant * /: This was only an issue in 1.4. This issue was fixed in trunk as a part of bbryant's patch to support named dynamic feature groups. Merged revisions 69579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r69579 | russell | 2007-06-15 15:18:58 -0500 (Fri, 15 Jun 2007) | 5 lines Fix a silly deadlock in res_features that I found while debugging on one of blitzrage's test machines. It was one of the situations where he was seeing hung channels, and may be the cause of some of the reports from other people. (related to issue #9235) ........ 2007-06-15 19:25 +0000 [r69559] Joshua Colp * /, apps/app_speech_utils.c: Merged revisions 69558 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r69558 | file | 2007-06-15 15:23:45 -0400 (Fri, 15 Jun 2007) | 2 lines Add support for setting the maximum length of acceptable DTMF in SpeechBackground. 2007-06-15 15:36 +0000 [r69519] Russell Bryant * /, apps/app_meetme.c: Merged revisions 69518 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r69518 | russell | 2007-06-15 10:27:34 -0500 (Fri, 15 Jun 2007) | 5 lines The SLATRUNK_STATUS variable indicated "SUCCESS" for both an answer of the incoming call on the trunk, or if the trunk reached its ring timeout. This patch changes the variable to say "RINGTIMEOUT" in that case. (issue #9973, reported by n00dle, patch by me) ........ 2007-06-14 23:23 +0000 [r69471] Jason Parker * /, main/config.c: Merged revisions 69470 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r69470 | qwell | 2007-06-14 18:22:51 -0500 (Thu, 14 Jun 2007) | 12 lines Merged revisions 69469 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r69469 | qwell | 2007-06-14 18:21:45 -0500 (Thu, 14 Jun 2007) | 4 lines Fix an issue where the line number in an unterminated comment block error message would show the wrong line number. "Reported" to me on #asterisk (somebody posted an error message, and I happened to catch it) ........ ................ 2007-06-14 23:01 +0000 [r69436] Russell Bryant * main/pbx.c, channels/chan_vpb.cc, apps/app_meetme.c, res/res_features.c, channels/iax2-provision.c, main/enum.c, res/res_monitor.c, apps/app_speech_utils.c, main/loader.c, main/cli.c, main/channel.c, channels/chan_misdn.c, apps/app_minivm.c, main/http.c, main/file.c, channels/chan_h323.c, res/res_indications.c, apps/app_directory.c, main/asterisk.c: Convert uses of strdup() to ast_strdup() (issue #9983, eliel) 2007-06-14 22:56 +0000 [r69435] Jason Parker * /, sounds/Makefile: Merged revisions 69434 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r69434 | qwell | 2007-06-14 17:56:09 -0500 (Thu, 14 Jun 2007) | 1 line Update to latest versions of sound files. ........ 2007-06-14 22:09 +0000 [r69394-69405] Kevin P. Fleming * include/asterisk/utils.h, main/pbx.c, /, main/say.c, cdr/cdr_pgsql.c, cdr/cdr_radius.c, channels/chan_iax2.c, cdr/cdr_tds.c, cdr/cdr_csv.c, main/cdr.c, channels/chan_phone.c, main/manager.c, cdr/cdr_sqlite.c, apps/app_minivm.c, main/callerid.c, main/logger.c, main/stdtime/localtime.c, cdr/cdr_odbc.c, main/asterisk.c, cdr/cdr_manager.c, channels/chan_mgcp.c, apps/app_voicemail.c: Merged revisions 69392 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r69392 | kpfleming | 2007-06-14 16:50:40 -0500 (Thu, 14 Jun 2007) | 2 lines use ast_localtime() in every place localtime_r() was being used ........ * formats/format_ogg_vorbis.c: oops... somebody patched this module without compile-testing it... bad :-) 2007-06-14 21:09 +0000 [r69327-69360] Russell Bryant * /, main/say.c: Merged revisions 69358 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r69358 | russell | 2007-06-14 16:08:23 -0500 (Thu, 14 Jun 2007) | 3 lines Fix some problems with saying dates and times for the "tw" langauge (issue #9964, ljmid) ........ * CHANGES: update CHANGES for tw support in voicemail * apps/app_voicemail.c: Add support for the tw language in voicemail (issue #9964, ljmid) * funcs/func_rand.c, main/frame.c, channels/chan_local.c, res/res_features.c, apps/app_record.c, funcs/func_strings.c, apps/app_test.c, main/devicestate.c, apps/app_alarmreceiver.c, apps/app_ices.c, channels/chan_iax2.c, main/config.c, res/res_smdi.c, channels/chan_skinny.c, apps/app_zapscan.c, apps/app_zapras.c, apps/app_amd.c, channels/chan_alsa.c, cdr/cdr_odbc.c, main/db.c, apps/app_dial.c, formats/format_wav.c, channels/chan_agent.c, apps/app_disa.c, formats/format_ogg_vorbis.c, channels/iax2-provision.c, apps/app_talkdetect.c, apps/app_db.c, res/res_monitor.c, apps/app_zapbarge.c, channels/chan_misdn.c, channels/chan_features.c, apps/app_macro.c, funcs/func_iconv.c, formats/format_g726.c, apps/app_chanspy.c, main/asterisk.c, apps/app_voicemail.c, channels/chan_vpb.cc, apps/app_meetme.c, res/res_musiconhold.c, cdr/cdr_pgsql.c, channels/chan_gtalk.c, apps/app_followme.c, codecs/codec_zap.c, cdr/cdr_radius.c, res/res_jabber.c, res/res_config_sqlite.c, main/enum.c, cdr/cdr_csv.c, main/cdr.c, main/channel.c, main/dial.c, channels/chan_phone.c, apps/app_osplookup.c, apps/app_minivm.c, res/res_agi.c, apps/app_mp3.c, main/app.c, apps/app_rpt.c, main/dns.c, channels/chan_mgcp.c, apps/app_nbscat.c, res/res_config_pgsql.c, funcs/func_version.c, channels/chan_zap.c, funcs/func_db.c, channels/chan_sip.c, apps/app_festival.c, apps/app_waitforsilence.c, res/res_crypto.c, res/res_adsi.c, main/acl.c, apps/app_queue.c, cdr/cdr_tds.c, channels/chan_jingle.c, apps/app_channelredirect.c, apps/app_directed_pickup.c, main/adsistub.c, main/callerid.c, main/file.c, channels/chan_h323.c, channels/chan_nbs.c, apps/app_stack.c, main/dsp.c: Add a massive set of changes for converting to use the ast_debug() macro. (issue #9957, patches from mvanbaak, caio1982, critch, and dimas) 2007-06-14 16:41 +0000 [r69308] Matthew Fredrickson * channels/chan_zap.c: Clean up debug messages a little bit for ss7 linkset debugging 2007-06-14 15:43 +0000 [r69261] Brett Bryant * main/manager.c: Couple of manager ssl options weren't loading because of a typo. 2007-06-14 15:25 +0000 [r69260] Jason Parker * funcs/func_groupcount.c, /: Merged revisions 69259 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r69259 | qwell | 2007-06-14 10:21:29 -0500 (Thu, 14 Jun 2007) | 12 lines Merged revisions 69258 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r69258 | qwell | 2007-06-14 10:15:53 -0500 (Thu, 14 Jun 2007) | 4 lines Change a quite broken while loop to a for loop, so "continue;" works as expected instead of eating 99% CPU... Issue 9966, patch by me. ........ ................ 2007-06-13 21:20 +0000 [r69223] Joshua Colp * /, channels/chan_iax2.c: Merged revisions 69221 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r69221 | file | 2007-06-13 17:17:28 -0400 (Wed, 13 Jun 2007) | 2 lines Let's make chan_iax2 media only native transfers actually work. (issue #9376 reported by simone cittadini) ........ 2007-06-13 20:03 +0000 [r69187] Russell Bryant * /, channels/chan_sip.c: Merged revisions 69183 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r69183 | russell | 2007-06-13 14:57:38 -0500 (Wed, 13 Jun 2007) | 9 lines Move the logic for destroying a call when no response is received to a BYE outside of the block that checks for FLAG_FATAL to be set. This flag is only set when the packet is transmitted with the reliability set to XMIT_CRITICAL when the original packet is transmitted. A BYE is always sent with it set to XMIT_RELIABLE, meaning this code could never be encountered. This resulted in seeing some SIP channels that would never go away with the last packet sent being a BYE. (part of issue #9235, patch from jcmoore) ........ 2007-06-13 20:00 +0000 [r69185] Joshua Colp * /, channels/iax2-parser.c: Merged revisions 69184 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r69184 | file | 2007-06-13 15:58:59 -0400 (Wed, 13 Jun 2007) | 2 lines Add TXMEDIA to list so that it is properly displayed during iax2 packet output. ........ 2007-06-13 19:47 +0000 [r69182] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 69181 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r69181 | mmichelson | 2007-06-13 14:41:13 -0500 (Wed, 13 Jun 2007) | 5 lines Contains a patch for fixing an encoding problem when using Outlook to view voicemail emails and attachments. This fix has also been tested on Thunderbird, Evolution, Pine, and Mutt. (Issue 9336, reported by marwick, patched by mutterc) ........ 2007-06-13 19:10 +0000 [r69147] Joshua Colp * /, apps/app_meetme.c: Merged revisions 69144 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r69144 | file | 2007-06-13 15:08:24 -0400 (Wed, 13 Jun 2007) | 2 lines Really ignore NULL frames and check whether the channel hungup or not. (issue #9912 reported by junky) ........ 2007-06-13 19:05 +0000 [r69137] Jason Parker * channels/chan_agent.c: Completely remove callback mode and all references to it from chan_agent. Issue 9969, patch by eliel. 2007-06-13 18:23 +0000 [r69129-69130] Joshua Colp * include/asterisk/app.h, funcs/func_groupcount.c, main/app.c, main/cli.c: Use read/write lock based lists for group counting. * /, main/app.c: Merged revisions 69128 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r69128 | file | 2007-06-13 14:16:00 -0400 (Wed, 13 Jun 2007) | 10 lines Merged revisions 69127 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r69127 | file | 2007-06-13 14:12:48 -0400 (Wed, 13 Jun 2007) | 2 lines Return group counting to previous behavior where you could only have one group per category. (issue #9711 reported by irroot) ........ ................ 2007-06-13 17:37 +0000 [r69081-69108] Jason Parker * res/res_config_pgsql.c: Continuation of issue 9968 (revision 69081). This should be the last one. * main/pbx.c, channels/chan_sip.c: Fixes for ast_strlen_zero() janitor project. Issue 9968, patch by eliel. 2007-06-13 16:59 +0000 [r69017-69072] Russell Bryant * /, channels/chan_sip.c: Merged revisions 69071 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r69071 | russell | 2007-06-13 11:56:16 -0500 (Wed, 13 Jun 2007) | 3 lines Clarify a bit of logic. This doesn't change behavior in any way, but it is helpful when following the logic to debug problems like 9235. ........ * /, channels/chan_iax2.c: Merged revisions 69069 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r69069 | russell | 2007-06-13 11:29:12 -0500 (Wed, 13 Jun 2007) | 3 lines Fix a place where a chan_iax2 pvt struct was accessed without the lock held. This issue was reported to me via email by Dmitry Mishchenko. Thanks! ........ * res/snmp/agent.c: Simplify some logic and convert spaces to tabs * res/snmp/agent.c: The variable used for the return value must be declared as static. I broke this when applying the patch, sorry! (issue #9637, jeffg) * include/asterisk/logger.h: Put parenthesis around the level argument to ast_debug() * /, cdr/cdr_pgsql.c: Merged revisions 69016 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r69016 | russell | 2007-06-12 14:40:17 -0500 (Tue, 12 Jun 2007) | 4 lines Fix a memory leak pointed out by prashant_jois in #asterisk-bugs. PQclear() was not called on the result structure after doing a PQexec(). Also, fix up some formatting in passing. ........ 2007-06-12 19:38 +0000 [r69013-69015] Joshua Colp * /, channels/chan_iax2.c: Merged revisions 69014 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r69014 | file | 2007-06-12 15:36:29 -0400 (Tue, 12 Jun 2007) | 2 lines Change the full frame dropping log message to debug to avoid future bug reports. ........ * /, channels/chan_iax2.c: Merged revisions 69012 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r69012 | file | 2007-06-12 15:26:38 -0400 (Tue, 12 Jun 2007) | 2 lines Schedule the sending of a PING packet a second later than previously so that it does not collide with the LAGRQ. ........ 2007-06-12 19:19 +0000 [r68970-69011] Russell Bryant * main/channel.c, /: Merged revisions 69010 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r69010 | russell | 2007-06-12 14:13:41 -0500 (Tue, 12 Jun 2007) | 12 lines In ast_channel_make_compatible(), just return if the channels' read and write formats already match up. There are code paths that call this function on a pair of channels multiple times. This made calls fail that were using g729 in some cases. The reason is that codec_g729a will unregister itself from the list of available translators will all licenses are in use. So, the first time the function got called, the right translation path was allocated. However, the second time it got called, the code would not find a translation path to/from g729 and make the call fail, even if the channel actually already had a g729 translation path allocated. (SPD-32) ........ * main/pbx.c: Convert pbx.c to use ast_debug() for debug logging. (issue #9925, dimas) * include/asterisk/logger.h: Add a new macro, ast_debug(), which combines the check of the value of option_debug and the actual call to ast_log(). (issue #9925, dimas) * doc/ast_appdocs.tex: update the dump of application docs * apps/app_dial.c, apps/app_privacy.c, apps/app_authenticate.c, channels/chan_agent.c, apps/app_image.c, apps/app_chanisavail.c, apps/app_transfer.c, apps/app_system.c, apps/app_queue.c, apps/app_playback.c, apps/app_controlplayback.c, apps/app_osplookup.c, apps/app_sendtext.c, apps/app_minivm.c, apps/app_url.c, pbx/pbx_config.c, include/asterisk/options.h, apps/app_voicemail.c: Completely remove all of the code related to jumping to priority n + 101. yay! (issue #9926, caio1982) 2007-06-12 14:26 +0000 [r68900-68923] Joshua Colp * /, main/rtp.c: Merged revisions 68922 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r68922 | file | 2007-06-12 10:23:11 -0400 (Tue, 12 Jun 2007) | 10 lines Merged revisions 68921 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r68921 | file | 2007-06-12 10:18:57 -0400 (Tue, 12 Jun 2007) | 2 lines Bring RTP back to Asterisk at the end of a native bridge no matter what. ........ ................ * main/autoservice.c, main/app.c: Even more minor code cleanup! * main/channel.c: Minor code cleanup. * channels/chan_agent.c: Remove old stuff from the AgentCallbackLogin days and only autocomplete agents in the agent logoff CLI command that are logged in. (issue #9952 reported by eliel) 2007-06-11 22:31 +0000 [r68855] Dwayne M. Hubbard * main/frame.c: corrected CLI 'core show codecs' syntax for issue 9945, thanks eserra. 2007-06-11 22:21 +0000 [r68854] Tilghman Lesher * apps/app_disa.c, UPGRADE.txt: Issue 8971 - Allow DISA input to be ended with a '#'. 2007-06-11 22:07 +0000 [r68816-68831] Jason Parker * main/manager.c, configs/manager.conf.sample: Change displayconnects option in manager.conf to be per-user. Issue 9932, patch by eliel * /, include/asterisk/time.h: Merged revisions 68814 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r68814 | qwell | 2007-06-11 16:20:15 -0500 (Mon, 11 Jun 2007) | 2 lines Solaris 10 sometimes (?) needs this include in order to have NULL defined. ........ 2007-06-11 20:51 +0000 [r68782] Tilghman Lesher * /, apps/app_directory.c: Merged revisions 68781 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r68781 | tilghman | 2007-06-11 15:45:53 -0500 (Mon, 11 Jun 2007) | 2 lines Issue 9947 - fn2 was unused / incorrectly used ........ 2007-06-11 17:05 +0000 [r68740] Christian Richter * channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c: Merged revisions 68733 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r68733 | crichter | 2007-06-11 18:57:59 +0200 (Mo, 11 Jun 2007) | 9 lines Merged revisions 68732 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r68732 | crichter | 2007-06-11 18:49:00 +0200 (Mo, 11 Jun 2007) | 1 line added check for NULL Pointer when calling misdn_new. Asterisk does not allow us to create channels anymore when stop gracefully is used :). also modified the restart_indicator to 0 ........ ................ 2007-06-11 14:41 +0000 [r68662-68685] Joshua Colp * main/channel.c: Change channel list to read/write list... I'm crazy. * main/channel.c, /: Merged revisions 68683 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r68683 | file | 2007-06-11 10:33:12 -0400 (Mon, 11 Jun 2007) | 10 lines Merged revisions 68682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r68682 | file | 2007-06-11 10:29:58 -0400 (Mon, 11 Jun 2007) | 2 lines Improve deadlock handling of the channel list. (issue #8376 reported by one47) ........ ................ * main/manager.c: Add username completion for manager show user CLI command. (issue #9929 reported by eliel) * configs/sip.conf.sample: Update documentation for proper CLI commands. (issue #9936 reported by eserra) 2007-06-11 11:40 +0000 [r68661] Christian Richter * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c: Merged revisions 68644 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r68644 | crichter | 2007-06-11 12:29:18 +0200 (Mo, 11 Jun 2007) | 9 lines Merged revisions 68631 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r68631 | crichter | 2007-06-11 11:18:01 +0200 (Mo, 11 Jun 2007) | 1 line fixed problem that the dummybc chanels had no lock, checking for the lock now. Also fixed the channel restart stuff, we can now specify and restart particular channels too. ........ ................ 2007-06-11 04:28 +0000 [r68596] Tilghman Lesher * /, pbx/pbx_config.c: Merged revisions 68595 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r68595 | tilghman | 2007-06-10 23:21:30 -0500 (Sun, 10 Jun 2007) | 2 lines "dialplan save" produced garbage in the config file ........ 2007-06-09 01:06 +0000 [r68575] Jason Parker * channels/chan_misdn.c: Fix compile errors in chan_misdn.c Reported by d1mas in #asterisk-bugs on IRC. 2007-06-08 22:23 +0000 [r68473-68528] Russell Bryant * /, apps/app_dictate.c: Merged revisions 68527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r68527 | russell | 2007-06-08 17:23:22 -0500 (Fri, 08 Jun 2007) | 12 lines Merged revisions 68526 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r68526 | russell | 2007-06-08 17:22:36 -0500 (Fri, 08 Jun 2007) | 4 lines Don't automatically hang up after running Dictate so that callers can exit cleanly using '#' (closes issue #9577, patch from Thomas Andrews) ........ ................ * doc/asterisk-mib.txt, res/snmp/agent.c: Add support for retrieving the number of channels that are currently bridged via SNMP. (closes issue #9637, initial patch from jeffg, modified by me) * include/asterisk/app.h, res/res_agi.c, main/app.c, apps/app_controlplayback.c, apps/app_voicemail.c: Add an option for ControlPlayback to be able to start at an offset from the beginning of the file. Also, add a channel variable that indicates the location in the file where the Playback was stopped. (closes issue #7655, patch from sharkey) * main/pbx.c: Add channel variable manager event (issue #7291, patch from tonyh and jontow) 2007-06-08 16:03 +0000 [r68453] Kevin P. Fleming * /, channels/chan_iax2.c: Merged revisions 68450 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r68450 | kpfleming | 2007-06-08 10:52:47 -0500 (Fri, 08 Jun 2007) | 2 lines actually remember the type/subclass of full frames that are in process ........ 2007-06-08 15:51 +0000 [r68449] Jason Parker * res/res_config_sqlite.c: Fix incorrect logic for param count. Issue 9918. 2007-06-08 15:32 +0000 [r68448] Russell Bryant * main/asterisk.c: Minor formatting change to test changes to mantis auto-closing issues (closes issue #6000) 2007-06-08 00:18 +0000 [r68374-68405] Joshua Colp * /, main/say.c: Merged revisions 68401 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r68401 | file | 2007-06-07 20:17:04 -0400 (Thu, 07 Jun 2007) | 10 lines Merged revisions 68397 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r68397 | file | 2007-06-07 20:15:33 -0400 (Thu, 07 Jun 2007) | 2 lines Don't call ast_waitstream_full when the control file descriptor and audio file descriptor are not set, simply call ast_waitstream! (issue #8530 reported by rickead2000) ........ ................ * main/dnsmgr.c, /: Merged revisions 68370 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r68370 | file | 2007-06-07 20:02:34 -0400 (Thu, 07 Jun 2007) | 10 lines Merged revisions 68368 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r68368 | file | 2007-06-07 19:59:04 -0400 (Thu, 07 Jun 2007) | 2 lines Do a DNS lookup immediately upon calling the dnsmgr function, don't wait until a refresh happens. (issue #9097 reported by plack) ........ ................ 2007-06-07 23:17 +0000 [r68339-68359] Russell Bryant * /, main/say.c: Merged revisions 68354 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r68354 | russell | 2007-06-07 18:14:45 -0500 (Thu, 07 Jun 2007) | 11 lines Merged revisions 68351 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r68351 | russell | 2007-06-07 18:13:33 -0500 (Thu, 07 Jun 2007) | 3 lines Fix a problem where saying a character wouldn't properly break out when the caller pressed '#' (issue #8113, reported by patbaker82, patch from jamesgolovich (hey, long time no see!) and patbaker82) ........ ................ * include/asterisk/devicestate.h, channels/chan_sip.c, contrib/asterisk-ng-doxygen, main/devicestate.c, include/asterisk/manager.h, res/res_config_sqlite.c, main/rtp.c, include/asterisk/http.h, include/asterisk/doxyref.h, main/manager.c, include/asterisk/event.h, funcs/func_shell.c, apps/app_skel.c, channels/chan_h323.c, include/asterisk/strings.h, include/asterisk/stringfields.h: Fix a bunch of doxygen errors and document more things (issue #9842, snuffy) 2007-06-07 23:00 +0000 [r68327] Jason Parker * /, apps/app_voicemail.c: Merged revisions 68326 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r68326 | qwell | 2007-06-07 18:00:01 -0500 (Thu, 07 Jun 2007) | 5 lines Fix incorrect French syntax of "old messages". Request for feedback was sent to asterisk-dev mailing list, with little response. Issue 9118, patch by junky. ........ 2007-06-07 22:38 +0000 [r68325] Russell Bryant * channels/chan_zap.c: Fix a couple of places that got missed in the conversion to using the new API call for creating detached threads. (issue #9915, reported by elguro, fixed by me) 2007-06-07 22:18 +0000 [r68321] Kevin P. Fleming * /, channels/chan_iax2.c: Merged revisions 68313 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r68313 | kpfleming | 2007-06-07 17:14:35 -0500 (Thu, 07 Jun 2007) | 6 lines some improvements to the IAX2 full frame dropping logic recently added: - use inaddrcmp(), since we have it - output the type of frame and subclass being dropped, and the type/subclass that is already being processed (which caused the drop) ........ 2007-06-07 21:22 +0000 [r68284-68289] Russell Bryant * res/res_jabber.c: Doxygenify a lot of the functions in res_jabber (issue #9886, snuffy) * /, channels/chan_agent.c, apps/app_queue.c: Merged revisions 68280 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r68280 | russell | 2007-06-07 16:16:07 -0500 (Thu, 07 Jun 2007) | 4 lines Fix loading persistent queue members when using realtime configuration for queues. Also, remove an unneeded leading slash for the astdb family. (issue #9911, patch by atis) ........ 2007-06-07 20:25 +0000 [r68220-68251] Jason Parker * /, channels/chan_skinny.c: Merged revisions 68249 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r68249 | qwell | 2007-06-07 15:25:18 -0500 (Thu, 07 Jun 2007) | 4 lines Fix an issue with newer phones which require packets be padded out to the correct length. Issue 9887, patch by DEA. ........ * /, apps/app_voicemail.c: Merged revisions 68211 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r68211 | qwell | 2007-06-07 15:06:00 -0500 (Thu, 07 Jun 2007) | 12 lines Merged revisions 68204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r68204 | qwell | 2007-06-07 15:02:50 -0500 (Thu, 07 Jun 2007) | 4 lines Don't try to save voicemail greetings unless the user presses '1' to accept/save. Issue 9904, patch by me. ........ ................ 2007-06-07 19:51 +0000 [r68201] Olle Johansson * CREDITS: Adding Philippe to CREDITS for hard work on detecting bugs in our jabber/jingle integration 2007-06-07 19:50 +0000 [r68200] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 68198 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r68198 | mmichelson | 2007-06-07 14:47:42 -0500 (Thu, 07 Jun 2007) | 5 lines Submitting a fix for Issue 8016. Added a check to make sure that greetings get stored properly. (Issue 8016, reported by edhorton, patched by alamantia with modification by me. Thanks to Jason Parker for the advice on this). ........ 2007-06-07 19:49 +0000 [r68195-68199] Olle Johansson * /, channels/chan_features.c: Merged revisions 68196 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r68196 | oej | 2007-06-07 21:46:10 +0200 (Thu, 07 Jun 2007) | 2 lines Disable chan_features by default in menuselect ........ * channels/chan_sip.c: - Doxygen updates - Adding docs on flags to be able to clean up a bit 2007-06-07 19:31 +0000 [r68193] Russell Bryant * /, main/strcompat.c: Merged revisions 68192 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r68192 | russell | 2007-06-07 14:30:30 -0500 (Thu, 07 Jun 2007) | 3 lines Include stdarg.h for build issues on Solaris (issue #9381) ........ 2007-06-07 18:41 +0000 [r68138-68158] Joshua Colp * main/channel.c, /: Merged revisions 68157 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r68157 | file | 2007-06-07 14:39:52 -0400 (Thu, 07 Jun 2007) | 2 lines Fix logic when doing a name based channel search for a structure when you want to start from a specific point in the channel list. (issue #9324 reported by slavon) ........ * doc/queues-with-callback-members.tex: AEL in trunk now uses GOSUB so we have to update the queues with callback members example. (issue #9813 reported by Mike Anikienko) 2007-06-07 15:48 +0000 [r68118] Russell Bryant * res/res_jabber.c: Minor formatting change ... testing mantis stuff to see if we're done (issue #9790) (closes issue #9816) 2007-06-07 14:23 +0000 [r68072] Joshua Colp * apps/app_dial.c, /: Merged revisions 68071 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r68071 | file | 2007-06-07 10:21:59 -0400 (Thu, 07 Jun 2007) | 10 lines Merged revisions 68070 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r68070 | file | 2007-06-07 10:19:40 -0400 (Thu, 07 Jun 2007) | 2 lines Allow the 'g' option to work if used with the 'S' option. (issue #9888 reported by gasparz) ........ ................ 2007-06-07 10:06 +0000 [r67991-68040] Olle Johansson * /, res/res_jabber.c: Merged revisions 68030 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r68030 | oej | 2007-06-07 12:00:17 +0200 (Thu, 07 Jun 2007) | 2 lines Adding a few Todo's to res_jabber so we don't forget. ........ * /, res/res_jabber.c: Merged revisions 68028 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r68028 | oej | 2007-06-07 11:55:13 +0200 (Thu, 07 Jun 2007) | 4 lines Ok, we found out that this is not about if you have any *active* clients using TLS, but if you have initialized TLS at all during the lifetime of the module. So if you reload to disable TLS, it won't help. ........ * /, res/res_jabber.c: Merged revisions 68027 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r68027 | oej | 2007-06-07 11:42:26 +0200 (Thu, 07 Jun 2007) | 8 lines If you have a jabber client that uses TLS, refuse unload. Bad fix, but will prevent crashes while we are trying to find a workaround. Iksemel development seems to have stalled and we might have to stop using the TCP/TLS connections in that library and use our own, which would scale better from a poll/select perspective I guess. It would also make it easier to migrate to OpenSSL and stop Asterisk from depending on both OpenSSL and GnuTLS. ........ * /, include/asterisk/jabber.h, res/res_jabber.c: Merged revisions 67993 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67993 | oej | 2007-06-07 11:00:44 +0200 (Thu, 07 Jun 2007) | 6 lines Issue #9738 - Make sure we can unload res_jabber. Patch by phsultan - thanks! Due to a bug in the iksemel library, this will not work if you are using GTLS in the connection. That's being investigated. If you figure out a way to handle that without us having to patch iksemel, let us know in the bug report. Thanks. ........ * res/res_jabber.c: Simplification of res_jabber code (done at Inria, Paris with Philippe) * main/strcompat.c: Reverting part of #67864 to be able to compile agi/eagi-test that relies on this without having ast_log and other asterisk api functions available - I could not compile on OS/X without reverting this. 2007-06-07 00:12 +0000 [r67925-67944] Joshua Colp * /, channels/chan_sip.c: Merged revisions 67941 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r67941 | file | 2007-06-06 20:10:48 -0400 (Wed, 06 Jun 2007) | 10 lines Merged revisions 67938 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r67938 | file | 2007-06-06 20:09:13 -0400 (Wed, 06 Jun 2007) | 2 lines Only notify the devicestate system of a peer state change when the peer is built from the config file. (issue #9900 reported by arkadia) ........ ................ * /, main/file.c: Merged revisions 67924 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67924 | file | 2007-06-06 19:38:15 -0400 (Wed, 06 Jun 2007) | 2 lines Properly handle cases where a stream can't be written to. (issue #9757 reported by junky) ........ 2007-06-06 23:12 +0000 [r67920] Matthew Fredrickson * channels/chan_zap.c: Allow overlapdialing directions to be configurable. Bug #8554 2007-06-06 22:35 +0000 [r67901] Dwayne M. Hubbard * channels/chan_iax2.c: added CLI 'iax2 unregister ' for issue 9812, thanks eliel 2007-06-06 22:27 +0000 [r67875-67895] Russell Bryant * channels/chan_sip.c, configs/sip.conf.sample: Remove our little joke that was making fun of email disclaimers which nobody else seemed to think was very funny. Oh well ... :) * /, res/res_snmp.c: Merged revisions 67872 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67872 | russell | 2007-06-06 17:08:02 -0500 (Wed, 06 Jun 2007) | 6 lines Disable reload functionality in res_snmp. It is not possible to initialize the snmp library more than once without completely unloading the module and loading it again. (issue #9571, reported by hristo, additional helpful debug information from festr, patch from me) ........ 2007-06-06 21:20 +0000 [r67864] Tilghman Lesher * main/udptl.c, main/autoservice.c, main/frame.c, channels/chan_local.c, apps/app_readfile.c, res/res_features.c, main/threadstorage.c, main/say.c, funcs/func_strings.c, apps/app_alarmreceiver.c, main/devicestate.c, cdr/cdr_adaptive_odbc.c, channels/chan_iax2.c, main/indications.c, main/config.c, main/loader.c, main/cli.c, res/res_smdi.c, channels/chan_skinny.c, main/strcompat.c, main/http.c, apps/app_externalivr.c, cdr/cdr_odbc.c, main/db.c, res/res_speech.c, apps/app_milliwatt.c, main/sched.c, apps/app_dial.c, main/pbx.c, channels/chan_agent.c, channels/iax2-provision.c, channels/iax2-parser.c, main/chanvars.c, res/res_monitor.c, main/netsock.c, apps/app_speech_utils.c, channels/chan_misdn.c, funcs/func_curl.c, main/fixedjitterbuf.c, apps/app_macro.c, res/res_indications.c, apps/app_mixmonitor.c, main/asterisk.c, res/res_odbc.c, main/dlfcn.c, apps/app_voicemail.c, channels/chan_vpb.cc, apps/app_meetme.c, main/utils.c, res/res_musiconhold.c, channels/chan_gtalk.c, cdr/cdr_pgsql.c, apps/app_followme.c, codecs/codec_zap.c, res/res_jabber.c, res/res_config_sqlite.c, main/enum.c, channels/misdn_config.c, main/io.c, main/channel.c, main/cdr.c, funcs/func_enum.c, main/dial.c, main/manager.c, apps/app_osplookup.c, main/tdd.c, funcs/func_odbc.c, cdr/cdr_sqlite.c, res/res_agi.c, apps/app_minivm.c, main/app.c, apps/app_directory.c, apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c, codecs/codec_lpc10.c, res/res_config_pgsql.c, channels/chan_zap.c, main/dnsmgr.c, channels/chan_sip.c, apps/app_festival.c, main/translate.c, main/jitterbuf.c, main/acl.c, apps/app_queue.c, channels/chan_oss.c, main/rtp.c, cdr/cdr_tds.c, main/file.c, main/callerid.c, main/event.c, funcs/func_devstate.c, funcs/func_callerid.c, main/dsp.c: Issue 9869 - replace malloc and memset with ast_calloc, and other coding guidelines changes 2007-06-06 21:16 +0000 [r67813-67863] Russell Bryant * /, channels/chan_sip.c: Merged revisions 67862 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67862 | russell | 2007-06-06 16:14:46 -0500 (Wed, 06 Jun 2007) | 4 lines Fix a crash when doing call pickups with SIP phones. The code unlocked the channel when it should not have. (issue #9652, reported by corruptor, fixed by me) ........ * res/res_features.c, include/asterisk/features.h: Constify the return values of ast_parking_ext() and ast_pickup_ext() * main/manager.c: Minor formatting change to test closing mantis issues through commit tags (closes issue #9828) * main/manager.c: Minor formatting change to test closing mantis issues through commit tags (closes issue #9828) * apps/app_voicemail.c: Please forgive this flood of tiny changes ... this will be cool when it works how we want it to :) (testing mantis+svn) (issue #9828) 2007-06-06 19:46 +0000 [r67808] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 67804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67804 | mmichelson | 2007-06-06 14:26:55 -0500 (Wed, 06 Jun 2007) | 10 lines Fix for Issue 9810. There was a segfault under a specific set of circumstances: 1. VoiceMailMain was configured in the dialplan with an extension as its argument 2. A message was left for this mailbox 3. Tried to call VoiceMailMain but hung up before entering password. This was fixed by checking that a pointer was non-null prior to trying to dereference it. (Issue 9810, reported by xmarksthespot, patched by Corydon76 with modifications by me). ........ 2007-06-06 19:44 +0000 [r67787-67807] Russell Bryant * apps/app_voicemail.c: minor formatting change ... testing mantis/svn (issue #9828) * apps/app_voicemail.c: Don't try to check the result of alloca ... ... testing mantis/svn stuff ... (issue #9828) * main/dsp.c: Yet another minor change to test mantis/svn (issue #9828) * main/dsp.c: minor formatting change ... testing mantis/svn (issue #9828) * main/dsp.c: minor formatting change ... testing mantis/svn (issue #9828) * main/app.c: Formatting change ... testing (issue #9828) 2007-06-06 19:02 +0000 [r67784] Mark Michelson * apps/app_voicemail.c: Fixing a crash wherein Asterisk would segfault when attempting to leave a voicemail when IMAP storage was enabled. Though no bug was reported to the bugtracker, there was mention of this made as a note on bug 9810 by edhorton. 2007-06-06 19:00 +0000 [r67697-67782] Russell Bryant * main/app.c: Make another formatting change ... testing mantis/svn stuff (issue #9828) * main/app.c: Another minor formatting change ... testing mantis/svn (issue #9828) * main/app.c: Minor formatting change ... testing mantis/svn (issue #9828) * channels/chan_iax2.c: Make another small tweak ... mantis/svn testing (issue #9828) * res/res_features.c: Another tiny formatting change for testing ... (issue #9828) * main/app.c: More random formatting changes to test Mantis/SVN integration (issue #9828) * main/app.c: Make a completely arbitrary formatting change to test out some Mantis/SVN integration stuff. (issue #9828) * main/channel.c, /, include/asterisk/linkedlists.h: Merged revisions 67716 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r67716 | russell | 2007-06-06 11:55:59 -0500 (Wed, 06 Jun 2007) | 13 lines Merged revisions 67715 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r67715 | russell | 2007-06-06 11:40:51 -0500 (Wed, 06 Jun 2007) | 5 lines We have some bug reports showing crashes due to a double free of a channel. Add a sanity check to ast_channel_free() to make sure we don't go on trying to free a channel that wasn't found in the channel list. (issue #8850, and others...) ........ ................ * res/res_features.c: Change "show parkedcalls" to "parkedcalls show" and mark the previous command as deprecated. Also, convert the CLI command to the new style. (issue #9861, patch from eliel) 2007-06-06 13:32 +0000 [r67595-67651] Joshua Colp * /, main/rtp.c: Merged revisions 67650 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r67650 | file | 2007-06-06 09:30:25 -0400 (Wed, 06 Jun 2007) | 10 lines Merged revisions 67649 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r67649 | file | 2007-06-06 09:28:34 -0400 (Wed, 06 Jun 2007) | 2 lines Reinvite the RTP back to the Asterisk machine when the timeout happens. (issue #9888 reported by gasparz) ........ ................ * /, main/translate.c: Merged revisions 67631 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67631 | file | 2007-06-06 09:18:39 -0400 (Wed, 06 Jun 2007) | 2 lines Fix plc_samples warning when registering a translator. (issue #9897 reported by xylome) ........ * /, apps/app_directed_pickup.c: Merged revisions 67626 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67626 | file | 2007-06-06 09:16:34 -0400 (Wed, 06 Jun 2007) | 2 lines Include macroexten while searching for a channel to pick up in case they are in a macro. (issue #9491 reported by jamesb63) ........ * /, res/res_agi.c: Merged revisions 67597 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67597 | file | 2007-06-06 08:34:06 -0400 (Wed, 06 Jun 2007) | 2 lines Make the new "agi debug off" CLI command work. (issue #9890 reported by eliel) ........ * channels/chan_zap.c: When SS7 is enabled add w/SS7 to the end. (issue #9893 reported by Mike Anikienko) * /, main/devicestate.c: Merged revisions 67594 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r67594 | file | 2007-06-06 08:20:27 -0400 (Wed, 06 Jun 2007) | 10 lines Merged revisions 67593 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r67593 | file | 2007-06-06 08:18:36 -0400 (Wed, 06 Jun 2007) | 2 lines Revert channel name splitting fix for Zap. The moral of the story is don't use - in your user/peer names. (issue #9668 reported by stevedavies) ........ ................ 2007-06-05 23:02 +0000 [r67560] Russell Bryant * /, apps/app_meetme.c: Merged revisions 67558 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67558 | russell | 2007-06-05 18:01:44 -0500 (Tue, 05 Jun 2007) | 5 lines Fix some crashes related to the use of the "meetme" CLI command. The code for this command was not locking the conference list at all. (issue #9351, reported by and patch submitted by Junk-Y, committed patch is different and by me) ........ 2007-06-05 22:59 +0000 [r67557] Mark Michelson * main/cli.c: Found a bug where when "core set debug #" is used, the verbosity is read as the old value instead of the old debug value, leading to an erroneous status message after setting. This was purely a cosmetic issue and had no other underlying problems. 2007-06-05 22:04 +0000 [r67529] Steve Murphy * utils/Makefile, /, pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/pbx_ael.c, pbx/Makefile: Merged revisions 67526 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67526 | murf | 2007-06-05 15:30:18 -0600 (Tue, 05 Jun 2007) | 1 line this fixes bug 9883, wherein macros were not allowing the includes construct. fixed and tested, looks OK. Now includes can serve as an adjunct to catch. ........ 2007-06-05 20:55 +0000 [r67493] Russell Bryant * /, include/asterisk/linkedlists.h: Merged revisions 67492 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67492 | russell | 2007-06-05 15:53:28 -0500 (Tue, 05 Jun 2007) | 16 lines This bug has been hanging over my head ever since I wrote this SLA code. Every time I tried to go debug it by adding some debug output, the behavior would change. It turns out I wasn't crazy. I had the following piece of code: if (remove) AST_LIST_REMOVE_CURRENT(...); Well, AST_LIST_REMOVE_CURRENT was not wrapped in braces, so my conditional statement didn't do much good at all. It always ran at least all of the macro minus the first statement, so I was seeing list entries magically disappear when they weren't supposed to. After many hours of debugging, I have come to this extremely irritating fix. :) (issues #9581, #9497) ........ 2007-06-05 20:16 +0000 [r67486] Mark Michelson * apps/app_voicemail.c: Merged revisions 67424 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67424 | mmichelson | 2007-06-05 13:32:50 -0500 (Tue, 05 Jun 2007) | 5 lines Fix for bug number 9786, wherein voicemails saved to IMAP storage using extensions other than gsm were unable to be played over the phone. (Issue 9786, reporter: xmarksthespot, Patched by xmarksthe spot with revisions by me, reviewed by Russell Bryant). ........ 2007-06-05 19:50 +0000 [r67458] Russell Bryant * channels/chan_zap.c, /: Merged revisions 67457 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67457 | russell | 2007-06-05 14:48:02 -0500 (Tue, 05 Jun 2007) | 2 lines Suppress a bunch of debug output unless option_debug is on ........ 2007-06-05 18:23 +0000 [r67423] Steve Murphy * /, pbx/pbx_ael.c: Merged revisions 67420 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67420 | murf | 2007-06-05 12:17:28 -0600 (Tue, 05 Jun 2007) | 1 line Added code to automatically add a default case to switches that don't have one. In some cases, rather than fall thru, it results in a goto with -1 result, which terminates the extension; a sort of dialplan seqfault, sort of. This was required to fix bug reported in 9881 ........ 2007-06-05 18:19 +0000 [r67398-67422] Jason Parker * /, channels/chan_skinny.c: Merged revisions 67421 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67421 | qwell | 2007-06-05 13:18:24 -0500 (Tue, 05 Jun 2007) | 4 lines Correctly update date/time on devices throughout the life of the device, instead of just at registration. Issue 9152, yet another patch by DEA. ........ * main/manager.c: Make sure we default allowmultiplelogin to on/yes, per the default stated in the config. Issue 9885, patch by eliel. 2007-06-05 17:24 +0000 [r67397] Dwayne M. Hubbard * channels/misdn/isdn_msg_parser.c: changed #if DEBUG to #ifdef DEBUG to fix make failure when configured with --enable-dev-mode 2007-06-05 17:11 +0000 [r67361-67380] Russell Bryant * channels/chan_zap.c: Improve the way that the zaptel channel name is created by using the Asterisk strings API and by only allocating space on the stack * /: Merged revisions 67360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67360 | russell | 2007-06-05 11:56:36 -0500 (Tue, 05 Jun 2007) | 5 lines Fix a problem that showed itself by causing Zap channel names to be completely bogus on my machine. ast_safe_string_alloc() was broken. It called vsnprintf() on a va_args list twice without re-initializing it. After the first usage, va_end() and va_start() must be called again. ........ 2007-06-05 16:21 +0000 [r67345-67350] Christian Richter * /, channels/misdn/chan_misdn_config.h: Merged revisions 67334 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r67334 | crichter | 2007-06-05 18:14:07 +0200 (Di, 05 Jun 2007) | 9 lines Merged revisions 67307 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r67307 | crichter | 2007-06-05 17:42:03 +0200 (Di, 05 Jun 2007) | 1 line briding is a bool, fixed copy and paste issue. ........ ................ * channels/chan_misdn.c, /: Merged revisions 67329 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r67329 | crichter | 2007-06-05 18:11:57 +0200 (Di, 05 Jun 2007) | 9 lines Merged revisions 67306 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r67306 | crichter | 2007-06-05 17:39:43 +0200 (Di, 05 Jun 2007) | 1 line simplified the EVENT_SETUP handling in the cb_events function a lot. Commented the different possibilities a bit and made functions of shared code. When the dialed extension does not exist in the extensions.conf we'll jump into the 'i' extension if this does exist, else we disconnect the call with the cause:1 = No Route to Destination. ........ ................ 2007-06-05 15:54 +0000 [r67310] Russell Bryant * /, include/asterisk/module.h, main/asterisk.c, main/loader.c: Merged revisions 67308 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67308 | russell | 2007-06-05 10:51:53 -0500 (Tue, 05 Jun 2007) | 5 lines When shutting down "gracefully", go through and run the unload() callbacks for all of the modules. "stop now" is considered a non-graceful shutdown and will not go through this process. (issue #9804, reported by chrisost, patch by me) ........ 2007-06-05 15:24 +0000 [r67305] Joshua Colp * /, channels/chan_iax2.c: Merged revisions 67304 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67304 | file | 2007-06-05 12:22:30 -0300 (Tue, 05 Jun 2007) | 2 lines Only muck with the thread structure if an idle one was found/created. ........ 2007-06-05 14:59 +0000 [r67272-67273] Russell Bryant * doc/CODING-GUIDELINES: add a note about inline comments * channels/chan_iax2.c: Doxygenify the comments for new members of the iax2_thread struct 2007-06-05 14:45 +0000 [r67271] Kevin P. Fleming * /, channels/chan_iax2.c: Merged revisions 67270 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67270 | kpfleming | 2007-06-05 09:35:52 -0500 (Tue, 05 Jun 2007) | 3 lines ensure that a burst of full frames (AST_FRAME_DTMF being the prime example) will not be processed out of order... this is a brute force fix, but seems to be the safest fix for now (thanks to the Digium PQ department for finding this bug) ........ 2007-06-05 11:48 +0000 [r67240] Christian Richter * channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h, channels/misdn_config.c: Merged revisions 67210 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r67210 | crichter | 2007-06-05 12:25:32 +0200 (Di, 05 Jun 2007) | 9 lines Merged revisions 67209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r67209 | crichter | 2007-06-05 12:05:45 +0200 (Di, 05 Jun 2007) | 1 line added possibility to deactivate bridging per port ........ ................ 2007-06-04 23:45 +0000 [r67164] Tilghman Lesher * /, funcs/func_math.c: Merged revisions 67162 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r67162 | tilghman | 2007-06-04 18:43:01 -0500 (Mon, 04 Jun 2007) | 10 lines Merged revisions 67161 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r67161 | tilghman | 2007-06-04 18:41:49 -0500 (Mon, 04 Jun 2007) | 2 lines According to MATH, 0+1181000386 = 1181000448. Oops. ........ ................ 2007-06-04 23:32 +0000 [r67160] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 67158 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67158 | russell | 2007-06-04 18:31:40 -0500 (Mon, 04 Jun 2007) | 5 lines Fix up a bunch of places where the iax2 pvt structure can disappear and the code did not account for it and crashes. (issues #9642, #9569, #9666, probably others ... based on the work by stevedavies and mihai, with additional changes from me) ........ 2007-06-04 23:29 +0000 [r67122-67157] Jason Parker * /, channels/chan_skinny.c: Merged revisions 67156 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67156 | qwell | 2007-06-04 18:26:28 -0500 (Mon, 04 Jun 2007) | 6 lines Fix for skinny keepalives. If there is no traffic from the phone for (keep_alive * 1100) ms (arbitrarily adding 10% for network issues, etc), unregister the device. Issue 8394, patch by DEA. ........ * /, channels/chan_mgcp.c: Merged revisions 67121 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67121 | qwell | 2007-06-04 17:36:57 -0500 (Mon, 04 Jun 2007) | 4 lines Fixes for dtmf/dialing with mgcp (similar to the recent fix for chan_skinny) Issue 9855, patch by DEA. ........ 2007-06-04 22:29 +0000 [r67120] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 67119 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67119 | russell | 2007-06-04 17:28:55 -0500 (Mon, 04 Jun 2007) | 6 lines Add comments for two functions that get called with the appropriate call locked, but perform operations that could result in the pvt structure getting destroyed before returning again, causing numerous seg faults all over the module. (inspired by issues #9642, #9569, and #9666, and the work done by stevedavies and mihai) ........ 2007-06-04 22:15 +0000 [r67095] Steve Murphy * main/cdr.c, /: Merged revisions 67073 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67073 | murf | 2007-06-04 15:59:34 -0600 (Mon, 04 Jun 2007) | 1 line This typo has been here since 1.4 forked. It has been the source of heartburn to many a dialplan/CDR programmer. ........ 2007-06-04 21:48 +0000 [r67070-67072] Russell Bryant * /, main/rtp.c: Merged revisions 67071 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67071 | russell | 2007-06-04 16:47:36 -0500 (Mon, 04 Jun 2007) | 2 lines Add a missing \n. (pointed out by jcmoore on IRC) ........ * channels/chan_iax2.c: Remove a leftover unlock and lock of the iax2 pvt struct lock that was left over from my attempt at putting pvt structs in a hash table. It can cause seg faults, and has no reason to stay. (issue #9642, pointed out by stevedavies) 2007-06-04 19:32 +0000 [r67063-67069] Joshua Colp * /, channels/chan_sip.c: Merged revisions 67068 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67068 | file | 2007-06-04 15:31:09 -0400 (Mon, 04 Jun 2007) | 2 lines Better handle SIP devices that say they have SDP content... but really don't. (issue #9398 reported by mthomasslo) ........ * apps/app_dial.c, /: Merged revisions 67066 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67066 | file | 2007-06-04 13:59:14 -0400 (Mon, 04 Jun 2007) | 2 lines Initialize cidname variable to nothing since it may be used without having been touched. (issue #9661 reported by dimas) ........ * /, res/res_features.c: Merged revisions 67064 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67064 | file | 2007-06-04 13:41:59 -0400 (Mon, 04 Jun 2007) | 2 lines Returning a value that indicates the parking of a call was a success when it really wasn't (because the parking slot selected was in use) is the wrong thing to do. (issue #9723 reported by mdu113) ........ * apps/app_directed_pickup.c: Minor clean up. Constify a few variables and use ast_strlen_zero in a few places. 2007-06-04 17:12 +0000 [r67062] Tilghman Lesher * contrib/init.d/rc.debian.asterisk, contrib/init.d/rc.mandrake.asterisk, /, contrib/init.d/rc.redhat.asterisk, contrib/init.d/rc.gentoo.asterisk, contrib/init.d/rc.mandrake.zaptel, contrib/init.d/rc.slackware.asterisk: Merged revisions 67061 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r67061 | tilghman | 2007-06-04 12:11:43 -0500 (Mon, 04 Jun 2007) | 10 lines Merged revisions 67060 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r67060 | tilghman | 2007-06-04 12:10:30 -0500 (Mon, 04 Jun 2007) | 2 lines Add revision Id tags (by request of tzafrir) ........ ................ 2007-06-04 16:03 +0000 [r67024-67029] Russell Bryant * /, configure, configure.ac: Merged revisions 67026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67026 | russell | 2007-06-04 11:02:31 -0500 (Mon, 04 Jun 2007) | 6 lines Change the configure script to build a test program against libcurl to make sure the results from curl-config can be used to compile successfully. This is intended to help prevent a situation where you are cross compiling, and the configure script finds the curl library installed on the host. (issue #9865, reported and patched by zandbelt) ........ * main/ast_expr2f.c, pbx/ael/ael_lex.c, main/app.c: Change javadoc style code documentation to the same format we use elsewhere. (issue #9864, patch from snuffy) 2007-06-04 15:53 +0000 [r67023] Tilghman Lesher * /, res/res_jabber.c: Merged revisions 67021 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67021 | tilghman | 2007-06-04 10:50:16 -0500 (Mon, 04 Jun 2007) | 2 lines Issue 9739 - Malformed jid causes a crash ........ 2007-06-04 15:50 +0000 [r67016-67022] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 67020 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67020 | russell | 2007-06-04 10:47:40 -0500 (Mon, 04 Jun 2007) | 7 lines Resolve a deadlock in chan_iax2. When handling an implicit ACK to a frame that was marked as the final transmission for a call, don't call iax2_destroy() for that call while the global frame queue is still locked. There is a very nice explanation of the deadlock in the report. (issue #9663, thorough report and patch from stevedavies, additional positive test reports from mihai and joff_oconnell) ........ * include/asterisk/stringfields.h: Fix some compiler warnings in C++ modules. (issue #9866, reported by osk, patch by Corydon76) * channels/chan_sip.c, main/netsock.c: Fix a couple of places where "tos" was used instead of "cos". (issue #9540, patch by IgorG) 2007-06-04 11:48 +0000 [r66998] Joshua Colp * apps/app_mixmonitor.c: Add support for autocompleting start/stop options of the mixmonitor CLI command. (issue #9862 reported by eliel) 2007-06-03 06:10 +0000 [r66981] Tilghman Lesher * channels/chan_jingle.c, channels/chan_phone.c, channels/chan_features.c, channels/chan_h323.c, channels/chan_gtalk.c, channels/chan_nbs.c, channels/chan_mgcp.c: ast_calloc janitor (Inspired by issue 9860) 2007-06-01 23:39 +0000 [r66957-66959] Russell Bryant * main/pbx.c: remove a bogus comment that came from copy/paste * include/asterisk/devicestate.h, include/asterisk.h, main/pbx.c, include/asterisk/event_defs.h, main/devicestate.c, include/asterisk/pbx.h, apps/app_queue.c, main/asterisk.c: Merge major changes to the way device state is passed around Asterisk. The two places that cared about device states were app_queue and the hint code in pbx.c. The changes include converting it to use the Asterisk event system, as well as other efficiency improvements. * app_queue: This module used to register a callback into devicestate.c to monitor device state changes. Now, it is just a subscriber to Asterisk events with the type, device state. * pbx.c hints: Previously, the device state processing thread in devicestate.c would call ast_hint_state_changed() each time the state of a device changed. Then, that code would go looking for all the hints that monitor that device, and call their callbacks. All of this blocked the device state processing thread. Now, the hint code is a subscriber of Asterisk events with the type, device state. Furthermore, when this code receives a device state change event, it queues it up to be processed by another thread so that it doesn't block one of the event processing threads. * channels/chan_iax2.c: Remove 80 bytes in the iax2_registry struct that weren't being used 2007-06-01 21:49 +0000 [r66920] Tilghman Lesher * /, funcs/func_odbc.c: Merged revisions 66919 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r66919 | tilghman | 2007-06-01 16:45:44 -0500 (Fri, 01 Jun 2007) | 2 lines On some drivers, deallocating the statement handle isn't enough. We also have to clear the cursor (nice, Oracle) ........ 2007-06-01 21:33 +0000 [r66910-66918] Mark Michelson * /: Merged revisions 66916 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ ........ * /, apps/app_voicemail.c: Merged revisions 66897 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r66897 | mmichelson | 2007-06-01 16:09:30 -0500 (Fri, 01 Jun 2007) | 3 lines Submitting a fix for voicemail with IMAP storage. Attachments with format specified as gsm were duplicated (i.e. two attachments) were left. Thank you very much to xmarksthespot for submitting the patch that fixed this. (Issues 9787 and 8873, Reported by xmarksthespot and jerjer, patched by xmarksthespot) ........ 2007-06-01 19:42 +0000 [r66880-66882] Russell Bryant * /, channels/chan_skinny.c: Merged revisions 66881 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r66881 | russell | 2007-06-01 14:41:30 -0500 (Fri, 01 Jun 2007) | 6 lines Changes to the way DTMF is handled in the core broke dialing in chan_skinny. This patch makes chan_skinny usable again. I did not end up testing this, but there are multiple positive test reports listed in the bug report. (issue #9596, reported by pj, testing by pj and mvanbaak, and the fix was written by DEA) ........ * /, apps/app_page.c: Merged revisions 66879 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r66879 | russell | 2007-06-01 14:35:13 -0500 (Fri, 01 Jun 2007) | 2 lines List app_meetme as a module that app_page depends on. ........ 2007-06-01 18:36 +0000 [r66878] Jason Parker * res/res_config_sqlite.c: Documentation fixes for res_config_sqlite. Issue 9854, patch by tzafrir. 2007-06-01 13:48 +0000 [r66856] Russell Bryant * configs/sip.conf.sample: Add some more information about the SIP Disclaimer header. 2007-05-31 23:04 +0000 [r66822] Tilghman Lesher * /, doc/asterisk.8: Merged revisions 66821 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r66821 | tilghman | 2007-05-31 18:03:28 -0500 (Thu, 31 May 2007) | 2 lines Issue 9850 - update preferred command line syntax ........ 2007-05-31 21:23 +0000 [r66772-66818] Russell Bryant * configs/sip.conf.sample: fix a typo. * channels/chan_sip.c, configs/sip.conf.sample: To satisfy some legal concerns, add an option for chan_sip to include a disclaimer along with SIP messages in the header, X-Disclaimer. This is off by default. Also, the text of the disclaimer can be customized in sip.conf. * include/asterisk/app.h, /, include/asterisk/speech.h, res/res_speech.c: Merged revisions 66775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r66775 | russell | 2007-05-31 13:41:58 -0500 (Thu, 31 May 2007) | 3 lines Change a couple of header files to not use "new", which is a reserved keyword in C++. (issue #9830, reported by osk) ........ * res/res_features.c, CHANGES, configs/features.conf.sample: Add support for configuring named groups of custom call features in features.conf. This allows you to create a feature one time, and then map it into groups for various different key mappings for the same feature, as well as easy access control to groups of features. (patch from bbryant) * res/res_features.c, configs/features.conf.sample: Revert changes that snuck in with revision 66724. * apps/app_minivm.c: - Don't check if the list is empty needlessly - Don't free structures before calling load_config(), because load_config() already does it - Use the existing functions for freeing the minivm structures instead of replicating the code (issue #9846, patch from eliel) 2007-05-31 17:16 +0000 [r66771] Tilghman Lesher * /, apps/app_macro.c: Merged revisions 66770 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r66770 | tilghman | 2007-05-31 12:15:09 -0500 (Thu, 31 May 2007) | 10 lines Merged revisions 66744 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r66744 | tilghman | 2007-05-31 10:58:45 -0500 (Thu, 31 May 2007) | 2 lines Issue 9818 - Fix for issue 8329 breaks pbx_realtime. Issue 8329 will remain unfixed for pbx_realtime, but only because we lack core API to do it. ........ ................ 2007-05-31 16:18 +0000 [r66769] Joshua Colp * /, channels/chan_sip.c: Merged revisions 66768 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r66768 | file | 2007-05-31 12:14:48 -0400 (Thu, 31 May 2007) | 10 lines Merged revisions 66764 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r66764 | file | 2007-05-31 12:12:39 -0400 (Thu, 31 May 2007) | 2 lines It is now possible for this path of execution to have the frame pointer be NULL, therefore we need to check for it before trying to access it. (issue #9836 reported by barthpbx) ........ ................ 2007-05-31 15:05 +0000 [r66734] Tilghman Lesher * configs/func_odbc.conf.sample, funcs/func_odbc.c: Issue 9799 - Multirow results for func_odbc 2007-05-31 14:52 +0000 [r66724] Russell Bryant * res/res_features.c, apps/app_minivm.c, configs/features.conf.sample: Fix a crash on reload by using calloc() instead of malloc() to ensure that data is properly initialized. (issue #9765, reported by MatsK, patch from eliel) 2007-05-31 10:26 +0000 [r66705] Olle Johansson * include/asterisk/app.h, apps/app_osplookup.c, include/asterisk/event.h, apps/app_meetme.c, channels/chan_sip.c, include/asterisk/event_defs.h, apps/app_skel.c, apps/app_minivm.c, res/res_jabber.c: Issue #9842 - Doxygen updates by snuffy. Thanks! (Committed from Media Plaza in Utrecht, Netherlands - Open Source VoIP conference) 2007-05-30 23:44 +0000 [r66672] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 66671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r66671 | mmichelson | 2007-05-30 18:26:39 -0500 (Wed, 30 May 2007) | 2 lines Fixed seg-faults when recording greetings in voicemail with IMAP enabled. (Issue No. 9734, reported by xmarksthespot, patched by me) ........ 2007-05-30 17:23 +0000 [r66603-66638] Joshua Colp * channels/chan_zap.c, channels/chan_features.c: This concludes my tweaking of things. 2007-05-30 05:17 +0000 [r66539-66585] Tilghman Lesher * apps/app_channelredirect.c, channels/chan_vpb.cc, res/res_config_odbc.c, funcs/func_shell.c, funcs/func_cdr.c, apps/app_zapras.c, res/res_indications.c, apps/app_transfer.c, apps/app_stack.c, funcs/func_devstate.c, res/res_config_sqlite.c, res/res_odbc.c: Issue 9477 - Improve menuselect labels * /, funcs/func_strings.c: Merged revisions 66538 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r66538 | tilghman | 2007-05-29 16:56:07 -0500 (Tue, 29 May 2007) | 10 lines Merged revisions 66537 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r66537 | tilghman | 2007-05-29 16:49:35 -0500 (Tue, 29 May 2007) | 2 lines If the value of a variable passed to FIELDQTY is blank, then FIELDQTY should return 0, not 1. ........ ................ * funcs/func_enum.c: Shorten description to a much more reasonable length 2007-05-29 19:53 +0000 [r66502-66505] Olle Johansson * channels/chan_sip.c: oops. Thanks Terry. * /, channels/chan_sip.c: Merged revisions 66503 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r66503 | oej | 2007-05-29 21:32:57 +0200 (Tue, 29 May 2007) | 2 lines Properly handle 408 request timeout - according to the RFC, the dialog dies if a request in a dialog gets this response. ........ * /, channels/chan_sip.c: Merged revisions 66474 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r66474 | oej | 2007-05-29 21:02:04 +0200 (Tue, 29 May 2007) | 2 lines Don't issue hangup on hangup on hangup on hangup (for jcmoore) ........ 2007-05-29 19:00 +0000 [r66471] Doug Bailey * main/dsp.c: Changed the dtmf detection to integer based goertzel algorithm. 2007-05-29 16:46 +0000 [r66438] Joshua Colp * /, main/rtp.c: Merged revisions 66437 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r66437 | file | 2007-05-29 12:44:34 -0400 (Tue, 29 May 2007) | 2 lines Handle cases where a frame may have no data. (issue #9519 reported by dmb) ........ 2007-05-29 16:19 +0000 [r66432-66433] Olle Johansson * /, channels/chan_sip.c: Merged revisions 66414 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r66414 | oej | 2007-05-29 18:07:44 +0200 (Tue, 29 May 2007) | 2 lines Don't reset hangupcause if we already have one ........ * /, channels/chan_sip.c: Merged revisions 66404 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r66404 | oej | 2007-05-29 18:02:50 +0200 (Tue, 29 May 2007) | 2 lines Tracking down hanging channels, killing them one by one. Issue #9235 and related ........ 2007-05-29 15:44 +0000 [r66399] Joshua Colp * /, doc/datastores.txt: Merged revisions 66398 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r66398 | file | 2007-05-29 11:43:16 -0400 (Tue, 29 May 2007) | 2 lines Update datastores documentation. (issue #9801 reported by mnicholson) ........ 2007-05-29 10:02 +0000 [r66367] Olle Johansson * /, channels/chan_sip.c: Merged revisions 66363 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r66363 | oej | 2007-05-29 11:41:40 +0200 (Tue, 29 May 2007) | 10 lines Merged revisions 66349 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r66349 | oej | 2007-05-29 09:53:14 +0200 (Tue, 29 May 2007) | 2 lines Issue #9802 - Change inuse counter on CANCEL ........ ................ 2007-05-28 23:28 +0000 [r66313-66315] Joshua Colp * channels/chan_sip.c: Don't try to unregister a peer using the sip unregister CLI command if they are not registered. (issue #9811 reported by eliel) * channels/chan_sip.c: Due to the way stringfields work the value of the url pointer will always be non-NULL so we have to use ast_strlen_zero to make sure it is not empty. (issue #9821 reported by pj) 2007-05-28 18:50 +0000 [r66295] Olle Johansson * apps/app_voicemail.c: - Don't re-invent existing headers (some already existed in chan_sip) - Rename command so taht module name comes first 2007-05-28 15:59 +0000 [r66278] Tilghman Lesher * funcs/func_iconv.c (added): Issue 7021 - Add ICONV function for converting between character sets 2007-05-26 19:35 +0000 [r66225] Joshua Colp * apps/app_minivm.c: Unlock the minivmlock when no configuration is found. (issue #9814 reported by eliel) 2007-05-26 06:07 +0000 [r66208] Russell Bryant * apps/app_meetme.c: Since this code now uses the API call for creating a detached thread, there is no reason to keep a thread attribute structure on the conference structure. (Pointed out by Tony Mountifield on the asterisk-dev list) 2007-05-25 15:08 +0000 [r66175-66178] Kevin P. Fleming * /: block change that is already here * channels/chan_jingle.c, configure, configure.ac: more minor fixes 2007-05-25 14:49 +0000 [r66161] Tilghman Lesher * /, main/say.c: Merged revisions 66159 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r66159 | tilghman | 2007-05-25 09:41:27 -0500 (Fri, 25 May 2007) | 10 lines Merged revisions 66127 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r66127 | tilghman | 2007-05-25 08:46:35 -0500 (Fri, 25 May 2007) | 2 lines Issue 9791 - Fix pronunciation of seconds in Dutch ........ ................ 2007-05-25 14:37 +0000 [r66158] Kevin P. Fleming * channels/chan_jingle.c, /, configure, configure.ac, channels/chan_gtalk.c, makeopts.in, res/res_jabber.c: Merged revisions 66157 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r66157 | kpfleming | 2007-05-25 10:28:46 -0400 (Fri, 25 May 2007) | 3 lines handle the GNUTLS library properly in the configure script and build system don't build in OSP support unless we have found and are allowed to use SSL support ........ 2007-05-25 13:26 +0000 [r66109-66126] Joshua Colp * main/slinfactory.c: Minor tweak... drop translation path if one exists when we get an already signed linear frame in. Chances are the stream has then switched to signed linear and we no longer need the path. * /, main/slinfactory.c: Merged revisions 66074 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r66074 | file | 2007-05-24 18:16:58 -0400 (Thu, 24 May 2007) | 2 lines Fix slinfactory logic when dealing with frames coming in that may already be in the signed linear format. ........ 2007-05-24 22:25 +0000 [r66072-66077] Russell Bryant * main/channel.c, /: Merged revisions 66076 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r66076 | russell | 2007-05-24 17:23:59 -0500 (Thu, 24 May 2007) | 1 line if the string field init fails, clean up the stuff that was allocated already ........ * main/channel.c, /: Merged revisions 66070 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r66070 | russell | 2007-05-24 17:07:39 -0500 (Thu, 24 May 2007) | 2 lines Check the result of ast_string_field_init() in ast_channel_alloc() ........ 2007-05-24 22:07 +0000 [r66071] Kevin P. Fleming * main/aescrypt.c, build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, include/asterisk/aes_internal.h (added), configure.ac, main/aestab.c, include/asterisk/aes.h, main/aeskey.c, pbx/pbx_dundi.c, channels/chan_iax2.c, makeopts.in: use the OpenSSL AES implementation if it's available (unless configured not to) 2007-05-24 20:55 +0000 [r66031] Jason Parker * /, configure, configure.ac: Merged revisions 66029-66030 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r66029 | qwell | 2007-05-24 15:53:18 -0500 (Thu, 24 May 2007) | 2 lines Following moving strip to AC_PATH_TOOL, we need to do something similar for ar. ........ r66030 | qwell | 2007-05-24 15:54:16 -0500 (Thu, 24 May 2007) | 2 lines Rebuild configure script for previous ar fix. ........ 2007-05-24 20:51 +0000 [r66028] Joshua Colp * CHANGES, apps/app_voicemail.c: Add ListAllVoicemailUsers manager command. (issue #8112 reported by Tony Zhao) 2007-05-24 20:44 +0000 [r65982-66027] Russell Bryant * /, configure, configure.ac: Merged revisions 66026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r66026 | russell | 2007-05-24 15:42:53 -0500 (Thu, 24 May 2007) | 3 lines Checking for the strip application needs to be done with AC_PATH_TOOL instead of AC_PATH_PROG to properly handle cross compilation environments. ........ * doc/CODING-GUIDELINES: add a note about using the intenal API for creating detached threads * Makefile, /: Merged revisions 65978 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65978 | russell | 2007-05-24 14:05:08 -0500 (Thu, 24 May 2007) | 3 lines Clear CFLAGS before running make for menuselect. (issue #9784, reported by ovi, patch by me) ........ 2007-05-24 19:05 +0000 [r65979] Kevin P. Fleming * /, channels/chan_gtalk.c: Merged revisions 65965-65967 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65965 | kpfleming | 2007-05-24 14:24:55 -0400 (Thu, 24 May 2007) | 2 lines don't use uninitialized variables ........ r65966 | kpfleming | 2007-05-24 14:25:21 -0400 (Thu, 24 May 2007) | 2 lines don't reference GnuTLS headers and functions unless the configure script found it ........ r65967 | kpfleming | 2007-05-24 14:28:48 -0400 (Thu, 24 May 2007) | 2 lines oops, use #ifdef instead of #if ........ 2007-05-24 18:30 +0000 [r65964-65968] Russell Bryant * main/pbx.c, include/asterisk/utils.h, channels/chan_zap.c, channels/chan_sip.c, apps/app_meetme.c, main/utils.c, channels/chan_iax2.c, main/cdr.c, main/manager.c, pbx/pbx_spool.c, channels/chan_skinny.c, main/http.c, channels/chan_h323.c, pbx/pbx_dundi.c, apps/app_rpt.c, apps/app_mixmonitor.c, main/asterisk.c, channels/chan_mgcp.c: Add a new API call for creating detached threads. Then, go replace all of the places in the code where the same block of code for creating detached threads was replicated. (patch from bbryant) * main/rtp.c: Make this build on *my* machine again, and hopefully not break others. 2007-05-24 15:35 +0000 [r65906] Dwayne M. Hubbard * /, funcs/func_math.c: Merged revisions 65866 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65866 | dhubbard | 2007-05-24 10:08:56 -0500 (Thu, 24 May 2007) | 1 line merged qwell's func_math patch for issue 9507 ........ 2007-05-24 15:30 +0000 [r65905] Joshua Colp * main/manager.c, /: Merged revisions 65902 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65902 | file | 2007-05-24 11:27:23 -0400 (Thu, 24 May 2007) | 2 lines Add the ability to blacklist certain commands from being executed using the Command AMI action. (issue #9240 reported by junky) ........ 2007-05-24 15:29 +0000 [r65904] Olle Johansson * /, channels/chan_gtalk.c: Merged revisions 65901 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65901 | oej | 2007-05-24 17:26:10 +0200 (Thu, 24 May 2007) | 2 lines Issue 7672 - fix by zandbelt - Asterisk core dump since the GnuTLS interface did not support multithreading correctly. ........ 2007-05-24 15:28 +0000 [r65903] Jason Parker * /, codecs/codec_speex.c, main/translate.c, codecs/codec_ilbc.c, .cleancount, include/asterisk/translate.h: Merged revisions 65877 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65877 | qwell | 2007-05-24 11:14:02 -0400 (Thu, 24 May 2007) | 4 lines Fix handling of zero-length frames when a codec is capable of native PLC. Issue 9183, patch by Mihai. ........ 2007-05-24 15:23 +0000 [r65894-65898] Olle Johansson * /, channels/chan_gtalk.c: Merged revisions 65892 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65892 | oej | 2007-05-24 17:20:54 +0200 (Thu, 24 May 2007) | 2 lines Issue 8193 - NAT issues with gtalk/STUN. Patch by phsultan. Thanks! ........ * /, channels/chan_gtalk.c: Merged revisions 65857 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65857 | oej | 2007-05-24 17:05:10 +0200 (Thu, 24 May 2007) | 2 lines Issue 7686, fix by phsultan, NAT issues when calling from gtalk to SIP over nat. ........ 2007-05-24 15:10 +0000 [r65869] Joshua Colp * /, main/rtp.c: Merged revisions 65863 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65863 | file | 2007-05-24 11:08:17 -0400 (Thu, 24 May 2007) | 2 lines I like it when the RTP stack compiles myself... ........ 2007-05-24 15:04 +0000 [r65855] Russell Bryant * /, apps/app_festival.c: Merged revisions 65853 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65853 | russell | 2007-05-24 10:04:14 -0500 (Thu, 24 May 2007) | 4 lines Ensure that frames are fully initialized. This will probably fix getting weird timestamp log messages in logs when using the Festival app. (issue #9781, patch by me) ........ 2007-05-24 14:52 +0000 [r65844] Olle Johansson * /, channels/chan_gtalk.c: Merged revisions 65841 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65841 | oej | 2007-05-24 16:48:55 +0200 (Thu, 24 May 2007) | 2 lines Issue #8536 - Caller ID not set in CDR for jingle ........ 2007-05-24 14:50 +0000 [r65843] Russell Bryant * /, main/rtp.c: Merged revisions 65842 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65842 | russell | 2007-05-24 09:49:05 -0500 (Thu, 24 May 2007) | 5 lines Fix the calculation of the RTT for RTCP. The previous code would result in oscillating and incorrect data. Additionally, the RTT would sometimes report negative values due to incorrect calculations. (issue #9601, patch from davetroy) ........ 2007-05-24 14:43 +0000 [r65840] Joshua Colp * /, channels/chan_sip.c: Merged revisions 65839 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r65839 | file | 2007-05-24 10:42:12 -0400 (Thu, 24 May 2007) | 10 lines Merged revisions 65837 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r65837 | file | 2007-05-24 10:40:38 -0400 (Thu, 24 May 2007) | 2 lines Allow RFC2833 to be negotiated when an INVITE comes in without SDP and is not matched to a user or peer. (issue #9546 reported by mcrawford) ........ ................ 2007-05-24 14:41 +0000 [r65838] Olle Johansson * /, channels/chan_sip.c, res/res_jabber.c: Issue #8409 and accidentally a fix to chan_sip that wasn't supposed to be there but is still ok... Sorry. Lack of Tea, really. 2007-05-24 11:38 +0000 [r65814] Kevin P. Fleming * channels/chan_sip.c: Yes Virginia, there is a reason why we have stringfields in the sip_pvt structure... 2007-05-24 09:51 +0000 [r65769] Christian Richter * channels/chan_misdn.c, /: Merged revisions 65768 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r65768 | crichter | 2007-05-24 11:37:32 +0200 (Do, 24 Mai 2007) | 9 lines Merged revisions 65767 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r65767 | crichter | 2007-05-24 11:19:58 +0200 (Do, 24 Mai 2007) | 1 line we should only activate the generator in chan_misdn, when asterisk hask not yet taken the call (WAITING4DIGS state). Alerting audio will be generated fomr asterisk for example. ........ ................ 2007-05-24 03:28 +0000 [r65749] Russell Bryant * channels/chan_sip.c: - Remove debug variable that was only used in one place - convert string handling to the ast_str API - Convert strdup() to ast_strdup() and check the result - Minor formatting changes 2007-05-24 03:27 +0000 [r65748] Tilghman Lesher * cdr/cdr_adaptive_odbc.c: Oops, should have released this when we were done with it. 2007-05-24 02:23 +0000 [r65731] Mark Spencer * channels/chan_sip.c: Add SendURL support 2007-05-23 21:01 +0000 [r65678-65688] Kevin P. Fleming * /, channels/chan_iax2.c: Merged revisions 65685 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65685 | kpfleming | 2007-05-23 16:59:19 -0400 (Wed, 23 May 2007) | 2 lines start the delayed PBX when receive voice or video full frames as well, and comment this delayed-PBX activity ........ * /, channels/chan_sip.c: Merged revisions 65683 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r65683 | kpfleming | 2007-05-23 16:51:56 -0400 (Wed, 23 May 2007) | 10 lines Merged revisions 65682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r65682 | kpfleming | 2007-05-23 16:46:22 -0400 (Wed, 23 May 2007) | 2 lines ensure that variables are set on a newly created channel before we start a PBX on it ........ ................ * /, channels/chan_iax2.c: Merged revisions 65679-65680 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65679 | kpfleming | 2007-05-23 16:30:24 -0400 (Wed, 23 May 2007) | 2 lines don't start a PBX on a new incoming IAX2 channel until we have some sort of response to our ACCEPT (ACK or anything else) ........ r65680 | kpfleming | 2007-05-23 16:35:50 -0400 (Wed, 23 May 2007) | 2 lines clear the 'delay PBX' flag when we are ready to start the PBX ........ * /, channels/chan_iax2.c: Merged revisions 65677 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r65677 | kpfleming | 2007-05-23 16:07:59 -0400 (Wed, 23 May 2007) | 10 lines Merged revisions 65676 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r65676 | kpfleming | 2007-05-23 16:06:13 -0400 (Wed, 23 May 2007) | 2 lines if we are going to set variables on a newly created channel, it should be done *before* we start the PBX on it ........ ................ 2007-05-23 17:17 +0000 [r65659] Russell Bryant * apps/app_voicemail.c: Don't check for MWI event subscribers before creating the MWI event in voicemail. MWI events get cached, so go ahead and always generate them so the cache gets populated. 2007-05-23 15:37 +0000 [r65640] Matthew Fredrickson * channels/chan_zap.c: Make sure we get the cause code in the REL 2007-05-23 13:10 +0000 [r65591] Russell Bryant * channels/chan_zap.c, /: Merged revisions 65589 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r65589 | russell | 2007-05-23 08:07:13 -0500 (Wed, 23 May 2007) | 11 lines Merged revisions 65588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r65588 | russell | 2007-05-23 08:06:17 -0500 (Wed, 23 May 2007) | 3 lines Revert revision 62417 as someone reported problems with it to Mark. This was related to issue #9588. ........ ................ 2007-05-23 13:07 +0000 [r65590] Joshua Colp * res/res_musiconhold.c: Fix compiling of res_musiconhold under dev mode. 2007-05-23 02:55 +0000 [r65573] Russell Bryant * main/devicestate.c: Fix a couple minor spelling mistakes. 2007-05-22 20:26 +0000 [r65542] Kevin P. Fleming * /, build_tools/make_version: Merged revisions 65541 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65541 | kpfleming | 2007-05-22 16:25:41 -0400 (Tue, 22 May 2007) | 2 lines when building a version string for a developer branch, include the base branch in the version string ........ 2007-05-22 18:52 +0000 [r65502-65505] Russell Bryant * main/channel.c, configs/musiconhold.conf.sample, include/asterisk/channel.h, res/res_musiconhold.c, CHANGES: Add a new feature for Music on Hold. If you set the "digit" option for a class in musiconhold.conf, a caller on hold may press this digit to switch to listening to that music class. This involved adding a new callback for generators, which allow generators to get notified of DTMF from the channel they are running on. Then, a callback was implemented for the music on hold generators. (patch from bbryant) * channels/chan_zap.c, /, apps/app_voicemail.c: Merged revisions 65501 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65501 | russell | 2007-05-22 13:40:38 -0500 (Tue, 22 May 2007) | 3 lines List res_smdi as a dependency for app_voicemail and chan_zap (Thanks to mnicholson for pointing it out) ........ 2007-05-22 15:25 +0000 [r65455] BJ Weschke * /, apps/app_followme.c: Merged revisions 65408 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65408 | bweschke | 2007-05-22 10:02:56 -0400 (Tue, 22 May 2007) | 3 lines Fix a problem with flag recognition. ........ 2007-05-22 15:08 +0000 [r65451-65454] Joshua Colp * channels/chan_agent.c: Use ast_strlen_zero where possible. (issue #9774 reported by eliel) * main/cdr.c: Make my compiler happy! Yay! 2007-05-22 12:58 +0000 [r65376] Joshua Colp * res/res_features.c: Don't overwrite a pointer to the first channel... that is bad. (issue #9770 reported by tfbu) 2007-05-22 12:52 +0000 [r65375] Russell Bryant * apps/app_queue.c: Fix a couple of spots in the handling of device states that could lead to a double free. (issue #9772, reported by Mike Anikienko, fix by me) 2007-05-22 08:21 +0000 [r65343] Christian Richter * channels/chan_misdn.c, /: Merged revisions 65342 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r65342 | crichter | 2007-05-22 10:12:20 +0200 (Di, 22 Mai 2007) | 9 lines Merged revisions 65328 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r65328 | crichter | 2007-05-22 09:46:39 +0200 (Di, 22 Mai 2007) | 1 line we stop the tones only when we're in the pre-call phase, otherwise e.g. when in CONNECTED state we should not stop tones when we receive an Information Message ........ ................ 2007-05-22 02:41 +0000 [r65313] Tilghman Lesher * cdr/cdr_adaptive_odbc.c: Fix for 64-bit platform 2007-05-21 06:56 +0000 [r65298] Russell Bryant * apps/app_queue.c: I know we have talked about rewriting app_queue for Asterisk 1.6, but once I saw this, I couldn't help myself from changing it. Previously, for *every* device state change, app_queue would spawn a thread to handle it. Now, the device state callback just puts the state change in a queue and it gets handled by a single state change processing thread. 2007-05-21 02:05 +0000 [r65283] Tilghman Lesher * cdr/cdr_adaptive_odbc.c: Comment a few more things, and remove an unnecessary db connection check 2007-05-20 18:01 +0000 [r65233-65253] Joshua Colp * /, res/res_agi.c: Merged revisions 65250 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65250 | file | 2007-05-20 13:59:58 -0400 (Sun, 20 May 2007) | 2 lines res_agi needs to export two symbols (ast_agi_register and ast_agi_unregister) for usage by others. (issue #9755 reported by mnicholson) ........ * res/res_crypto.c, res/res_musiconhold.c: Music on hold and crypto no longer need their symbols globally exported. They register the function pointers upon loading with their respective stubs. * main/adsistub.c, main/cryptostub.c: Clean up adsistub file a bit (just spacing) and change over the crypto sub to use this build_stub macro strategy. * main/Makefile, main/adsistub.c, res/res_adsi.c: Add the adsistub file to the Asterisk makefile, fix a stub definition, and no longer make the symbols from res_adsi global since they don't need to be. 2007-05-18 22:35 +0000 [r65202-65203] Steve Murphy * main/cdr.c, /: Merged revisions 65201 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65201 | murf | 2007-05-18 16:26:51 -0600 (Fri, 18 May 2007) | 1 line Ugh. The svnmerge didn't catch the shift from cdr.c to main/cdr.c, and neither did I. This is the remainder of the 9717 patch, the fix for the run-away FAIL status for a call ........ * apps/app_dial.c, /, include/asterisk/cdr.h: Merged revisions 65200 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r65200 | murf | 2007-05-18 16:06:27 -0600 (Fri, 18 May 2007) | 9 lines Merged revisions 65172 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r65172 | murf | 2007-05-18 14:56:20 -0600 (Fri, 18 May 2007) | 1 line This update will fix the situation that occurs as described by 9717, where when several targets are specified for a dial, if any one them reports FAIL, the whole call gets FAIL, even though others were ringing OK. I rearranged the priorities, so that a new disposition, NULL, is at the lowest level, and the disposition get init'd to NULL. Then, next up is FAIL, and next up is BUSY, then NOANSWER, then ANSWERED. All the related set routines will only do so if the disposition value to be set to is greater than what's already there. This gives the intended effect. So, if all the targets are busy, you'd get BUSY for the call disposition. If all get BUSY, but one, and that one rings is not answered, you get NOANSWER. If by some freak of nature, the NULL value doesn't get overridden, then the disp2str routine will report NOANSWER as before. ........ ................ 2007-05-18 20:21 +0000 [r65169] Tilghman Lesher * cdr/cdr_adaptive_odbc.c (added), configs/cdr_adaptive_odbc.conf.sample (added): Merge cdr_adaptive_odbc from developer branch 2007-05-18 18:18 +0000 [r65077-65124] Olle Johansson * /, channels/chan_sip.c: Related to issue #9235 btw. Merged revisions 65123 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r65123 | oej | 2007-05-18 20:16:09 +0200 (Fri, 18 May 2007) | 10 lines Merged revisions 65122 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r65122 | oej | 2007-05-18 20:10:46 +0200 (Fri, 18 May 2007) | 2 lines Not getting an ACK to a 200 OK in the initial invite is critical to the call. ........ ................ * /, channels/chan_sip.c: Merged revisions 65076 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r65076 | oej | 2007-05-18 17:18:13 +0200 (Fri, 18 May 2007) | 13 lines Merged revisions 65075 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r65075 | oej | 2007-05-18 17:12:09 +0200 (Fri, 18 May 2007) | 5 lines Issue 9235 - part of the problem, maybe not all. Please retry with this patch (and no other patch) if you have problems with hanging SIP channels. Thank you. A special Thank You to WeBRainstorm that gave me access to his system. ........ ................ 2007-05-18 12:43 +0000 [r65006-65040] Christian Richter * /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: Merged revisions 65039 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r65039 | crichter | 2007-05-18 14:40:46 +0200 (Fr, 18 Mai 2007) | 9 lines Merged revisions 65007 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r65007 | crichter | 2007-05-18 13:23:11 +0200 (Fr, 18 Mai 2007) | 1 line fixed a warning regarding Keypad encoding. encode the IE sending_complete at the right position. ........ ................ * channels/chan_misdn.c, /: Merged revisions 64904 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r64904 | crichter | 2007-05-18 10:58:51 +0200 (Fr, 18 Mai 2007) | 9 lines Merged revisions 64902 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r64902 | crichter | 2007-05-18 10:24:08 +0200 (Fr, 18 Mai 2007) | 1 line we *need* to send a PROCEEDING when sending_complete is set, even if need_more_infos is requested. ........ ................ 2007-05-18 10:41 +0000 [r64973-64975] Olle Johansson * /, channels/chan_sip.c: Merged revisions 64974 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64974 | oej | 2007-05-18 12:37:44 +0200 (Fri, 18 May 2007) | 2 lines Issue 9487 - stop media flows at hangup of call ........ * channels/chan_sip.c: Makeup, darling. 2007-05-18 10:03 +0000 [r64951-64963] Christian Richter * channels/chan_misdn.c, /: Merged revisions 64515 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r64515 | crichter | 2007-05-16 10:44:51 +0200 (Mi, 16 Mai 2007) | 9 lines Merged revisions 64513 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r64513 | crichter | 2007-05-16 10:23:42 +0200 (Mi, 16 Mai 2007) | 1 line in the case immediate=yes, we directly jump into the dialplan, where people can use PlayTones to indicate a Dialtone, so we don't need to to that by ourself. also we should not do a dialtone_indicate for incoming calls on a TE port in overlapdialmode. ........ ................ * channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c: Merged revisions 63534 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r63534 | crichter | 2007-05-09 15:17:10 +0200 (Mi, 09 Mai 2007) | 17 lines Merged revisions 62945,63402,63519 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r62945 | crichter | 2007-05-03 17:39:21 +0200 (Do, 03 Mai 2007) | 1 line when we're in state WAITING4DIGS, we use the asterisk tone-generator which prods us, so we can't just return -1 in misdn_write in this case. Added a MISDN_KEYPAD channel variable, and fixed the sending of keypad. this enables us to modify the call forward parameters in the switch. ........ r63402 | crichter | 2007-05-08 17:07:37 +0200 (Di, 08 Mai 2007) | 1 line added application misdn_check_l2l1 which tries to pull up the L1/L2 on all ports that have the layers down in a group. It waits then for a timeout. This helps for scenarios where multiple PMP BRIs are grouped together, or where a provider has a faulty PTP Implementation, that looses the L2 after a while. ........ r63519 | crichter | 2007-05-09 13:26:16 +0200 (Mi, 09 Mai 2007) | 1 line release_chan frees ch, so we should never touch ch after release_chan, this may cause segfaults. ........ ................ * channels/misdn/isdn_lib_intern.h, channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c: Merged revisions 62912 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r62912 | crichter | 2007-05-03 16:36:32 +0200 (Do, 03 Mai 2007) | 17 lines Merged revisions 61357,61770,62885 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61357 | crichter | 2007-04-11 14:05:57 +0200 (Mi, 11 Apr 2007) | 1 line some fixes for PMP Hold/Retrieve, it should work now, when briding=no ........ r61770 | crichter | 2007-04-24 15:50:05 +0200 (Di, 24 Apr 2007) | 1 line added lock for sending messages to avoid double sending. shuffled some empty_chans after the cb_event calls, this avoids that a release_complete from a quite different call releases a fresh created setup by accident. ........ r62885 | crichter | 2007-05-03 15:59:00 +0200 (Do, 03 Mai 2007) | 1 line fixed the problem that misdn_write did not return -1 when called with 0 samples in a frame this resultet in a deadlock in some circumstances, when the call ended because of a busy extension. added encoding of keypad. ........ ................ * channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h, channels/misdn/isdn_lib.c, channels/misdn_config.c: Merged revisions 59774 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r59774 | crichter | 2007-04-03 09:20:27 +0200 (Di, 03 Apr 2007) | 17 lines Merged revisions 59623-59624,59639 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59623 | crichter | 2007-04-02 09:12:24 +0200 (Mo, 02 Apr 2007) | 1 line we can now make 30 channels on a PRI (before we forgot chan 31..) ........ r59624 | crichter | 2007-04-02 09:25:54 +0200 (Mo, 02 Apr 2007) | 1 line don't be verbose if no need ........ r59639 | crichter | 2007-04-02 14:08:12 +0200 (Mo, 02 Apr 2007) | 1 line added option which allows us to accept incoming SETUP Messages without automatically sending Proceeding or Setup Acknowledge, this is useful with some broken switches and if you want to Release incoming calls without previously having acknowledged them. The new option is noautorespond_on_setup=yes|no default is no, so we don't break the existing behaviour ........ ................ * channels/chan_misdn.c, /: Merged revisions 59254 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r59254 | crichter | 2007-03-27 17:00:10 +0200 (Di, 27 Mär 2007) | 9 lines Merged revisions 59252 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59252 | crichter | 2007-03-27 15:56:15 +0200 (Di, 27 Mär 2007) | 1 line fixed #9355 ........ ................ * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h, channels/misdn/isdn_lib.c, channels/misdn_config.c: Merged revisions 59064 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r59064 | crichter | 2007-03-20 14:16:06 +0100 (Di, 20 Mär 2007) | 21 lines Merged revisions 58849-58850,59062-59063 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r58849 | crichter | 2007-03-13 12:58:16 +0100 (Di, 13 Mär 2007) | 1 line added method standard_dec for dialing out on groups, to avoid conflicts, which caused issues with some ISDN providers ........ r58850 | crichter | 2007-03-13 13:58:32 +0100 (Di, 13 Mär 2007) | 1 line fixed the crypt_keys stuff ........ r59062 | crichter | 2007-03-20 10:18:06 +0100 (Di, 20 Mär 2007) | 1 line avoid sending a disconnect when we already received one. ........ r59063 | crichter | 2007-03-20 10:23:22 +0100 (Di, 20 Mär 2007) | 1 line modified a loglevel ........ ................ * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c: Merged revisions 58825-58826 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r58825 | crichter | 2007-03-12 13:43:24 +0100 (Mo, 12 Mär 2007) | 1 line added UU transceiving and corect handling for rdnis ................ r58826 | crichter | 2007-03-12 14:08:06 +0100 (Mo, 12 Mär 2007) | 21 lines Merged revisions 57034,57523,57753,58558 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r57034 | crichter | 2007-02-28 17:09:27 +0100 (Mi, 28 Feb 2007) | 1 line fixed bugs.digium.com bugs: #9157 and bugs.beronet.com bugs: #302, #303, #304 ........ r57523 | crichter | 2007-03-02 19:32:51 +0100 (Fr, 02 Mar 2007) | 1 line fixed typo ........ r57753 | crichter | 2007-03-04 11:39:50 +0100 (So, 04 Mar 2007) | 1 line fixed another place where the out_cause was hardcoded to 16 ........ r58558 | crichter | 2007-03-09 15:43:58 +0100 (Fr, 09 Mar 2007) | 1 line we can free channel 31 as well, since we can occupy it ........ ................ 2007-05-18 09:10 +0000 [r64903-64921] Olle Johansson * include/asterisk/adsi.h, main/adsistub.c (added), res/res_adsi.c, apps/app_voicemail.c: Issue #5930 - Remove dependencies on res_adsi.so - clwade A big THANK YOU to clwade for this patch. Minor modifications by me. * channels/chan_sip.c: Another fix for the support for recordings controlled by INFO-packets We still lack a setting to enable/disable this per peer 2007-05-18 02:55 +0000 [r64869-64870] Russell Bryant * CHANGES: Add ENUMQUERY and ENUMRESULT to the CHANGES file. * /, apps/app_queue.c: Merged revisions 64868 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64868 | russell | 2007-05-17 21:48:51 -0500 (Thu, 17 May 2007) | 5 lines Fix a small bug I noticed while working on something else. app_queue did not unregister its device state monitoring callback in unload_module(). So, this would make Asterisk crash on the first device state change after you unload the module. ........ 2007-05-17 21:20 +0000 [r64821] Tilghman Lesher * /, include/asterisk/linkedlists.h: Merged revisions 64820 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r64820 | tilghman | 2007-05-17 16:19:34 -0500 (Thu, 17 May 2007) | 10 lines Merged revisions 64819 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r64819 | tilghman | 2007-05-17 16:14:36 -0500 (Thu, 17 May 2007) | 2 lines How is it that we never caught that this is returning the opposite of our documentation, until now? ........ ................ 2007-05-17 17:12 +0000 [r64786] Russell Bryant * main/manager.c, configs/manager.conf.sample: Add an option that lets you only allow one connection at a time for each manager user. (issue #8664, reported and original patch by ssokol, patch updated by bkruse, and further updated by me) 2007-05-17 16:54 +0000 [r64762] Jason Parker * /, apps/app_voicemail.c: Merged revisions 64761 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r64761 | qwell | 2007-05-17 11:53:27 -0500 (Thu, 17 May 2007) | 12 lines Merged revisions 64758 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r64758 | qwell | 2007-05-17 11:52:38 -0500 (Thu, 17 May 2007) | 4 lines If we have a negative current message, we shouldn't go back even further... Issue 9727. ........ ................ 2007-05-17 16:53 +0000 [r64757-64760] Russell Bryant * /, contrib/scripts/astxs (removed): Merged revisions 64759 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64759 | russell | 2007-05-17 11:52:53 -0500 (Thu, 17 May 2007) | 3 lines Remove script that is no longer functional since the build system was redone. (issue #9340, reported by junky) ........ * apps/app_dial.c, /: Merged revisions 64756 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64756 | russell | 2007-05-17 11:47:29 -0500 (Thu, 17 May 2007) | 3 lines Increase the size of a buffer to support longer dial strings for channels. (issue #9291, reported and fix suggested by meni) ........ 2007-05-17 16:11 +0000 [r64721-64755] Joshua Colp * /, channels/chan_sip.c: Merged revisions 64754 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64754 | file | 2007-05-17 12:10:12 -0400 (Thu, 17 May 2007) | 2 lines Even more direct RTP setup fixes! Don't allow a codec that isn't supported to creep into the SDP of either side. (issue #9446 reported by marcelbarbulescu) ........ * /, apps/app_voicemail.c: Merged revisions 64720 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64720 | file | 2007-05-17 09:48:44 -0400 (Thu, 17 May 2007) | 2 lines Fix authuser support. (issue #9740 reported by xmarksthespot) ........ 2007-05-17 06:14 +0000 [r64657-64687] Russell Bryant * README, /: Merged revisions 64686 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64686 | russell | 2007-05-17 01:13:53 -0500 (Thu, 17 May 2007) | 3 lines Update the main README to reflect the new build process for 1.4 and above. (issue #9725, patch by eliel) ........ * main/app.c: Ignore this ... playing with jira (AST-1) 2007-05-16 11:01 +0000 [r64494-64611] Olle Johansson * /: Blocking patch * /, channels/chan_sip.c: Below patches with some re-structuring for trunk --- Merged revisions 64602 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64602 | oej | 2007-05-16 12:38:18 +0200 (Wed, 16 May 2007) | 2 lines Issue #9681 - Handle www-auth on BYE ........ * /, channels/chan_sip.c: Merged revisions 64578 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64578 | oej | 2007-05-16 12:05:47 +0200 (Wed, 16 May 2007) | 2 lines Final part of issue #9483 - fixing transfer() of sip calls in the dial plan (twilson) ........ * /: Blocking patch that was already committed to trunk * /, channels/chan_sip.c: Merged revisions 64543 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r64543 | oej | 2007-05-16 11:12:34 +0200 (Wed, 16 May 2007) | 10 lines Merged revisions 64535 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r64535 | oej | 2007-05-16 11:08:22 +0200 (Wed, 16 May 2007) | 2 lines Support SIP uri's starting with SIP: and sip: (reported by Tony Mountfield on the mailing list. Thanks!) ........ ................ * /, channels/chan_sip.c: Merged revisions 64516 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r64516 | oej | 2007-05-16 10:46:18 +0200 (Wed, 16 May 2007) | 17 lines Merged following patch with a lot of changes for 1.4 ------ Merged revisions 64514 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r64514 | oej | 2007-05-16 10:25:56 +0200 (Wed, 16 May 2007) | 6 lines Issue #9726 - rlister - Better logging for ACL denials While at it, also added better logging and handling of peers that are not supposed to register. My patch, stole the issue report from Russell. My apologies, Russell :-) ........ ................ * channels/chan_sip.c: Issue #9304 - Update help text to match functionality. Patch by kshumard with changes by oej * channels/chan_sip.c, configs/sip.conf.sample: Issue #6789 - Marquis - Add option to support regexten removal when host becomes unreachable * main/event.c: This file really needs more documentation... When we implement new API's - please include a small general overview in Doxygen * main/dial.c: Small doxygen updates 2007-05-15 23:05 +0000 [r64469-64480] Russell Bryant * funcs/func_enum.c, include/asterisk/enum.h, main/enum.c: Add two new dialplan functions: ENUMQUERY and ENUMRESULT. These functions allow you to initiate an ENUM query using ENUMQUERY, and then access the details of all of the results using ENUMRESULT. Previously, if you wanted to access multiple results, Asterisk would have to do a new DNS lookup every time. (patch by bbryant) * pbx/pbx_dundi.c: Make sure that DUNDIRESULT is given an ID. 2007-05-15 20:45 +0000 [r64455] Matthew Fredrickson * channels/chan_zap.c, configs/zapata.conf.sample: XXX-XXX-XXX appears to be the standard ANSI pointcode format 2007-05-15 19:57 +0000 [r64427] Russell Bryant * /, res/res_features.c: Merged revisions 64426 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64426 | russell | 2007-05-15 14:52:18 -0500 (Tue, 15 May 2007) | 3 lines Properly fix a problem that occurs when you set PARKINGEXTEN to an exten where a call is already parked. (issue #9723, patch by me) ........ 2007-05-14 23:43 +0000 [r64399] Kevin P. Fleming * /: this does not belong here 2007-05-14 22:25 +0000 [r64384] Matthew Fredrickson * channels/chan_zap.c: Only print the SS7 UP once. Not every time we get the test messages on the line. 2007-05-14 21:51 +0000 [r64355] Jason Parker * main/Makefile: With libmmime.a as a .PHONY target, asterisk gets rebuilt every time, but without proper ASTCFLAGS. This caused a problem with the buildinfo.o file not being able to find asterisk/build.h This was affecting DESTDIR, but I *think* that if asterisk had never been installed before, it would've failed also. 2007-05-14 21:17 +0000 [r64354] Russell Bryant * /, res/res_features.c: Merged revisions 64353 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64353 | russell | 2007-05-14 16:16:39 -0500 (Mon, 14 May 2007) | 4 lines When someone requests a specific parking space using the PARKINGEXTEN variable, ensure that no other caller is already there. (issue #9723, reported by mdu113, patch by me) ........ 2007-05-14 19:35 +0000 [r64323-64325] Olle Johansson * /, channels/chan_sip.c: Merged revisions 64324 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64324 | oej | 2007-05-14 21:26:50 +0200 (Mon, 14 May 2007) | 2 lines Change -2 to XMIT_ERROR to clarify a bit more ........ * /: Blocking patch already committed to trunk 2007-05-14 19:21 +0000 [r64322] Russell Bryant * /, channels/chan_alsa.c: Merged revisions 64306 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64306 | russell | 2007-05-14 14:13:00 -0500 (Mon, 14 May 2007) | 3 lines Properly handle AST_CONTROL_PROGRESS by just ignoring it. An unknown indication will trigger an error and cause sounds to stop, which in this case, is ringing. ........ 2007-05-14 18:49 +0000 [r64274-64279] Joshua Colp * /, codecs/codec_speex.c: Merged revisions 64278 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64278 | file | 2007-05-14 14:48:33 -0400 (Mon, 14 May 2007) | 2 lines Properly set datalen field when doing PLC in codec_speex. (issue #9722 reported by mihai) ........ * /, main/devicestate.c: Merged revisions 64276 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r64276 | file | 2007-05-14 14:36:34 -0400 (Mon, 14 May 2007) | 10 lines Merged revisions 64275 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r64275 | file | 2007-05-14 14:34:06 -0400 (Mon, 14 May 2007) | 2 lines Only perform stripping of - strings from the channel name for Zap channels. Anywhere else we might remove a legitimate part of a device name. (issue #9668 reported by stevedavies) ........ ................ * channels/chan_sip.c: If no port is specified in the outboundproxy setting then use the standard SIP port. (issue #9665 reported by tootai) 2007-05-14 18:14 +0000 [r64243-64273] Jason Parker * configs/queues.conf.sample: oops - silly typo there * configs/queues.conf.sample, apps/app_queue.c: Don't allow rounding seconds to weird values that may cause "unexpected" results. Issue 9514. * apps/app_queue.c: Add 'c' option to app_queue which allows for continuing in the dialplan if the callee hangs up. Issue 9284, patch by lyl, modified a little bit by me (I felt 'continue' was better than 'keepalive') 2007-05-14 17:25 +0000 [r64242] Joshua Colp * main/channel.c, /: Merged revisions 64240 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64240 | file | 2007-05-14 13:23:51 -0400 (Mon, 14 May 2007) | 2 lines Fix scenario where if a phone that simply called Echo() put itself on hold it could never get off hold. ........ 2007-05-14 16:08 +0000 [r64225-64226] Russell Bryant * configure: Regenerate configure script after last change to acinclude.m4 * acinclude.m4: Remove an extra space from the macro that checks for C defines. (issue #9715, tzafrir) 2007-05-14 14:13 +0000 [r64208] Steve Murphy * main/cdr.c, main/pbx.c, channels/chan_local.c, /: Merged revisions 64193 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64193 | murf | 2007-05-14 07:58:42 -0600 (Mon, 14 May 2007) | 1 line As per 9570, worrisome CDR warnings have been removed, that are either not helpful, or not relevant. ........ 2007-05-14 10:40 +0000 [r64142-64158] Olle Johansson * main/channel.c, /: Merged revisions 64157 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64157 | oej | 2007-05-14 12:39:12 +0200 (Mon, 14 May 2007) | 2 lines Add hangupcause when we lack codecs for transcoding ........ * channels/chan_sip.c: Improve handling network errors on transmission to hosts that don't reply or are unreachable With this code, the call will fail as soon as we get a network error. This may happen on first xmit or a later one, so the retransmit code handles this too. 2007-05-12 22:28 +0000 [r64087-64115] Joshua Colp * /, channels/chan_sip.c: Merged revisions 64114 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64114 | file | 2007-05-12 18:27:04 -0400 (Sat, 12 May 2007) | 2 lines This concludes my final adventure with bitmasks and the onhold flag. Would anyone care for some peanuts? ........ * /, channels/chan_sip.c: Merged revisions 64086 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64086 | file | 2007-05-12 17:10:45 -0400 (Sat, 12 May 2007) | 2 lines Tweak hold flags some more. They can be of three states when active: active, inactive, one direction. ........ 2007-05-12 19:38 +0000 [r64072] Tilghman Lesher * funcs/func_enum.c: Issue 9716 - doc/enum.txt no longer exists in trunk 2007-05-12 16:33 +0000 [r64045] Joshua Colp * /, channels/chan_sip.c: Merged revisions 64044 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64044 | file | 2007-05-12 12:32:15 -0400 (Sat, 12 May 2007) | 2 lines Ensure the onhold flag is set no matter what when being put on hold. ........ 2007-05-11 22:52 +0000 [r63967-64030] Jason Parker * channels/chan_skinny.c, configs/skinny.conf.sample: Add/fix support for Redial, Speeddial, and Messages buttons. Combined effort by DEA and mvanbaak. * main/asterisk.c: oops.. Fix the logic of the last commit. * Makefile, main/asterisk.c: Better fallback method for autosystemname. Issue 9713, patch by Juggie with minor mods by me. * main/manager.c, /: Merged revisions 63982 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r63982 | qwell | 2007-05-11 15:16:17 -0500 (Fri, 11 May 2007) | 7 lines Hide manager password from "manager show user foo". I realize that there are other ways to get this, but we really don't need to just show it in plain text so easily. Issue 9273, patch by junky ........ * Makefile, main/asterisk.c: Add autosystemname setting to asterisk.conf When enabled, it will set the systemname to be the hostname of the system Issue 9713, patch by Juggie - slightly modified by me, to "failover" to localhost 2007-05-11 18:31 +0000 [r63946] Russell Bryant * doc/qos.tex: Fix some syntax errors. 2007-05-11 16:37 +0000 [r63906] Tilghman Lesher * Makefile, /, contrib/scripts/safe_asterisk: Merged revisions 63905 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r63905 | tilghman | 2007-05-11 11:35:51 -0500 (Fri, 11 May 2007) | 10 lines Merged revisions 63903 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r63903 | tilghman | 2007-05-11 11:31:03 -0500 (Fri, 11 May 2007) | 2 lines Issue 9121 - fixups for safe_asterisk script ........ ................ 2007-05-11 16:21 +0000 [r63901-63902] Russell Bryant * main/manager.c, /: Merged revisions 63886 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r63886 | russell | 2007-05-11 11:05:43 -0500 (Fri, 11 May 2007) | 6 lines When MD5 authentication is not possible because there is no challenge present, either because the Challenge action was never issued, or some other reason, give a proper error message and return an error instead of claiming that the user wasn't found. (reported by jsmith on IRC) ........ * res/res_agi.c: Add gender support for AGI SAY NUMBER. (issue #9537, patch by chappell) 2007-05-11 15:48 +0000 [r63873] Joshua Colp * /, res/res_features.c: Merged revisions 63872 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r63872 | file | 2007-05-11 11:43:14 -0400 (Fri, 11 May 2007) | 2 lines Make the PARKINGEXTEN feature of parking actually work. (issue #9708 reported by mdu113) ........ 2007-05-10 23:16 +0000 [r63832] Jason Parker * /, channels/chan_iax2.c: Merged revisions 63830 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r63830 | qwell | 2007-05-10 18:15:37 -0500 (Thu, 10 May 2007) | 12 lines Merged revisions 63828 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r63828 | qwell | 2007-05-10 18:14:55 -0500 (Thu, 10 May 2007) | 4 lines Fix an issue with trying to kill a thread before it gets created. Issue 9709, patch by nic_bellamy. ........ ................ 2007-05-10 22:25 +0000 [r63805] Russell Bryant * main/manager.c, /: Merged revisions 63804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r63804 | russell | 2007-05-10 17:23:42 -0500 (Thu, 10 May 2007) | 4 lines Strip terminal escape sequences from CLI command output that is going to be sent out over the manager interface. (issue #9659, reported by pari, fixed by me) ........ 2007-05-10 21:25 +0000 [r63786] Doug Bailey * main/callerid.c: Added check for negative offset in cid spill to prevent infinite loops 2007-05-10 20:51 +0000 [r63730-63751] Olle Johansson * /, channels/chan_sip.c: Merged revisions 63749 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r63749 | oej | 2007-05-10 22:46:41 +0200 (Thu, 10 May 2007) | 12 lines Merged revisions 63748 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r63748 | oej | 2007-05-10 22:38:54 +0200 (Thu, 10 May 2007) | 4 lines Do not allocate SIP pvt's for PEERs we can not reach. This was seen as a lot of dialogs being created then immediately destroyed at reload/restart of the SIP channel. ........ ................ * apps/app_minivm.c: Fixing reload. Thanks to Mats Karlsson! 2007-05-09 19:24 +0000 [r63699] Joshua Colp * main/channel.c, /: Merged revisions 63698 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r63698 | file | 2007-05-09 15:22:39 -0400 (Wed, 09 May 2007) | 2 lines Use the DTMF frame on the channel when returning a DTMF frame from AST_FRAME_NULL or AST_FRAME_VOICE. ........ 2007-05-09 19:21 +0000 [r63697] Russell Bryant * main/channel.c, /: Merged revisions 63612 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r63612 | russell | 2007-05-09 11:55:27 -0500 (Wed, 09 May 2007) | 5 lines Modify ast_senddigit_begin() to use the same assumptions used elsewhere in the code in that if a channel does not have a send_digit_begin() callback, it only cares about DTMF END events. (pointed out by Michael Neuhauser on the asterisk-dev list) ........ 2007-05-09 17:35 +0000 [r63655] Matthew Fredrickson * channels/chan_zap.c: Merged revisions 63654 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r63654 | mattf | 2007-05-09 12:25:21 -0500 (Wed, 09 May 2007) | 10 lines Merged revisions 63653 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r63653 | mattf | 2007-05-09 12:20:20 -0500 (Wed, 09 May 2007) | 2 lines Make sure we only create a DSP if it's requested on SUB_REAL ........ ................ 2007-05-09 16:56 +0000 [r63613] Joshua Colp * /, channels/chan_sip.c: Merged revisions 63611 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r63611 | file | 2007-05-09 12:54:56 -0400 (Wed, 09 May 2007) | 10 lines Merged revisions 63610 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r63610 | file | 2007-05-09 12:51:03 -0400 (Wed, 09 May 2007) | 2 lines Properly handle hints that point to multiple devices in chan_sip. Why chan_sip is even doing this I have no idea but I would rather not go into a rant. (issue #9536 reported by rlister) ........ ................ 2007-05-09 16:44 +0000 [r63609] Russell Bryant * main/channel.c, /: Merged revisions 63608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r63608 | russell | 2007-05-09 11:43:50 -0500 (Wed, 09 May 2007) | 5 lines Only call ast_senddigit_begin() in ast_senddigit() if the channel has a send_digit_begin() callback. Checking the END_DTMF_ONLY flag was the wrong thing to do, because that flag indicates that a *bridged* channel only wants DTMF END events coming from this channel. ........ 2007-05-09 14:52 +0000 [r63567] Tilghman Lesher * /, apps/app_directory.c: Merged revisions 63566 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r63566 | tilghman | 2007-05-09 09:50:33 -0500 (Wed, 09 May 2007) | 10 lines Merged revisions 63565 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r63565 | tilghman | 2007-05-09 09:48:06 -0500 (Wed, 09 May 2007) | 2 lines Replicate fix from 51158 (app_voicemail) to app_directory (Issue 9224) ........ ................ 2007-05-09 13:24 +0000 [r63536] Russell Bryant * Makefile, /: Merged revisions 63535 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r63535 | russell | 2007-05-09 08:24:03 -0500 (Wed, 09 May 2007) | 6 lines I have seen multiple people post questions trying to figure out what the message "The configure script must be executed before running 'make'" means. So, add another like that says to specifically run ./configure. If this isn't obvious enough, then they should be using something like AsteriskNOW and not installing from source. ........ 2007-05-09 13:07 +0000 [r63533] Olle Johansson * /, channels/chan_sip.c: Merged revisions 63532 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r63532 | oej | 2007-05-09 15:04:14 +0200 (Wed, 09 May 2007) | 2 lines Don't retransmit 200 OK's on ignore status. (Reported on asterisk-users) ........ 2007-05-08 22:40 +0000 [r63479] Tilghman Lesher * /, apps/app_macro.c: Merged revisions 63478 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r63478 | tilghman | 2007-05-08 17:38:02 -0500 (Tue, 08 May 2007) | 10 lines Merged revisions 63477 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r63477 | tilghman | 2007-05-08 17:19:15 -0500 (Tue, 08 May 2007) | 2 lines Issue 9602 - segfault in app_macro ........ ................ 2007-05-08 16:54 +0000 [r63404-63449] Russell Bryant * /, res/res_features.c: Merged revisions 63448 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r63448 | russell | 2007-05-08 11:53:09 -0500 (Tue, 08 May 2007) | 4 lines I mixed up the use of the find_feature() function, so I renamed it find_dynamic_feature, and changed the code to use the correct lock when using it. ........ * channels/chan_sip.c, res/res_features.c, include/asterisk/features.h: I noted this on the dev list but got no response, so I just did it myself. Lock the call features when being used in chan_sip. * /, res/res_features.c: Merged revisions 63445 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r63445 | russell | 2007-05-08 11:30:43 -0500 (Tue, 08 May 2007) | 2 lines Use a read/write lock when accessing the built-in features. ........ * contrib/scripts/realtime_pgsql.sql (added), /, contrib/realtime_pgsql.sql (removed): Merged revisions 63403 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r63403 | russell | 2007-05-08 10:10:37 -0500 (Tue, 08 May 2007) | 3 lines Move realtime_pgsql.sql to contrib/scripts to be with the rest of the sql examples. (issue #9676, suretec) ........ 2007-05-08 06:26 +0000 [r63361] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 63360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r63360 | tilghman | 2007-05-08 01:22:37 -0500 (Tue, 08 May 2007) | 10 lines Merged revisions 63359 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r63359 | tilghman | 2007-05-08 01:20:16 -0500 (Tue, 08 May 2007) | 2 lines Issue 9527 - upon entering a folder, no message is selected (curmsg == -1), so deleting causes memory corruption (beyond bounds) ........ ................ 2007-05-07 22:32 +0000 [r63319-63330] Russell Bryant * /, contrib/realtime_pgsql.sql (added), configs/res_pgsql.conf.sample (added): Merged revisions 63329 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r63329 | russell | 2007-05-07 17:28:50 -0500 (Mon, 07 May 2007) | 3 lines Add a sample configuration file and example tables for use with res_config_pgsql. (issue #9676, suretec) ........ * apps/app_meetme.c: Make a minor tweak to admin_exec() - don't lock the conference list until it is actually necessary. * apps/app_meetme.c, CHANGES: Add a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin, except it lets you operate on a channel by name instead of conference member number. It is very useful in combination with the 'X' option to ChanSpy. (issue #9671, patch by mnicholson, with some small modifications by me) 2007-05-07 21:47 +0000 [r63284-63287] Joshua Colp * main/channel.c, include/asterisk/app.h, /, main/app.c: Merged revisions 63286 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r63286 | file | 2007-05-07 17:45:01 -0400 (Mon, 07 May 2007) | 10 lines Merged revisions 63285 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r63285 | file | 2007-05-07 17:39:52 -0400 (Mon, 07 May 2007) | 2 lines Properly handle what happens during a masquerade in relation to group counting. (issue #9657 reported by ramonpeek) ........ ................ 2007-05-07 20:07 +0000 [r63228-63255] Olle Johansson * /, main/config.c: Merged revisions 63254 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r63254 | oej | 2007-05-07 22:05:15 +0200 (Mon, 07 May 2007) | 2 lines Don't remove configuration from memory just because one section failed. ........ * include/asterisk/module.h, main/loader.c: Constifications * channels/chan_jingle.c, res/res_jabber.c: Adding external referenses for doxygen See http://www.asterisk.org/doxygen/trunk/extref.html * channels/chan_misdn.c: Adding external reference * channels/chan_misdn.c: Doxyfication... There's a shortage of comments in this file... 2007-05-06 20:09 +0000 [r63182] Joshua Colp * channels/chan_iax2.c: Lock iax2 pvt structure when passing off to the AMI function, and make sure it exists. (issue #9674 reported by arabe) 2007-05-06 13:11 +0000 [r63168] Olle Johansson * /, main/file.c: Merged revisions 63152 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r63152 | oej | 2007-05-06 14:28:38 +0200 (Sun, 06 May 2007) | 2 lines Stop the video stream when you stop playback of all streams for a call ........ 2007-05-05 08:05 +0000 [r63136] Olle Johansson * channels/chan_sip.c: - Adding some missing spaces - Correcting error messages - Disabling code that doesn't do anything - Making sure we always respond to this request, happily 2007-05-04 20:11 +0000 [r63105] Pari Nannapaneni * /, configs/manager.conf.sample: Merged revisions 63047 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r63047 | pari | 2007-05-04 11:45:29 -0500 (Fri, 04 May 2007) | 1 line explanation for httptimeout in manager.conf ........ 2007-05-04 20:06 +0000 [r63104] Jason Parker * /, res/res_jabber.c: Merged revisions 63099 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r63099 | qwell | 2007-05-04 15:03:49 -0500 (Fri, 04 May 2007) | 4 lines Fix a crash when checking version attribute in an incoming XML caps element. Issue 9667, patch by phsultan. ........ 2007-05-04 19:48 +0000 [r63089] Russell Bryant * main/manager.c: Convert spaces to tabs for indentation. 2007-05-04 18:47 +0000 [r63046-63076] Steve Murphy * res/res_features.c: According to my testing, it's better if the ast_find_call_feature func ran this way instead, as far as the snom record button is concerned * doc/CODING-GUIDELINES, channels/chan_sip.c, res/res_features.c, include/asterisk/features.h: a small upgrade to the coding standard, and an update to the code that triggered the upgrade. * channels/chan_sip.c, res/res_features.c, UPGRADE.txt, include/asterisk/features.h: Added a small bit of code to support the SNOM 360's Record button. Made the find_feature func in res_features.c public, so I could use it to find the automon dial sequence as configured by the user. When the INFO packet has a Record: header with on/off, the sequence is sent as consecutive DTMF frames on the phone's channel, triggering the automon functionality. The user has to configure the automon in features.conf, and set up his dialplan accordingly. 2007-05-04 13:56 +0000 [r63030-63032] Olle Johansson * channels/chan_sip.c, channels/chan_iax2.c: Add the new ChannelUpdate event to inform manager clients about the PVT ID and some other channel driver data that is needed to follow the call through the PBX. * main/manager.c: Add "CoreStatus" - from the moremanager branch. This can be extended with more information, ideas and patches are welcome, as usual :-) * include/asterisk.h, main/manager.c, include/asterisk/manager.h, include/asterisk/options.h: - Add manager command CoreSettings - Add missing option to options.h - Add missing variables to asterisk.h - Move manager version to manager.h include file 2007-05-03 16:45 +0000 [r62990] Joshua Colp * /, channels/chan_sip.c: Merged revisions 62989 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r62989 | file | 2007-05-03 13:44:00 -0300 (Thu, 03 May 2007) | 10 lines Merged revisions 62987 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r62987 | file | 2007-05-03 13:42:19 -0300 (Thu, 03 May 2007) | 2 lines When a peer is seeded or built tell the devicestate core to update it's status. This is easier then having chan_sip load before pbx_config. (issue #9658 reported by dlynes) ........ ................ 2007-05-03 16:43 +0000 [r62988] Kevin P. Fleming * /, main/loader.c: Merged revisions 62986 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r62986 | kpfleming | 2007-05-03 11:38:56 -0500 (Thu, 03 May 2007) | 2 lines improve loader a bit, by avoiding trying to initialize embedded modules twice and avoiding trying to load modules from disk when they have been loaded already during the 'preload' pass (reported by blitzrage on IRC, patch by me) ........ 2007-05-03 15:23 +0000 [r62943] Russell Bryant * main/channel.c, /: Merged revisions 62942 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r62942 | russell | 2007-05-03 10:23:13 -0500 (Thu, 03 May 2007) | 17 lines Fix YADB (Yet Another DTMF Bug) ((C) Russell Bryant, 2007, TM, Patent Pending). This set of changes came from a debugging session I had with Dwayne Hubbard. When he called into his home FXO, ran the Echo application, and pressed a digit, the digit would be echoed back and would never end. This is fixed, along with a couple other little improvements. * When chan_zap is in the middle of playing a digit to a channel, it feeds back null frames, not voice frames. So, I have modified ast_read to check the timing on emulated DTMF when it receives null frames, in addition to where it was doing this on voice frames. * Make a tweak to setting the duration on emulated DTMF digits. If there was no duration specified, it set it to be the minimum, instead of the default. * Instead of timing the emulated digits off of the number of samples in audio frames that pass through, just use time values. Now there is no code in this section that assumes 8kHz audio. ........ 2007-05-03 14:44 +0000 [r62911-62914] Steve Murphy * /: blocking 62913 (1.4) from trunk, as it's already done here * /, pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/ael/ael-test/ref.ael-test20 (added), pbx/ael/ael.tab.h, pbx/ael/ael-test/ael-test20/extensions.ael (added), pbx/ael/ael-test/ael-test20 (added): Merged revisions 62883 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r62883 | murf | 2007-05-03 07:54:56 -0600 (Thu, 03 May 2007) | 1 line These mods fix bug 9623, where an '@' in the eswitch contents causes a syntax error. I also updated the regressions. ........ 2007-05-03 00:25 +0000 [r62824-62843] Kevin P. Fleming * res/res_config_odbc.c, /: Merged revisions 62842 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r62842 | kpfleming | 2007-05-02 20:23:37 -0400 (Wed, 02 May 2007) | 10 lines Merged revisions 62841 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r62841 | kpfleming | 2007-05-02 20:23:00 -0400 (Wed, 02 May 2007) | 2 lines doh... initializing the pointer variable will work just a bit better ........ ................ * main/minimime: ignore the archive we build in this directory * res/res_config_pgsql.c, res/res_config_odbc.c, /: Merged revisions 62797,62807 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r62797 | kpfleming | 2007-05-02 19:57:23 -0400 (Wed, 02 May 2007) | 7 lines improve static Realtime config loading from PostgreSQL: don't request sorting on fields that are pointless to sort on use ast_build_string() instead of snprintf() don't request the list of fieldnames that resulted from the query when we both knew what they were before we ran the query _AND_ we aren't going to do anything with them anyway (patch by me, inspired by blitzrage's bug report about res_config_odbc) ................ r62807 | kpfleming | 2007-05-02 20:02:57 -0400 (Wed, 02 May 2007) | 15 lines Merged revisions 62796 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r62796 | kpfleming | 2007-05-02 19:53:46 -0400 (Wed, 02 May 2007) | 7 lines increase reliability and efficiency of static Realtime config loading via ODBC: don't request fields we aren't going to use don't request sorting on fields that are pointless to sort on explicitly request the fields we want, because we can't expect the database to always return them in the order they were created (reported by blitzrage in person (!), patch by me) ........ ................ 2007-05-02 23:50 +0000 [r62791-62795] Russell Bryant * CHANGES: Fix some bad grammar. * apps/app_meetme.c, CHANGES: When a conference is created, the UNIQUEID of the channel that caused it to be created will now be stored. Then, every channel that joins the conference will have the MEETMEUNIQUEID channel variable set with this ID. This can be used to relate callers that come and go from long standing conferences. (issue #7295, patch by softins) * CHANGES: Note Hungarian language support in CHANGES * main/say.c, configs/say.conf.sample: Add Hungarian language support to say.c and say.conf. (issue #7077, patch by adomjan) * main/channel.c, /: Merged revisions 62789 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r62789 | russell | 2007-05-02 17:59:09 -0500 (Wed, 02 May 2007) | 20 lines Merge changes from team/russell/inband_dtmf ... Fix some issues related to generating inband DTMF. There are two changes here: 1) The list of DTMF tones in the senddigit_begin() function explicitly specified 100ms of the tone followed by 100ms of silence. This really broke things with the way that Asterisk now wants complete control over when the digit begins and ends. So, regardless of what Asterisk really wanted to do, this was going to play out the tone at the length it wanted to. This caused various problems like DTMF translation to inband to be extremely unreliable. The list of tones has been changed so that the correct DTMF tone is played indefinitely until Asterisk tells it to stop. 2) ast_write() had to be modified to let a DTMF_END frame get processed even when a generator is present. This is how the tone will finally get stopped. (issues #8944, #9250, #9348, maybe others. Thanks to mdu113 from #8944 for the testing and feedback!) ........ 2007-05-02 20:57 +0000 [r62741] Steve Murphy * main/cdr.c, main/pbx.c, /: Merged revisions 62738 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r62738 | murf | 2007-05-02 14:46:07 -0600 (Wed, 02 May 2007) | 9 lines Merged revisions 62737 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r62737 | murf | 2007-05-02 14:10:32 -0600 (Wed, 02 May 2007) | 1 line Some tweaks to satisfy CDR bug 8796, where being in 'h' extension louses up the dst field ........ ................ 2007-05-02 17:49 +0000 [r62693] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 62692 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r62692 | tilghman | 2007-05-02 12:43:48 -0500 (Wed, 02 May 2007) | 12 lines Merged revisions 62691 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r62691 | tilghman | 2007-05-02 12:38:16 -0500 (Wed, 02 May 2007) | 4 lines Issue 9638 - if a text frame is sent with no terminating NULL through a bridged IAX connection, the remote end will receive garbage characters tacked onto the end. ........ ................ 2007-05-02 17:24 +0000 [r62690] Steve Murphy * main/channel.c, main/pbx.c, channels/chan_zap.c, /, cdr/cdr_radius.c: Merged revisions 62689 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r62689 | murf | 2007-05-02 11:10:50 -0600 (Wed, 02 May 2007) | 1 line a)In chan_zap, set the clid, src fields in channel_alloc call. b)in the channel_alloc func, set the cid_num and name fields from the arglist[blush]. c) don't update the channel app & app data fields if you are in the 'h' extension. d)the load_module func in cdr_radius needs to return DECLINE, SUCCESS. ........ 2007-05-02 15:46 +0000 [r62671-62673] Russell Bryant * channels/chan_local.c, CHANGES: Update the device state functionality of chan_local such that it will return NOT_INUSE or INUSE when Local channels are in use as opposed to just UNKNOWN. It will still return INVALID if the extension doesn't exist at all. (issue #8048, patch from tim_ringenbach) * CHANGES: Add the new options for attended transfer to the CHANGES file. * doc/ip-tos.tex (removed), doc/qos.tex (added): For some reason when I merged 802.1p support, the new documentation file was not properly added. Thanks to IgorG for pointing it out! :) 2007-05-02 12:12 +0000 [r62609-62656] Olle Johansson * channels/chan_sip.c: Add a small message that we're doing something. On my systems, there's a long dead period with a non-responsive CLI after I issue "load chan_sip.so" * channels/chan_sip.c: More username body parts to fix... If working, this needs to be backported to 1.2, 1.4. But first, some serious SIP testing :-) * channels/chan_sip.c: Handle sip:username;parameter=12345@example.com;parameter=1234 URI's properly * /, channels/chan_sip.c: Merged revisions 62624 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r62624 | oej | 2007-05-02 08:15:43 +0200 (Wed, 02 May 2007) | 2 lines Don't unlock a channel that we already know does not exist (propably isue 8228) ........ * CREDITS: Updating CREDITS 2007-05-01 22:24 +0000 [r62549-62593] Russell Bryant * res/res_features.c, configs/features.conf.sample: In addition to making it so attended transfers don't fail unnecessarily, add some new options to control what happens when you hangup on an attended transfer before the target extension answers the transferred channel. You can now have it send the transferee back to the transferer. (issue #8413, patch from sergee with very minor modifications by me) * /, res/res_features.c: Merged revisions 62548 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r62548 | russell | 2007-05-01 16:57:10 -0500 (Tue, 01 May 2007) | 12 lines Merged revisions 62547 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r62547 | russell | 2007-05-01 16:55:19 -0500 (Tue, 01 May 2007) | 4 lines Remove an unnecessary check that makes it so if you hang up after doing an attended transfer before the target extension answers the channel, the transfer is not successful. (issue #9338, patch by svanlund) ........ ................ 2007-05-01 21:41 +0000 [r62546] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 62545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r62545 | tilghman | 2007-05-01 16:34:43 -0500 (Tue, 01 May 2007) | 2 lines Bug 9590 - Memory leaks around find_user() (found by rayjay, different fixes by me) ........ 2007-05-01 16:27 +0000 [r62415-62498] Russell Bryant * /, configs/indications.conf.sample: Merged revisions 62497 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r62497 | russell | 2007-05-01 11:26:48 -0500 (Tue, 01 May 2007) | 11 lines Merged revisions 62496 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r62496 | russell | 2007-05-01 11:26:23 -0500 (Tue, 01 May 2007) | 3 lines Add indications.conf information for the Philippines. (issue #9525, reported and patched by loloski) ........ ................ * CHANGES: Add a note to CHANGES about the new support for 802.1p. Thanks IgorG! * CHANGES, apps/app_queue.c, doc/queuelog.tex: This patch adds additional information to the EXITWITHKEY and EXITWITHTIMEOUT entries in the queue log. (issue #7561, reported and originally patched by fkasumovic, patch slightly modified and updated to trunk by me) * include/asterisk/acl.h, main/udptl.c, channels/chan_sip.c, include/asterisk/rtp.h, main/acl.c, include/asterisk/netsock.h, channels/iax2-provision.c, channels/chan_iax2.c, main/rtp.c, main/netsock.c, configs/h323.conf.sample, configs/iax.conf.sample, configs/mgcp.conf.sample, configs/iaxprov.conf.sample, channels/chan_h323.c, pbx/pbx_dundi.c, include/asterisk/udptl.h, configs/sip.conf.sample, doc/asterisk.tex, channels/chan_mgcp.c: Add support for setting the CoS for VLAN traffic (802.1p) in Linux. The file doc/qos.tex has been updated to document the new functionality. (issue #9540, patch submitted by IgorG) * channels/chan_zap.c, /: Merged revisions 62419 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r62419 | russell | 2007-04-30 10:58:28 -0500 (Mon, 30 Apr 2007) | 12 lines Merged revisions 62417 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r62417 | russell | 2007-04-30 10:57:26 -0500 (Mon, 30 Apr 2007) | 4 lines This patch fixes an issue where depending on the cause code, when the network sends a PRI disconnect, the call may not be properly hung up. (issue #9588, reported and patched by softins) ........ ................ * channels/chan_sip.c: Don't crash when invalid arguments are provided to the CHANNEL() function for a SIP channel. (issue #9619, reported by jtodd, original patch by Corydon76, committed patch slightly modified by me) * include/asterisk/http.h, /, main/http.c: Merged revisions 62414 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r62414 | russell | 2007-04-30 10:25:31 -0500 (Mon, 30 Apr 2007) | 4 lines When serving dynamic content, include a Cache-Control header to instruct the browsers to not store the resulting content. (issue #9621, reported by Pari, patch by me) ........ 2007-04-30 14:56 +0000 [r62372] Jason Parker * configs/iax.conf.sample, /: Merged revisions 62371 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r62371 | qwell | 2007-04-30 09:52:31 -0500 (Mon, 30 Apr 2007) | 2 lines Remove unused (and potentially confusing) jitterbuffer options from sample config. ........ 2007-04-30 14:37 +0000 [r62370] Joshua Colp * /, main/asterisk.c: Merged revisions 62369 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r62369 | file | 2007-04-30 11:36:11 -0300 (Mon, 30 Apr 2007) | 10 lines Merged revisions 62368 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r62368 | file | 2007-04-30 11:34:07 -0300 (Mon, 30 Apr 2007) | 2 lines Update copyright notice. It's now the year 2007! ........ ................ 2007-04-29 05:51 +0000 [r62219-62332] Russell Bryant * channels/chan_zap.c, /: Merged revisions 62331 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r62331 | russell | 2007-04-29 00:50:37 -0500 (Sun, 29 Apr 2007) | 3 lines Fix a bug that made the "language" setting in zapata.conf not functional. (issue #9626, reported and fixed by sergee) ........ * CHANGES: note MeetMe change in CHANGES * apps/app_meetme.c: Enable the functionality of the 'o' option to "optimize talker" by default. * channels/iax2.h: Reformat some of iax2.h and convert comments to doxygen format * include/asterisk.h, channels/chan_zap.c, channels/chan_sip.c, main/Makefile, res/res_eventtest.c (added), configs/voicemail.conf.sample, UPGRADE.txt, CHANGES, channels/chan_iax2.c, main/dial.c, include/asterisk/event.h (added), include/asterisk/event_defs.h (added), main/event.c (added), configs/sip.conf.sample, main/asterisk.c, channels/chan_mgcp.c, apps/app_voicemail.c: Merge changes from team/russell/events This set of changes introduces a new generic event API for use within Asterisk. I am still working on a way for events to be shared between servers, but this part is ready and can already be used inside of Asterisk. This set of changes introduces the first use of the API, as well. I have restructured the way that MWI (message waiting indication) is handled. It is now event based instead of polling based. For example, if there are a bunch of SIP phones subscribed to mailboxes, then chan_sip will not have to constantly poll the mailboxes for changes. app_voicemail will generate events when changes occur. See UPGRADE.txt and CHANGES for some more information on the effects of these changes from the user perspective. For developer information, see the text in include/asterisk/event.h. As always, additional feedback is welcome on the asterisk-dev mailing list. * doc/ast_appdocs.tex, doc/dundi.tex: Update the DUNDi section of the documentation with example usage of DUNDIQUERY and DUNDIRESULT. Also, update the automatically generated application docs. * pbx/pbx_dundi.c, CHANGES: Merge changes from team/russell/dundi_results This introduces two new dialplan functions: DUNDIQUERY and DUNDIRESULT. DUNDIQUERY lets you intitiate a DUNDi query from the dialplan. Then, DUNDIRESULT will let you find out how many results there are, and access each one without having to the query again. * include/asterisk/lock.h: Remove a message that goes to LOG_ERROR that's not really an error. * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Add a min-announce-frequency option to queues.conf which allows you to control the minimum amount of time between queue announcements for use when the caller's queue position changes frequently. (issue #9604, patch by Matthew Roth) * /, channels/chan_agent.c: Merged revisions 62218 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r62218 | russell | 2007-04-27 16:10:51 -0500 (Fri, 27 Apr 2007) | 11 lines Fix a weird problem where when a caller talking to someone sitting behind an agent channel sent a digit, the digit would be played to the agent for forever. This is because chan_agent always returned -1 from its send_digit_begin and _end callbacks. This non-zero return value indicates to the Asterisk core that it would like an inband DTMF generator put on the channel. However, this is the wrong thing to do. It should *always* return 0, instead. When the digit begin and end functions are called on the proxied channel, the underlying channel will indicate whether inband DTMF is needed or not, and the generator will be put on that one, and not the Agent channel. (issue #9615, #9616, reported by jiddings and BigJimmy, and fixed by me) ........ 2007-04-27 16:18 +0000 [r62175] Jason Parker * /, codecs/codec_zap.c: Merged revisions 62174 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r62174 | qwell | 2007-04-27 11:17:46 -0500 (Fri, 27 Apr 2007) | 11 lines Merged revisions 62173 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r62173 | qwell | 2007-04-27 11:16:16 -0500 (Fri, 27 Apr 2007) | 3 lines This transcoder message needn't be a NOTICE. I've seen it cause confusion more than a few times. ........ ................ 2007-04-27 16:15 +0000 [r62172] Russell Bryant * main/pbx.c, /: Merged revisions 62171 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r62171 | russell | 2007-04-27 11:14:11 -0500 (Fri, 27 Apr 2007) | 6 lines If no variables were passed into pbx_substitute_variables_helper_full(), then don't even bother creating a temporary bogus channel, since that is only for allowing certain functions to operate on the variables as if they were on a channel. Most importantly, this fixes a crash. (issue #9613, reported by callguy, fixed by me) ........ 2007-04-27 14:40 +0000 [r62096-62141] Olle Johansson * channels/chan_sip.c: Issue #9545 Autocomplete for "sip unregister" cli command. (eliel) Thanks! * /, channels/chan_sip.c: Merged revisions 62137 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r62137 | oej | 2007-04-27 16:04:07 +0200 (Fri, 27 Apr 2007) | 12 lines Merged revisions 62126 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r62126 | oej | 2007-04-27 15:57:45 +0200 (Fri, 27 Apr 2007) | 4 lines Issue #7351 - SIP Cancel fails due to the wrong contact uri. Reported by PPYY, failed to fix by OEJ final fix by wojtekka - THANKS!!!! THis was a hard one to catch. ........ ................ * /: Blocking patch to 1.4 that was alredy in trunk 2007-04-26 16:35 +0000 [r62039] Joshua Colp * /, channels/chan_iax2.c: Merged revisions 62038 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r62038 | file | 2007-04-26 12:33:52 -0400 (Thu, 26 Apr 2007) | 10 lines Merged revisions 62037 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r62037 | file | 2007-04-26 12:30:57 -0400 (Thu, 26 Apr 2007) | 2 lines Revert previous fix for when the IAX2 channel goes funky (that's the technical term). This is causing legit calls to be prematurely hung up. (issue #9600 reported by justdave) ........ ................ 2007-04-26 03:24 +0000 [r62006] Russell Bryant * main/channel.c, /: Merged revisions 62005 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r62005 | file | 2007-04-25 22:19:51 -0500 (Wed, 25 Apr 2007) | 2 lines Missed an ast_app_group_discard during merge. Thanks blitzrage! ........ 2007-04-26 01:50 +0000 [r61960-61962] Joshua Colp * /, res/res_monitor.c: Merged revisions 61961 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61961 | file | 2007-04-25 21:48:55 -0400 (Wed, 25 Apr 2007) | 2 lines Don't always say that the channel is being paused if it is actually being unpaused in the Manager ack message. (reported by jsmith in #asterisk-bugs) ........ * /, main/config.c: Merged revisions 61959 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61959 | file | 2007-04-25 21:27:18 -0400 (Wed, 25 Apr 2007) | 10 lines Merged revisions 61958 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61958 | file | 2007-04-25 21:25:03 -0400 (Wed, 25 Apr 2007) | 2 lines Don't count failed include attempts against the configuration include level. (issue #9593 reported by mostyn) ........ ................ 2007-04-25 22:34 +0000 [r61915] Kevin P. Fleming * channels/chan_zap.c, /: Merged revisions 61914 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61914 | kpfleming | 2007-04-25 17:29:53 -0500 (Wed, 25 Apr 2007) | 10 lines Merged revisions 61913 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61913 | kpfleming | 2007-04-25 17:24:59 -0500 (Wed, 25 Apr 2007) | 2 lines handle a very bizarre race condition with channels being redirected before a simple switch can be started on them (issue #9286) ........ ................ 2007-04-25 22:01 +0000 [r61864-61876] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 61870 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61870 | russell | 2007-04-25 16:59:07 -0500 (Wed, 25 Apr 2007) | 10 lines Merged revisions 61866 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61866 | russell | 2007-04-25 16:55:23 -0500 (Wed, 25 Apr 2007) | 2 lines If the callerid= option is specified, but empty, clear any previous data. ........ ................ * /, channels/chan_iax2.c: Merged revisions 61863 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61863 | russell | 2007-04-25 16:13:15 -0500 (Wed, 25 Apr 2007) | 10 lines Merged revisions 61862 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61862 | russell | 2007-04-25 16:06:22 -0500 (Wed, 25 Apr 2007) | 2 lines Ensure that callerid settings are reset on a reload. ........ ................ 2007-04-25 19:27 +0000 [r61806] Joshua Colp * main/channel.c, include/asterisk/app.h, funcs/func_groupcount.c, /, main/app.c, main/cli.c: Merged revisions 61805 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61805 | file | 2007-04-25 15:21:54 -0400 (Wed, 25 Apr 2007) | 10 lines Merged revisions 61804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61804 | file | 2007-04-25 14:52:50 -0400 (Wed, 25 Apr 2007) | 2 lines Merge rewritten group counting support. No more storing data on the variable list of the channels. That was bad, mmmk? (issue #7497 reported by sabbathbh) ........ ................ 2007-04-25 16:23 +0000 [r61788-61800] Russell Bryant * channels/chan_zap.c, /: Merged revisions 61799 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61799 | russell | 2007-04-25 11:22:07 -0500 (Wed, 25 Apr 2007) | 11 lines Merged revisions 61798 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61798 | russell | 2007-04-25 11:20:38 -0500 (Wed, 25 Apr 2007) | 3 lines Fix a typo where cid_num got copied instead of cid_ani. (issue #9587, reported and patched by xrg) ........ ................ * main/manager.c, /: Merged revisions 61787 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61787 | russell | 2007-04-24 16:34:53 -0500 (Tue, 24 Apr 2007) | 12 lines Merged revisions 61786 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61786 | russell | 2007-04-24 16:33:59 -0500 (Tue, 24 Apr 2007) | 4 lines Don't crash if a manager connection provides a username that exists in manager.conf but does not have a password, and also requests MD5 authentication. (ASA-2007-012) ........ ................ 2007-04-24 19:08 +0000 [r61784] Dwayne M. Hubbard * channels/chan_zap.c, /: removed #if 0 block from chan_zap restart_monitor() 2007-04-24 19:03 +0000 [r61775-61782] Russell Bryant * main/channel.c, /, include/asterisk/channel.h: Merged revisions 61781 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61781 | russell | 2007-04-24 14:00:06 -0500 (Tue, 24 Apr 2007) | 6 lines Improve DTMF handling in ast_read() even more in response to a discussion on the asterisk-dev mailing list. I changed the enforced minimum length of a digit from 100ms to 80ms. Furthermore, I made it now enforce a gap of 45ms in between digits. These values are not configurable in a configuration file right now, but they can be easily changed near the top of main/channel.c. ........ * main/dial.c, /: Merged revisions 61774 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61774 | russell | 2007-04-24 11:16:41 -0500 (Tue, 24 Apr 2007) | 5 lines Add a few more state changes in handle_frame_ownerless() so that the SLA code will get notified of these changes even when an owner channel is not provided. This isn't from a specific bug report, it's just something I noticed while poking around. ........ 2007-04-24 16:10 +0000 [r61773] Joshua Colp * /, channels/chan_sip.c: Merged revisions 61772 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61772 | file | 2007-04-24 12:07:02 -0400 (Tue, 24 Apr 2007) | 10 lines Merged revisions 61771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61771 | file | 2007-04-24 12:05:06 -0400 (Tue, 24 Apr 2007) | 2 lines Allow RFC2833 to be sent in the response SDP when an INVITE comes in without SDP. (issue #9546 reported by mcrawford) ........ ................ 2007-04-23 18:49 +0000 [r61760-61767] Russell Bryant * main/manager.c: When building a JSON encoded string in the GetConfigJSON manager action, escape the '\' and '"' characters. (issue #9475, reported by pari, patch by me) * main/pbx.c, /: Merged revisions 61765 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61765 | russell | 2007-04-23 13:17:00 -0500 (Mon, 23 Apr 2007) | 5 lines Some dialplan functions, such as CUT(), expect to operate on variables on a channel. So, this little hack lets them work in places where a channel doesn't exist, such as within DUNDi configuration. (issue #9465, reported and patched by Corydon76, testing by blitzrage) ........ * main/channel.c, /: Merged revisions 61763 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61763 | russell | 2007-04-23 12:57:32 -0500 (Mon, 23 Apr 2007) | 4 lines Ensure that digits passing through Asterisk have a reasonable minimum length. It is currently 100 ms. If someone thinks this should be different, feel free to speak up. (related to issues #8944, #9250, and #9348) ........ * CHANGES: Add OSP support for IAX2 to the changes file. Also, slightly reorganize some of the content. 2007-04-20 21:37 +0000 [r61706-61708] Jason Parker * /, main/rtp.c: Merged revisions 61707 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61707 | qwell | 2007-04-20 16:35:27 -0500 (Fri, 20 Apr 2007) | 8 lines Avoid invalid seqno cycling detection. Per comment from Dave Troy: This adds back in some simple typecasting I had in an earlier version which I realize now may be breaking things. Issue #9554. ........ * /, main/loader.c: Merged revisions 61705 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61705 | qwell | 2007-04-20 16:15:29 -0500 (Fri, 20 Apr 2007) | 12 lines Merged revisions 61704 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61704 | qwell | 2007-04-20 16:14:27 -0500 (Fri, 20 Apr 2007) | 4 lines Fix an issue that I noticed while looking over issue 9571. The reload timestamp was getting set after reloading the built-in stuff, and before the modules. ........ ................ 2007-04-20 21:12 +0000 [r61698-61702] Russell Bryant * channels/iax2-parser.h, funcs/func_channel.c, channels/iax2.h, channels/chan_iax2.c, channels/iax2-parser.c: Merge changes from team/russell/iax2_osp This set of changes adds OSP support to chan_iax2. However, I have modified the patch a bit from what was submitted. You now use the CHANNEL() function to get and set the OSP token for IAX2. (issue #8531, reported by and original patch by homesick, patch updated by me) * /, main/rtp.c: Merged revisions 61697 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61697 | russell | 2007-04-20 15:42:02 -0500 (Fri, 20 Apr 2007) | 2 lines Remove a stray debug message introduced by a recent commit. ........ 2007-04-20 19:54 +0000 [r61695] Jason Parker * /, apps/app_queue.c: Merged revisions 61694 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61694 | qwell | 2007-04-20 14:51:49 -0500 (Fri, 20 Apr 2007) | 13 lines Merged revisions 61692 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61692 | qwell | 2007-04-20 14:49:54 -0500 (Fri, 20 Apr 2007) | 5 lines If the '* to hangup' option is not enabled, we don't need to disable * as a valid exit key. If it was enabled, this statement would've never been checked in the first place. Issue #9552 ........ ................ 2007-04-20 18:23 +0000 [r61691] Russell Bryant * main/manager.c, /, include/asterisk/config.h, main/config.c, apps/app_voicemail.c: Merged revisions 61690 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61690 | russell | 2007-04-20 13:19:18 -0500 (Fri, 20 Apr 2007) | 4 lines Fix the UpdateConfig manager action to properly treat "variables" and "objects" differently (a=b versus a=>b). (issue #9568, reported by pari, patch by me) ........ 2007-04-20 08:41 +0000 [r61689] Olle Johansson * /, channels/chan_sip.c: Use the last line in the SDP, even if it has no CRLF. Remember Jon Postel :-) This code exists in 1.2 and 1.4 but was removed from trunk for some unknown reason. 2007-04-19 04:37 +0000 [r61682-61684] Tilghman Lesher * main/manager.c, /: Merged revisions 61683 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61683 | tilghman | 2007-04-18 23:36:20 -0500 (Wed, 18 Apr 2007) | 2 lines Bug 9557 - simple reason why reading a function always returned NULL ........ * funcs/func_groupcount.c, /, funcs/func_timeout.c, funcs/func_cdr.c, funcs/func_callerid.c: Merged revisions 61681 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61681 | tilghman | 2007-04-18 21:45:05 -0500 (Wed, 18 Apr 2007) | 13 lines Merged revisions 61680 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61680 | tilghman | 2007-04-18 21:30:18 -0500 (Wed, 18 Apr 2007) | 5 lines Bug 9557 - Specifying the GetVar AMI action without a Channel parameter can cause Asterisk to crash. The reason this needs to be fixed in the functions instead of in AMI is because Channel can legitimately be NULL, such as when retrieving global variables. ........ ................ 2007-04-18 22:11 +0000 [r61679] Kevin P. Fleming * /, sounds/Makefile: Merged revisions 61678 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61678 | kpfleming | 2007-04-18 17:10:23 -0500 (Wed, 18 Apr 2007) | 2 lines allow external build systems to extract the required sound file versions ........ 2007-04-18 20:48 +0000 [r61671-61677] Olle Johansson * /, main/rtp.c: Merged revisions 61676 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61676 | oej | 2007-04-18 22:46:23 +0200 (Wed, 18 Apr 2007) | 2 lines Clean upp formatting, add some doxygen stuff while we're in cleaning mode... Thanks Kevin! ........ * /, main/rtp.c: Merged revisions 61674 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61674 | oej | 2007-04-18 22:28:53 +0200 (Wed, 18 Apr 2007) | 2 lines Issue #9554 - Improve RTCP (Dave Troy) ........ * apps/app_minivm.c (added), configs/extensions_minivm.conf.sample (added), configs/minivm.conf.sample (added): Mini-voicemail - an embryo for a new voicemail system based on building blocks instead of one large monolithic app. Supports multiple templates and is designed mostly for voicemail delivery over e-mail. There's a todo with a list of ideas in the source code if you want to contribute. Feedback is appreciated! 2007-04-16 15:40 +0000 [r61667] Olle Johansson * include/asterisk/rtp.h: Doxygen changes 2007-04-14 18:22 +0000 [r61661] Claude Patry * main/say.c: test my new trunk access ;) 2007-04-13 21:23 +0000 [r61660] Dwayne M. Hubbard * channels/chan_sip.c: added CLI 'sip unregister ' for issue 9326. thanks eliel 2007-04-13 21:22 +0000 [r61659] Steve Murphy * main/cdr.c, /: Merged revisions 61658 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61658 | murf | 2007-04-13 15:17:20 -0600 (Fri, 13 Apr 2007) | 1 line This is a fix to the way CDR merge handles the data that results from ForkCDR. ........ 2007-04-13 19:18 +0000 [r61649-61657] Joshua Colp * apps/app_dial.c, /: Merged revisions 61656 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61656 | file | 2007-04-13 15:17:08 -0400 (Fri, 13 Apr 2007) | 10 lines Merged revisions 61655 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61655 | file | 2007-04-13 15:15:12 -0400 (Fri, 13 Apr 2007) | 2 lines Add OUTBOUND_GROUP_ONCE variable to app_dial. This behaves the same as OUTBOUND_GROUP except it will get unset after use so it won't get accidentally inherited. (issue #BE-140) ........ ................ * /, apps/app_speech_utils.c: Merged revisions 61651 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61651 | file | 2007-04-13 14:08:02 -0400 (Fri, 13 Apr 2007) | 2 lines Do not bother looking for a result if none are present. ........ * /, channels/chan_sip.c: Merged revisions 61648 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61648 | file | 2007-04-13 13:19:53 -0400 (Fri, 13 Apr 2007) | 2 lines For those very verbose SIP implementations that attach tons of info to the Contact header... let's increase our variable sizes. (issue #9535 reported by jeffg) ........ 2007-04-13 17:15 +0000 [r61647] Russell Bryant * /, apps/app_voicemail.c: Merged revisions 61645 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61645 | russell | 2007-04-13 12:10:19 -0500 (Fri, 13 Apr 2007) | 3 lines Eliminate a compiler warning with ODBC_STORAGE enabled so that it will build under dev-mode. ........ 2007-04-13 17:11 +0000 [r61646] Steve Murphy * /, channels/chan_oss.c: Merged revisions 61644 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61644 | murf | 2007-04-13 11:01:02 -0600 (Fri, 13 Apr 2007) | 1 line A fix for chan_oss that resulted from the CDR changes; it helps to use the right info. ........ 2007-04-13 16:35 +0000 [r61618-61642] Joshua Colp * /, channels/chan_sip.c: Merged revisions 61641 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61641 | file | 2007-04-13 12:32:03 -0400 (Fri, 13 Apr 2007) | 2 lines Don't assume the callid of a dialog will be set, as in some circumstances it may not. (issue #9534 reported by tecnoxarxa) ........ * channels/chan_sip.c: Don't treat a host lookup as failed if sipregs is not in use when doing a realtime lookup. (issue #9255 reported by sergee) 2007-04-11 22:19 +0000 [r61575-61599] Dwayne M. Hubbard * doc/asterisk-conf.tex: clarified 'minmemfree' description in doc/asterisk-conf.tex * main/asterisk.c, doc/asterisk-conf.tex: fixed the '-e' command line option for minmemfree. updated doc/asterisk-conf.tex * main/pbx.c, include/asterisk/options.h, main/asterisk.c: changed #if HAVE_SYSINFO to #if defined(HAVE_SYSINFO) * main/pbx.c, include/asterisk/options.h, main/asterisk.c: added HAVE_SYSINFO preprocessor directives for portability and general happiness 2007-04-11 20:21 +0000 [r61557] Joshua Colp * configure, include/asterisk/autoconfig.h.in, configure.ac: Add a configure script check for sysinfo support. 2007-04-11 19:11 +0000 [r61539] Dwayne M. Hubbard * main/pbx.c, include/asterisk/options.h, main/asterisk.c: added option_minmemfree for use in asterisk.conf to specify the amount of minimum free memory prior to accepting calls. added CLI 'core show sysinfo' to display system information 2007-04-11 17:07 +0000 [r61522] Joshua Colp * main/logger.c: Output verbose messages to the normal logger as well. (issue #9476 reported by gdalgliesh) 2007-04-11 16:06 +0000 [r61478] Russell Bryant * /, channels/chan_sip.c: Merged revisions 61477 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61477 | russell | 2007-04-11 11:05:29 -0500 (Wed, 11 Apr 2007) | 13 lines Merged revisions 61476 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61476 | russell | 2007-04-11 11:01:25 -0500 (Wed, 11 Apr 2007) | 5 lines If someone sets the "useragent" option in sip.conf to be empty, then don't add the User-Agent header at all. It is an optional header, anyway. Also, the bug report says that some of Japan's SIP providers don't allow it for some weird reason. (issue #9488, reported by makoto, fixed by me) ........ ................ 2007-04-11 15:48 +0000 [r61460] Nadi Sarrar * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged revisions 61342,61372-61373,61443 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61342 | nadi | 2007-04-11 12:52:28 +0200 (Mi, 11 Apr 2007) | 2 lines AOCD's are now exported to asterisk channel variables. ........ r61372 | nadi | 2007-04-11 15:33:30 +0200 (Mi, 11 Apr 2007) | 2 lines Ignore facility messages in case we don't have a corresponding channel object. ........ r61373 | nadi | 2007-04-11 15:40:26 +0200 (Mi, 11 Apr 2007) | 2 lines Export AOCD variables on misdn_hangup. ........ r61443 | nadi | 2007-04-11 17:39:14 +0200 (Mi, 11 Apr 2007) | 2 lines Don't export AOCD variables on misdn_hangup anymore, this was mainly a fix for trunk.. ........ 2007-04-11 15:25 +0000 [r61379-61429] Russell Bryant * funcs/func_devstate.c: Add a minor loop optimization to the custom device state callback. Once the correct device is found, it should just break out of the loop ... * /, channels/chan_sip.c: Merged revisions 61427 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61427 | russell | 2007-04-11 10:09:39 -0500 (Wed, 11 Apr 2007) | 14 lines Merged revisions 61426 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61426 | russell | 2007-04-11 10:05:36 -0500 (Wed, 11 Apr 2007) | 6 lines Fix a bug with switching between host=dynamic and using specific hosts for peers. The code would only reset the peer's address when it is dynamic if it was a new peer structure. Now, it will also reset the address if it was already in the peer list, but before the reload, it was not dynamic. (issue #9515, reported by caio1982, fixed by me) ........ ................ * /, main/http.c: Merged revisions 61407 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61407 | russell | 2007-04-11 09:48:01 -0500 (Wed, 11 Apr 2007) | 4 lines Add "svgz" to the mimetypes table. (issue #9510, bkruse) In passing, constify the elements of the mimetypes table. ........ * /, channels/chan_sip.c: Merged revisions 61377 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61377 | russell | 2007-04-11 09:04:44 -0500 (Wed, 11 Apr 2007) | 13 lines Merged revisions 61376 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61376 | russell | 2007-04-11 09:02:54 -0500 (Wed, 11 Apr 2007) | 5 lines Remove the attempt at reporting configuration errors in sip.conf. This can cause a bunch of improper messages when using realtime. I give up. As oej tried to convince me when I put this in, there is just no easy way to do it. (inspired by a message on the -dev list) ........ ................ 2007-04-11 14:09 +0000 [r61378] Steve Murphy * apps/app_voicemail.c: via 8119, a patch to allow voicemail data to be stored in RealTime. 2007-04-11 14:01 +0000 [r61375] Joshua Colp * channels/chan_sip.c: Remove duplicate prototype declaration. (issue #9517 reported by junky) 2007-04-11 13:41 +0000 [r61374] Steve Murphy * include/asterisk/config.h, main/config.c: via 8118, a RealTime upgrade to make RT a complete storage abstraction. The store/destroy mechanisms needed these missing peices. 2007-04-10 23:55 +0000 [r61324] Tilghman Lesher * main/channel.c, main/manager.c, configs/manager.conf.sample, include/asterisk/manager.h: Issue 6082 - New DTMF event for manager 2007-04-10 22:02 +0000 [r61303] Doug Bailey * channels/chan_zap.c: Added zapata.conf parameter "cid_rxgain" to allow the user to adjust the gain bump used during CID acquisition. 2007-04-10 20:50 +0000 [r61222-61283] Russell Bryant * CHANGES: Note the bridge manager action and application in the CHANGES file. * res/res_features.c: Merge changes from team/russell/issue_5841: This patch adds a "Bridge" Manager action, as well as a "Bridge" dialplan application. The manager action will allow you to steal two active channels in the system and bridge them together. Then, the one that did not hang up will continue in the dialplan. Using the application will bridge the calling channel to an arbitrary channel in the system. Whichever channel does not hang up here will continue in the dialplan, as well. This patch has been touched by a bunch of people over the course of a couple years. Please forgive me if I have missed your name in the history of things. The most recent patch came from issue #5841, but there is also a reference to an earlier version of this patch from issue #4297. The people involved in writing and/or reviewing the code include at least: twisted, mflorrel, heath1444, davetroy, tim_ringenbach, moy, tmancill, serge-v, and me. There are also positive test reports from many people. * main/dial.c, include/asterisk/dial.h: Add an option to the dial API for playing music instead of ringing to the caller. I started this for use with SLA but ended up deciding not to use it. However, there is no reason not to put this part in, anyway. 2007-04-10 16:07 +0000 [r61221] Steve Murphy * channels/chan_jingle.c: updated ast_channel_alloc() call to include the 4 extra args everyone got. Not much info there, as the config file evidently does not allow amaflags, or accountcode settings; and the pvt's exten doesn't sound like what we need in the cdr, either. 2007-04-10 12:47 +0000 [r61184] Nadi Sarrar * /, channels/misdn_config.c: Merged revisions 61183 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61183 | nadi | 2007-04-10 14:43:40 +0200 (Di, 10 Apr 2007) | 10 lines Merged revisions 61170 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61170 | nadi | 2007-04-10 14:31:45 +0200 (Di, 10 Apr 2007) | 2 lines msns config parameter defaults to '*' ........ ................ 2007-04-10 05:41 +0000 [r61152] Steve Murphy * main/pbx.c, channels/chan_local.c, channels/chan_vpb.cc, channels/chan_zap.c, /, channels/chan_sip.c, res/res_features.c, channels/chan_agent.c, include/asterisk/channel.h, channels/chan_gtalk.c, channels/chan_iax2.c, channels/chan_oss.c, main/channel.c, main/cdr.c, channels/chan_phone.c, channels/chan_misdn.c, channels/chan_skinny.c, channels/chan_features.c, channels/chan_h323.c, channels/chan_alsa.c, channels/chan_nbs.c, channels/chan_mgcp.c, apps/app_cdr.c, apps/app_voicemail.c: Merged revisions 60989 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60989 | murf | 2007-04-09 12:32:07 -0600 (Mon, 09 Apr 2007) | 1 line This is a big improvement over the current CDR fixes. It may still need refinement, but this won't have as many folks bothered. This also adds the mods from 1.4/r.61136; ........ 2007-04-09 22:49 +0000 [r61116] Russell Bryant * apps/app_dial.c: Remove unused instances of unnamed enums. 2007-04-09 20:01 +0000 [r61073] Olle Johansson * /, channels/chan_sip.c: Merged revisions 61072 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61072 | oej | 2007-04-09 21:58:17 +0200 (Mon, 09 Apr 2007) | 11 lines Merged revisions 61038 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61038 | oej | 2007-04-09 21:38:59 +0200 (Mon, 09 Apr 2007) | 3 lines - Don't send ActionID before Response: header. - Don't use a blank in an AMI header ........ ................ 2007-04-09 19:57 +0000 [r61065-61071] Kevin P. Fleming * main/minimime/mm_envelope.c, /: Merged revisions 61070 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61070 | kpfleming | 2007-04-09 14:55:14 -0500 (Mon, 09 Apr 2007) | 2 lines fix up some warnings found using --enable-dev-mode ........ * /, main/minimime/tests/CVS (removed), main/minimime/Doxyfile (removed), main/minimime/tests/messages/CVS (removed): Merged revisions 61062 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61062 | kpfleming | 2007-04-09 14:49:09 -0500 (Mon, 09 Apr 2007) | 2 lines remove some more stuff we don't need ........ 2007-04-09 19:06 +0000 [r61023] Jason Parker * /, apps/app_queue.c: Merged revisions 61022 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61022 | qwell | 2007-04-09 14:05:48 -0500 (Mon, 09 Apr 2007) | 4 lines Use the appropriate interface name with COMPLETECALLER. Issue 9395. ........ 2007-04-09 19:05 +0000 [r60985-61021] Olle Johansson * main/manager.c: Add hint to ExtensionStatus AMI event in manager * channels/chan_sip.c, CHANGES, channels/chan_iax2.c: use "ChannelType" in events to indicate which channel driver that generates the event. This replaces "ChannelDriver" and "Channel", previously used to indicate channel driver. ChannelType is more in line with "core show channeltypes" * res/res_jabber.c: Fix JabberEvents * /, res/res_jabber.c: Fix missing newline in JabberEvent 2007-04-09 17:23 +0000 [r60937] Jason Parker * /, apps/app_directory.c: Merged revisions 60936 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r60936 | qwell | 2007-04-09 12:22:59 -0500 (Mon, 09 Apr 2007) | 13 lines Merged revisions 60935 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r60935 | qwell | 2007-04-09 12:22:15 -0500 (Mon, 09 Apr 2007) | 5 lines Allow matching on names shorter than 3 chars. This also fixes the case where somebody wants to match on less then 3 chars. Issue 9071 ........ ................ 2007-04-09 16:30 +0000 [r60917] Dwayne M. Hubbard * UPGRADE.txt: updated UPGRADE.txt to include format_wav changes 2007-04-09 12:33 +0000 [r60898] Joshua Colp * channels/chan_sip.c: Make RTP session ID and session version generation random. (issue #9456 reported by tjardick) 2007-04-09 03:04 +0000 [r60848-60851] Tilghman Lesher * include/asterisk.h, /, main/asterisk.c: Merged revisions 60850 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r60850 | tilghman | 2007-04-08 22:01:12 -0500 (Sun, 08 Apr 2007) | 10 lines Merged revisions 60849 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r60849 | tilghman | 2007-04-08 21:49:06 -0500 (Sun, 08 Apr 2007) | 2 lines Don't check for error when lowering priority (according to the manpage, it should never happen anyway). It might could happen, though, if another thread messed with the priority, so safeguard against that (reported via -dev list). ........ ................ * channels/chan_local.c, /: Merged revisions 60847 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r60847 | tilghman | 2007-04-08 21:42:48 -0500 (Sun, 08 Apr 2007) | 10 lines Merged revisions 60846 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r60846 | tilghman | 2007-04-08 21:37:18 -0500 (Sun, 08 Apr 2007) | 2 lines Bug 9505 - If the return value for local_queue_frame is set, then p->lock is no longer valid. ........ ................ 2007-04-09 01:06 +0000 [r60763-60799] Joshua Colp * apps/app_dial.c, /: Merged revisions 60798 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r60798 | file | 2007-04-08 21:03:14 -0400 (Sun, 08 Apr 2007) | 10 lines Merged revisions 60797 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r60797 | file | 2007-04-08 20:59:29 -0400 (Sun, 08 Apr 2007) | 2 lines When calling a device that then forwards us elsewhere... we have to make our channels compatible if it is the only channel being dialed. (issue #9445 reported by marcelbarbulescu) ........ ................ * channels/chan_sip.c: Add counter for sip show registry CLI command. (issue #9352 reported by junky) * /, apps/app_queue.c: Merged revisions 60762 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60762 | file | 2007-04-08 13:04:44 -0400 (Sun, 08 Apr 2007) | 2 lines Allow app_queue to use MONITOR_EXEC even if MONITOR_OPTIONS is not set. (issue #9495 reported by cduffy) ........ 2007-04-08 14:23 +0000 [r60662-60715] Tilghman Lesher * /, apps/app_macro.c: Merged revisions 60713 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r60713 | tilghman | 2007-04-08 09:14:29 -0500 (Sun, 08 Apr 2007) | 10 lines Merged revisions 60711 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r60711 | tilghman | 2007-04-08 09:00:22 -0500 (Sun, 08 Apr 2007) | 2 lines Gosub called within a Macro resets the arguments improperly and causes general weirdness. (Issue 8329) ........ ................ * /, formats/format_wav.c, main/http.c: Merged revisions 60712 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60712 | tilghman | 2007-04-08 09:12:00 -0500 (Sun, 08 Apr 2007) | 2 lines Fix --enable-dev-mode ........ * /, main/file.c: Merged revisions 60661 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r60661 | tilghman | 2007-04-07 20:40:47 -0500 (Sat, 07 Apr 2007) | 10 lines Merged revisions 60660 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r60660 | tilghman | 2007-04-07 20:39:25 -0500 (Sat, 07 Apr 2007) | 2 lines Bug 9486 - memory leak when opening a filestream ........ ................ 2007-04-06 22:29 +0000 [r60641] Dwayne M. Hubbard * formats/format_wav.c: removed GAIN preprocessor definition, removed needsgain from struct wav_desc, removed unnecessary gain code from wav_read() and wav_write() 2007-04-06 21:43 +0000 [r60566-60623] Russell Bryant * main/minimime/Makefile: Filter out -Wundef so that the automatically generated C files will compile cleanly * main/minimime/mytest_files (removed), main/minimime/sys/CVS (removed), main/minimime/.cvsignore (removed), main/minimime/mm-docs (removed), main/minimime/test (removed): Remove a bunch of files that weren't supposed to get added. * main/minimime/mm-docs/html/mm__envelope_8c.html, main/minimime/tests/messages, include/asterisk/autoconfig.h.in, main/minimime/mm-docs/html/mm__context_8c.html, main/minimime/sys, main/minimime/tests/Makefile, main/minimime/tests/CVS/Root, main/minimime/sys/CVS/Entries, main/minimime/mm-docs/latex/mm__mimeutil_8c.tex, configure, main/strcompat.c, main/http.c, main/minimime/mm_error.c, main/minimime/mm-docs/html/globals_func.html, main/minimime/mm-docs/html/group__mimeutil.html, main/minimime/mm-docs/latex/doxygen.sty, main/minimime/mm_param.c, main/minimime/test/CVS, configure.ac, main/minimime/.cvsignore, main/minimime/mm_init.c, main/minimime/mm-docs/html/mm__queue_8h-source.html, main/minimime/mm-docs/html/mm__error_8c.html, main/minimime/mm-docs/html/tabs.css, main/minimime/mm_envelope.c, main/minimime/mimeparser.h, main/minimime/mimeparser.l, main/minimime/mm_context.c, main/minimime/mm-docs/html/group__mimepart.html, main/minimime/mm-docs/latex/group__envelope.tex, main/minimime/tests/messages/CVS, main/minimime/mm-docs/html/mm__contenttype_8c.html, main/minimime/mm-docs/html/pages.html, main/minimime/mm-docs/html/group__error.html, main/minimime/mm-docs/latex/group__context.tex, main/minimime/mimeparser.y, Makefile.moddir_rules, main/minimime/sys/mm_queue.h, main/minimime/mm-docs/html/bug.html, main/minimime/mm-docs/html/mimeparser_8tab_8h-source.html, main/minimime/tests/messages/CVS/Root, main/minimime/mm_mimepart.c, main/minimime/mm-docs/latex/Makefile, main/minimime/mm_internal.h, main/minimime/tests/CVS, main/minimime/mm-docs/latex/mm__param_8c.tex, main/minimime/tests/parse.c, main/minimime/mm_base64.c, main/minimime/mm.h, main/minimime/mm_header.c, main/minimime/mm-docs/latex/mm__parse_8c.tex, main/minimime/mm-docs/html/mimeparser_8h-source.html, main/minimime/mm-docs/html/files.html, main/minimime/mm-docs/latex/mm__contenttype_8c.tex, main/minimime/mm-docs/html/mm__mem_8h-source.html, main/minimime/mm_codecs.c, main/minimime/mm-docs/latex/mm__mimepart_8c.tex, main/minimime/mytest_files/mytest.c, main/minimime/mm-docs/html/mm__mimeutil_8c.html, main/minimime/mm-docs/latex/files.tex, main/minimime/test/CVS/Entries, main/minimime/mm-docs/latex/modules.tex, main/minimime/tests/messages/CVS/Repository, configs/http.conf.sample, main/minimime/mm_contenttype.c, main/minimime/tests/messages/test1.txt, main/minimime/mm-docs/html/mm__param_8c.html, main/minimime/tests/messages/test3.txt, main/minimime/tests/messages/test5.txt, main/minimime/tests/messages/test7.txt, main/minimime/mm-docs/html/group__contenttype.html, main/minimime/mm-docs, main/minimime/mytest_files/ast_postdata3.gz, main/minimime (added), main/minimime/Make.conf, main/minimime/mm-docs/latex/group__contenttype.tex, main/minimime/mm_warnings.c, main/minimime/mm_queue.h, main/minimime/mm-docs/html/mm__util_8c.html, main/minimime/mm-docs/html/doxygen.css, /, main/minimime/mm-docs/html/mm__internal_8h.html, main/minimime/tests/messages/CVS/Entries, main/minimime/Doxyfile, main/minimime/minimime.c, main/minimime/mimeparser.yy.c, main/minimime/tests/CVS/Entries.Log, main/minimime/test.sh, include/asterisk/compat.h, main/minimime/test/CVS/Repository, main/minimime/mm_mimeutil.c, main/minimime/tests, main/minimime/mm-docs/latex/group__mimepart.tex, main/minimime/tests/CVS/Entries, main/Makefile, main/minimime/mm-docs/latex/mm__envelope_8c.tex, main/minimime/mm-docs/latex/mm__util_8c.tex, main/minimime/mm-docs/latex/pages.tex, main/minimime/mm-docs/latex/group__mimeutil.tex, main/minimime/mm-docs/latex, main/minimime/mm-docs/html/mm_8h-source.html, main/minimime/Makefile, main/minimime/mm-docs/latex/mm__internal_8h.tex, main/minimime/mm-docs/refman.pdf, include/asterisk/manager.h, main/minimime/mm-docs/latex/mm__context_8c.tex, main/minimime/mm-docs/latex/group__param.tex, main/minimime/mm-docs/latex/group__codecs.tex, main/minimime/tests/create.c, main/minimime/mm_util.c, main/minimime/mm-docs/latex/bug.tex, main/minimime/mimeparser.tab.c, main/minimime/mm_util.h, main/minimime/mytest_files/ast_postdata, main/minimime/mm-docs/html/group__envelope.html, main/minimime/mm-docs/html/group__util.html, main/minimime/mimeparser.tab.h, main/minimime/mm-docs/html/mm__parse_8c.html, main/minimime/mm-docs/html, main/minimime/mm-docs/latex/group__util.tex, main/minimime/mm-docs/html/group__context.html, main/minimime/mm-docs/html/mm__internal_8h-source.html, main/minimime/mytest_files, main/minimime/mm-docs/html/mm__util_8h-source.html, main/minimime/sys/CVS, main/minimime/mm-docs/html/group__codecs.html, main/manager.c, main/minimime/sys/CVS/Repository, main/minimime/mm-docs/html/globals.html, main/minimime/mm-docs/html/mm__mimepart_8c.html, main/minimime/tests/CVS/Repository, main/minimime/mm-docs/html/index.html, main/minimime/mm-docs/html/modules.html, main/minimime/test, main/minimime/mytest_files/ast_postdata2, main/minimime/mm-docs/latex/group__error.tex, main/minimime/mm-docs/html/mm__header_8c.html, main/minimime/strlcpy.c, main/minimime/mm-docs/html/group__param.html, main/minimime/mm-docs/latex/refman.tex, main/minimime/mm_parse.c, main/minimime/mm-docs/latex/mm__header_8c.tex, main/minimime/mm-docs/latex/mm__error_8c.tex, main/minimime/mm_mem.c, main/minimime/mm-docs/html/mm__codecs_8c.html, main/minimime/tests/messages/test2.txt, main/minimime/tests/messages/test4.txt, main/minimime/sys/CVS/Root, main/minimime/tests/messages/test6.txt, main/minimime/test/CVS/Root, main/minimime/strlcat.c, main/minimime/mm_mem.h, main/minimime/mm-docs/latex/mm__codecs_8c.tex: Merged revisions 60603 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60603 | russell | 2007-04-06 15:58:43 -0500 (Fri, 06 Apr 2007) | 13 lines To be able to achieve the things that we would like to achieve with the Asterisk GUI project, we need a fully functional HTTP interface with access to the Asterisk manager interface. One of the things that was intended to be a part of this system, but was never actually implemented, was the ability for the GUI to be able to upload files to Asterisk. So, this commit adds this in the most minimally invasive way that we could come up with. A lot of work on minimime was done by Steve Murphy. He fixed a lot of bugs in the parser, and updated it to be thread-safe. The ability to check permissions of active manager sessions was added by Dwayne Hubbard. Then, hacking this all together and do doing the modifications necessary to the HTTP interface was done by me. ........ * /, apps/app_meetme.c: Merged revisions 60565 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60565 | russell | 2007-04-06 14:50:52 -0500 (Fri, 06 Apr 2007) | 3 lines When a station picks up a trunk that was on hold, make the hints reflect that nobody has the trunk on hold anymore. ........ 2007-04-06 19:26 +0000 [r60531] Olle Johansson * channels/chan_sip.c: Use the same parameter to the two "Registry" AMI events - ChannelDriver 2007-04-06 18:59 +0000 [r60522] Russell Bryant * /, apps/app_meetme.c: Merged revisions 60521 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60521 | russell | 2007-04-06 13:58:46 -0500 (Fri, 06 Apr 2007) | 16 lines Fix a few problems with SLA. (issue #9459, reported by francesco_r, fixed by me) * The original behavior was that if one station put a call on hold, another one picked it up, and then hung up, the code would still consider the call on hold by the first station, so the trunk would not be hung up. However, to better comply with what most people seem to expect it to behave, it will now hang up the trunk. * Fix a problem with "barge=no". This was only intended to prevent people from joining calls that are in progress. However, it also prevented other people from picking up a call that was on hold. This has been fixed. * When there are no active stations on a trunk and it is on hold, the code now indicates the HOLD and UNHOLD conditions to the trunk channel. This allows music on hold to be played to the trunk when it is on hold. ........ 2007-04-06 18:26 +0000 [r60486-60487] Matthew Fredrickson * channels/chan_zap.c, /: Merged revisions 60485 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60485 | mattf | 2007-04-06 13:21:52 -0500 (Fri, 06 Apr 2007) | 2 lines Make sure we check the faxdetect option before doing fax processing ........ * channels/chan_zap.c, /: Merged revisions 60459 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r60459 | mattf | 2007-04-06 12:32:31 -0500 (Fri, 06 Apr 2007) | 10 lines Merged revisions 60456 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r60456 | mattf | 2007-04-06 12:03:15 -0500 (Fri, 06 Apr 2007) | 2 lines There should only be one code path for doing DTMF conditionals on channels. This fixes it. ........ ................ 2007-04-06 14:53 +0000 [r60400] Kevin P. Fleming * /, codecs/codec_zap.c: Merged revisions 60399 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r60399 | kpfleming | 2007-04-06 09:49:51 -0500 (Fri, 06 Apr 2007) | 10 lines Merged revisions 60398 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r60398 | kpfleming | 2007-04-06 09:41:37 -0500 (Fri, 06 Apr 2007) | 2 lines remove undocumented 'cardsmode' parameter and stop searching for transcoders during reload() ........ ................ 2007-04-06 01:29 +0000 [r60362-60363] Joshua Colp * include/asterisk/speech.h, res/res_speech.c: Major res_speech cleanup. It looks much better now! * /, include/asterisk/speech.h, res/res_speech.c, apps/app_speech_utils.c: Merged revisions 60361 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60361 | file | 2007-04-05 22:14:00 -0300 (Thu, 05 Apr 2007) | 2 lines Add support for returning different types of results (ie: NBest). ........ 2007-04-05 23:08 +0000 [r60326] Dwayne M. Hubbard * /, formats/format_wav.c: Merged revisions 60325 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60325 | dhubbard | 2007-04-05 17:58:01 -0500 (Thu, 05 Apr 2007) | 1 line modified default GAIN for issue 5823, thanks jrwalliker ........ 2007-04-05 22:40 +0000 [r60324] Steve Murphy * configs/cdr_custom.conf.sample, /, configs/cdr.conf.sample: Merged revisions 60323 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60323 | murf | 2007-04-05 16:35:11 -0600 (Thu, 05 Apr 2007) | 1 line Added some clarification to the example configs for CDRs, on how to select a backend. Also, made cdr-csv the default if you 'make samples', and no other changes. ........ 2007-04-05 16:11 +0000 [r60269] Jason Parker * /, apps/app_voicemail.c: Merged revisions 60268 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r60268 | qwell | 2007-04-05 11:10:48 -0500 (Thu, 05 Apr 2007) | 13 lines Merged revisions 60267 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r60267 | qwell | 2007-04-05 11:09:41 -0500 (Thu, 05 Apr 2007) | 5 lines Just because we can't find the voicemail configuration file, doesn't mean that the module failed to load. The user could be using realtime. Issue #9473 ........ ................ 2007-04-05 15:48 +0000 [r60266] Russell Bryant * /, main/http.c: Merged revisions 60265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60265 | russell | 2007-04-05 10:47:17 -0500 (Thu, 05 Apr 2007) | 2 lines Add the MIME type for gif by request from Pari ........ 2007-04-05 12:57 +0000 [r60215] Joshua Colp * /, channels/chan_sip.c: Merged revisions 60214 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r60214 | file | 2007-04-05 08:55:02 -0400 (Thu, 05 Apr 2007) | 10 lines Merged revisions 60213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r60213 | file | 2007-04-05 08:52:50 -0400 (Thu, 05 Apr 2007) | 2 lines Only unlock our pvt and net locks if we are actually going to try to lock the owner again. (issue #9472 reported by zoa) ........ ................ 2007-04-04 23:45 +0000 [r60193] Dwayne M. Hubbard * main/callerid.c: ast_shrink_phone_number() must ignore whitespace, otherwise my CIDCO callerid box gets LINE ERROR 2007-04-04 17:41 +0000 [r60011-60141] Russell Bryant * main/manager.c, /: Merged revisions 60137 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r60137 | russell | 2007-04-04 12:40:10 -0500 (Wed, 04 Apr 2007) | 14 lines Merged revisions 60134 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r60134 | russell | 2007-04-04 12:38:47 -0500 (Wed, 04 Apr 2007) | 6 lines It is valid to redirect channels via the manager interface that are not in the UP state. Instead of checking for that to prevent to ensure a dead channel doesn't get redirected, just use the ast_check_hangup() API call. (issue #9457, reported by Callmewind, patch by me) (related to issue #8977) ........ ................ * /, channels/chan_sip.c: Merged revisions 60112 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60112 | russell | 2007-04-04 11:49:45 -0500 (Wed, 04 Apr 2007) | 3 lines Add a Content-Length of 0 to the response built by transmit_response_with_unsupported(). (issue #9454, reported by makoto, fixed by me) ........ * /, channels/chan_sip.c: Merged revisions 60088 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r60088 | russell | 2007-04-04 11:39:04 -0500 (Wed, 04 Apr 2007) | 12 lines Merged revisions 60083 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r60083 | russell | 2007-04-04 11:37:04 -0500 (Wed, 04 Apr 2007) | 4 lines Fix the return value of handle_common_options() so that it always properly indicates whether it handled the option or not. (issue #9455, reported by Netview, fixed by me) ........ ................ * /, apps/app_meetme.c: Merged revisions 60069 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60069 | russell | 2007-04-04 11:26:23 -0500 (Wed, 04 Apr 2007) | 4 lines Fix a problem where if a trunk was hung up while it was on hold, all of the hints would reflect the line still on hold, even though it should reflect that it is back to not in use. (issue #9459, reported by francesco_r, fixed by me) ........ * channels/chan_jingle.c, channels/chan_gtalk.c, doc/rtp-packetization.txt: Add support for RTP packetization in chan_jingle and chan_gtalk. (issue #9416, phsultan) 2007-04-03 19:43 +0000 [r59969] Joshua Colp * /, apps/app_speech_utils.c: Merged revisions 59963 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59963 | file | 2007-04-03 15:40:59 -0400 (Tue, 03 Apr 2007) | 2 lines Don't clash when a person both speaks and uses DTMF. ........ 2007-04-03 19:17 +0000 [r59854-59940] Russell Bryant * /, channels/chan_sip.c: Merged revisions 59939 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r59939 | russell | 2007-04-03 14:16:53 -0500 (Tue, 03 Apr 2007) | 12 lines Merged revisions 59938 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59938 | russell | 2007-04-03 14:15:04 -0500 (Tue, 03 Apr 2007) | 4 lines Don't attempt to report configuration errors in build_user(). oej pointed out that for a "friend" entry, this won't work, because all user options are valid for peers, but not the other way around. ........ ................ * /, channels/chan_sip.c: Merged revisions 59936 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r59936 | russell | 2007-04-03 13:55:57 -0500 (Tue, 03 Apr 2007) | 11 lines Merged revisions 59916 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59916 | russell | 2007-04-03 13:43:54 -0500 (Tue, 03 Apr 2007) | 3 lines Make chan_sip report when it encounters an unknown option. (issue #9440, reported by nightcrawler) ........ ................ * channels/chan_sip.c: Remove a duplicate function prototype. (issue #9444, junky) * /, main/app.c: Merged revisions 59887 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r59887 | russell | 2007-04-03 13:01:49 -0500 (Tue, 03 Apr 2007) | 13 lines Merged revisions 59886 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59886 | russell | 2007-04-03 12:58:19 -0500 (Tue, 03 Apr 2007) | 5 lines When doing a built-in blind or attended transfer, restore the ability to use '#' to terminate the number and immediately do the transfer instead of having to dial the number and just wait for the feature digit timeout. (issue #8366, xueliangliang) ........ ................ * Makefile, /: Merged revisions 59853 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59853 | russell | 2007-04-03 11:03:35 -0500 (Tue, 03 Apr 2007) | 1 line Ensure that menuselect gets executed in dependency check mode every time you run make. ........ 2007-04-03 11:15 +0000 [r59805] Nadi Sarrar * /, channels/misdn/chan_misdn_config.h, channels/misdn_config.c: Merged revisions 59804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r59804 | nadi | 2007-04-03 13:02:46 +0200 (Di, 03 Apr 2007) | 15 lines Merged revisions 59788,59803 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59788 | nadi | 2007-04-03 11:37:00 +0200 (Di, 03 Apr 2007) | 2 lines Use the new sysfs way of mISDN 1.2 to check if a port is NT or not. ........ r59803 | nadi | 2007-04-03 12:40:58 +0200 (Di, 03 Apr 2007) | 2 lines ptp is the 5th bit, not the 4th. ........ ................ 2007-04-02 19:01 +0000 [r59725] Joshua Colp * /, apps/app_voicemail.c: Merged revisions 59724 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r59724 | file | 2007-04-02 14:58:24 -0400 (Mon, 02 Apr 2007) | 10 lines Merged revisions 59723 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59723 | file | 2007-04-02 14:55:25 -0400 (Mon, 02 Apr 2007) | 2 lines Increase the maximum size for a string of mailboxes to 1024. (issue #9270 reported by rtucker) ........ ................ 2007-04-02 17:40 +0000 [r59693] Russell Bryant * channels/chan_iax2.c: This hashing code is still causing some random crashes on my system, and probably others, too. I don't really have time to work on it at the moment, so I am just going to revert it for now. 2007-04-02 17:38 +0000 [r59692] Steve Murphy * /, pbx/pbx_ael.c: Merged revisions 59688 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59688 | murf | 2007-04-02 11:31:32 -0600 (Mon, 02 Apr 2007) | 1 line continue in for-loop should go to the incrementer, not the test. As per 9435, thanks to marcelbarbulescu ........ 2007-04-02 16:08 +0000 [r59655] Russell Bryant * /, main/netsock.c: Merged revisions 59654 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r59654 | russell | 2007-04-02 10:39:07 -0500 (Mon, 02 Apr 2007) | 14 lines Merged revisions 59608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59608 | russell | 2007-04-01 17:35:25 -0500 (Sun, 01 Apr 2007) | 6 lines Add the SO_REUSEADDR flag to sockets handled by netsock. This is needed by the patch that went in for issue 7874. chan_iax2 needs to be able to create socket that is lisetning on INADDR_ANY, but also be able to bind sockets to specific addresses. (Thanks to Stevenson on the asterisk-dev mailing list for explaining why this flag was needed.) ........ ................ 2007-03-30 22:54 +0000 [r59574] Jason Parker * /, configure, main/Makefile, acinclude.m4: Merged revisions 59573 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59573 | qwell | 2007-03-30 17:50:31 -0500 (Fri, 30 Mar 2007) | 2 lines Add linux-uclibc host arch..."thingy". Sorry, I don't know what it's called... ........ 2007-03-30 20:54 +0000 [r59555] Matthew Fredrickson * channels/chan_zap.c: Update to support multiple CIC groups and DPCs per linkset. 2007-03-30 17:57 +0000 [r59453-59523] Steve Murphy * main/cdr.c, main/channel.c, main/pbx.c, /, res/res_features.c, include/asterisk/cdr.h: Merged revisions 59522 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59522 | murf | 2007-03-30 11:51:17 -0600 (Fri, 30 Mar 2007) | 1 line several changes via kpflemings review ........ * main/cdr.c, main/channel.c, main/pbx.c, /, res/res_features.c, include/asterisk/cdr.h: Merged revisions 59486 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59486 | murf | 2007-03-30 08:11:59 -0600 (Fri, 30 Mar 2007) | 1 line These mods fix CDR issues from 8221, 8593, 8680, 8743, and perhaps others. Mainly with CDRs generated from transfer situations. ........ * /, configs/extensions.conf.sample: Merged revisions 59452 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59452 | murf | 2007-03-29 18:56:36 -0600 (Thu, 29 Mar 2007) | 1 line A small clarification to keep bugs from being filed, and confusion from rising, if clearglobalvars is set, and globals are set in the AEL file. (9419) ........ 2007-03-29 23:27 +0000 [r59364-59433] Russell Bryant * apps/app_voicemail.c: Reduce the ridiculous number of variables used in the load_config() function by just having one that can be re-used. There is no functional change here (that is intentional, anyway!). * CHANGES, apps/app_voicemail.c: Add the ability for the "voicemail show users" CLI command to show users configured in realtime. * channels/chan_iax2.c: Fix an issue with hashing iax2 pvt structures that caused random crashes on systems under heavy load such as IAXtel. (possibly related to issue #9403) * /, res/res_jabber.c: Merged revisions 59363 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59363 | russell | 2007-03-29 12:43:52 -0500 (Thu, 29 Mar 2007) | 6 lines When building a response to a subscription, the "from" must be the full Jabber ID. This fixes some problems where jabber users are not able to add their Asterisk account to their user list, since they are unable to get Asterisk to approve their subscription. (issue #8210, reported by caspy, and verified by bradtem) ........ 2007-03-29 17:42 +0000 [r59362] Joshua Colp * /, apps/app_meetme.c: Merged revisions 59361 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r59361 | file | 2007-03-29 13:38:55 -0400 (Thu, 29 Mar 2007) | 10 lines Merged revisions 59360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59360 | file | 2007-03-29 13:33:58 -0400 (Thu, 29 Mar 2007) | 2 lines Keep a global array of variables indicating whether certain conference rooms are in use. This ensures that two people going into a new dynamic conference when the 'e' option is set don't go into the same conference room. (issue #8835 reported by eliel) ........ ................ 2007-03-29 17:20 +0000 [r59305-59359] Russell Bryant * /, main/rtp.c: Merged revisions 59358 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r59358 | russell | 2007-03-29 12:17:41 -0500 (Thu, 29 Mar 2007) | 13 lines Merged revisions 59357 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59357 | russell | 2007-03-29 12:14:33 -0500 (Thu, 29 Mar 2007) | 5 lines If an error occurs when reading from an RTP socket, and the error code does not indicate that we should try again, then return NULL instead of a "null frame". This will prevent Asterisk from trying over and over again, and eventually causing the system to crash. (issue #8285, john) ........ ................ * /, channels/chan_iax2.c: Merged revisions 59341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59341 | russell | 2007-03-29 11:55:39 -0500 (Thu, 29 Mar 2007) | 8 lines When the IAX2 read callback gets called, return NULL instead of a "null frame". This will cause Asterisk to hangup the call instead of keep trying whatever it was doing. Under normal conditions, this function would *never* be called. However, the author of this patch says an error will occur that will cause it to get called every 100 thousand calls or so. When this does happen, it puts the channel in a loop that eventually brings down the system. So, hangup up the call is certainly a better alternative. (issue #8286, john) ........ * Makefile, /: Merged revisions 59304 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59304 | russell | 2007-03-29 11:25:41 -0500 (Thu, 29 Mar 2007) | 2 lines Export the GTK2 library and include information to sub Makefiles. ........ 2007-03-29 16:08 +0000 [r59303] Tilghman Lesher * /, cdr/cdr_odbc.c: Merged revisions 59302 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r59302 | tilghman | 2007-03-29 11:07:05 -0500 (Thu, 29 Mar 2007) | 11 lines Merged revisions 59301 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59301 | tilghman | 2007-03-29 11:04:46 -0500 (Thu, 29 Mar 2007) | 3 lines Issue 9415 - No point to getting a diagnostic field if we aren't doing anything with the information. (Plus, it tends to crash the Postgres ODBC driver.) ........ ................ 2007-03-28 03:40 +0000 [r59290] Tilghman Lesher * /, res/res_odbc.c: Merged revisions 59289 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59289 | tilghman | 2007-03-27 22:38:09 -0500 (Tue, 27 Mar 2007) | 2 lines Another crash that I thought we had fixed already - Issue 9396 ........ 2007-03-28 00:09 +0000 [r59286] Dwayne M. Hubbard * channels/chan_zap.c: added filtering options to 'zap show channels' to implement functionality described in issue 6520 2007-03-27 23:38 +0000 [r59282-59285] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 59284 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r59284 | tilghman | 2007-03-27 18:37:31 -0500 (Tue, 27 Mar 2007) | 10 lines Merged revisions 59283 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59283 | tilghman | 2007-03-27 18:36:49 -0500 (Tue, 27 Mar 2007) | 2 lines Oops ........ ................ * /, apps/app_voicemail.c: Merged revisions 59281 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r59281 | tilghman | 2007-03-27 18:32:46 -0500 (Tue, 27 Mar 2007) | 10 lines Merged revisions 59280 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59280 | tilghman | 2007-03-27 18:31:20 -0500 (Tue, 27 Mar 2007) | 2 lines Fix a few remaining bad mmap(2) return values ........ ................ 2007-03-27 23:22 +0000 [r59274-59279] Russell Bryant * /, apps/app_directory.c: Merged revisions 59278 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r59278 | russell | 2007-03-27 18:20:22 -0500 (Tue, 27 Mar 2007) | 11 lines Merged revisions 59277 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59277 | russell | 2007-03-27 18:19:41 -0500 (Tue, 27 Mar 2007) | 3 lines Fix the check of the return value from mmap(). Thanks to Corydon for catching this one. ........ ................ * /, apps/app_directory.c: Merged revisions 59275 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59275 | russell | 2007-03-27 18:16:27 -0500 (Tue, 27 Mar 2007) | 3 lines Fix app_directory to actually compile with ODBC_STORAGE, and update the code to the latest res_odbc API. ........ * /, apps/Makefile: Merged revisions 59273 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59273 | russell | 2007-03-27 18:02:12 -0500 (Tue, 27 Mar 2007) | 4 lines Fix app_directory when ODBC_STORAGE is being used. The Makefile did not properly ensure that this information got copied from what was selected for app_voicemail. (issue #9224) ........ 2007-03-27 20:11 +0000 [r59272] Joshua Colp * channels/chan_zap.c: Use better english. Renegotiate! Repeat after me: renegotiate. 2007-03-27 18:21 +0000 [r59264] Steve Murphy * /, pbx/pbx_ael.c: Merged revisions 59261 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59261 | murf | 2007-03-27 12:16:32 -0600 (Tue, 27 Mar 2007) | 1 line via 9373 (duplicate context in AEL crashes asterisk), kpfleming pointed on asterisk-dev, that DECLINE in this case the proper thing to do. This change now has it doing the proper thing. ........ 2007-03-27 18:18 +0000 [r59257-59263] Russell Bryant * /, channels/chan_sip.c: Merged revisions 59262 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59262 | russell | 2007-03-27 13:17:47 -0500 (Tue, 27 Mar 2007) | 3 lines Fix the check that ensures that the CHANNEL function's first argument is "rtpqos". Thanks, Corydon. :) ........ * /, channels/chan_iax2.c: Merged revisions 59259 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r59259 | russell | 2007-03-27 13:05:46 -0500 (Tue, 27 Mar 2007) | 12 lines Merged revisions 59258 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59258 | russell | 2007-03-27 13:04:02 -0500 (Tue, 27 Mar 2007) | 4 lines Fix the use of the "sourceaddress" option when "bindaddr" is set to 0.0.0.0 instead of having each interface explicitly listed. (issue #7874, patch by stevens) ........ ................ * /, channels/chan_sip.c, funcs/func_channel.c: Merged revisions 59256 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59256 | russell | 2007-03-27 11:20:53 -0500 (Tue, 27 Mar 2007) | 4 lines Convert the RTPQOS function to just be additional parameter of the CHANNEL function. This way, it will be possible for other RTP based channel drivers to expose this information in the future. ........ 2007-03-27 14:09 +0000 [r59233-59253] Steve Murphy * include/asterisk/config.h: Enhancement via 8118: Realtime API extension: add methods store_func and destroy_func, to make Realtime a complete database abstraction * pbx/ael/ael-test/ael-test18/extensions.ael (added), pbx/ael/ael-test/ael-test18 (added), pbx/ael/ael-test/ref.ael-test18 (added): added the no. 18 regression test * pbx/ael/ael-test/ael-test19/extensions.ael (added), pbx/ael/ael-test/ref.ael-vtest17, pbx/ael/ael-test/ael-test19 (added), pbx/ael/ael-test/ref.ael-test7, pbx/ael/ael-test/ref.ael-test19 (added), pbx/ael/ael-test/ref.ael-vtest13: updated the regressions with regards to 9373, the crash on double contexts, and brought other regressions up to date * /, pbx/pbx_ael.c: Merged revisions 59228 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59228 | murf | 2007-03-26 15:41:32 -0600 (Mon, 26 Mar 2007) | 1 line fix for 9373 (duplicate context in AEL crashes asterisk). I turned a duplicate context from a WARNING to an ERROR. Now you get a module load failure, and asterisk just exits. That's better than a crash, right\? ........ 2007-03-26 21:46 +0000 [r59229-59231] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 59227 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59227 | tilghman | 2007-03-26 16:37:41 -0500 (Mon, 26 Mar 2007) | 2 lines Change this to a single dp function to make oej happy. ........ 2007-03-26 20:27 +0000 [r59226] Steve Murphy * /, main/config.c: Merged revisions 59225 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59225 | murf | 2007-03-26 14:06:12 -0600 (Mon, 26 Mar 2007) | 1 line Fix for 9257; by eliminating the globals in main/config.c, we make it thread-safe, which is a minimum requirement. ........ 2007-03-26 19:35 +0000 [r59224] Joshua Colp * /, apps/app_speech_utils.c: Merged revisions 59223 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59223 | file | 2007-03-26 16:34:14 -0300 (Mon, 26 Mar 2007) | 2 lines Add ability to specify no timeout. This means as soon as the prompt is done playing it moves on to the next priority. ........ 2007-03-26 18:34 +0000 [r59216-59218] Russell Bryant * /: Merged revisions 59217 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59217 | russell | 2007-03-26 13:33:50 -0500 (Mon, 26 Mar 2007) | 4 lines Somehow the code for building the email for voicemail got out of sync. This change makes a few tweaks to get 1.4 in sync with trunk. (issue #9301) ........ * /, apps/app_meetme.c: Merged revisions 59215 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59215 | russell | 2007-03-26 13:28:29 -0500 (Mon, 26 Mar 2007) | 3 lines Fix some codec negotiation problems when CallerID support is not enabled in SLA. (issue #9308, reported by twilson) ........ 2007-03-26 18:14 +0000 [r59214] Joshua Colp * /, apps/app_speech_utils.c: Merged revisions 59213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59213 | file | 2007-03-26 14:13:06 -0400 (Mon, 26 Mar 2007) | 2 lines Make SpeechBackground obey the digit timeout value. ........ 2007-03-26 17:57 +0000 [r59211] Russell Bryant * channels/chan_sip.c: Merged revisions 59209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59209 | russell | 2007-03-26 12:53:07 -0500 (Mon, 26 Mar 2007) | 1 line Rename the new dialplan functions to match the variable name ........ 2007-03-26 17:56 +0000 [r59210] Steve Murphy * /, main/ast_expr2f.c, pbx/ael/ael.flex, main/ast_expr2.fl: Merged revisions 59206 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59206 | murf | 2007-03-26 11:38:29 -0600 (Mon, 26 Mar 2007) | 1 line A fix for the flex input files, DONT_COMPILE, and STANDALONE_AEL ........ 2007-03-26 17:51 +0000 [r59208] Russell Bryant * /, channels/chan_sip.c, include/asterisk/rtp.h, main/rtp.c: Merged revisions 59207 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59207 | russell | 2007-03-26 12:45:55 -0500 (Mon, 26 Mar 2007) | 7 lines The AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in some because they get set in sip_hangup. So, there are common situations where the variables will not be available in the dialplan at all. So, this patch provides an alternate method for getting to this information by introducing AUDIORTPQOS and VIDEORTPQOS dialplan functions. (issue #9370, patch by Corydon76, with some testing by blitzrage) ........ 2007-03-26 16:48 +0000 [r59204-59205] Matthew Fredrickson * channels/chan_zap.c: Fix bug in which parameter type we are passing. This shouldn't be a problem since both types are the same underneath. * channels/chan_zap.c: Small API related SS7 updates. 2007-03-26 15:59 +0000 [r59203] Nadi Sarrar * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, configure, include/asterisk/autoconfig.h.in, channels/misdn/Makefile, channels/misdn/chan_misdn_config.h, configure.ac, channels/misdn/isdn_lib.c, channels/misdn_config.c: Merged revisions 59202 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59202 | nadi | 2007-03-26 17:25:53 +0200 (Mo, 26 Mär 2007) | 4 lines * mISDN >= 1.2 provides a dsp pipeline for i.e. echo cancellation modules, make chan_misdn use it. * add a check for linux/mISDNdsp.h to configure.ac and update the autogenerated files: 'configure', 'autoconfig.h.in' (the 'configure' script was not in sync with the latest configure.ac, so the diff is a bit bigger than expected). ........ 2007-03-26 15:20 +0000 [r59201] Joshua Colp * /, pbx/ael/ael_lex.c: Merged revisions 59200 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59200 | file | 2007-03-26 11:16:29 -0400 (Mon, 26 Mar 2007) | 2 lines Have ast_copy_string magically appear in the aelparse binary! DONT_OPTIMIZE should now work once again. ........ 2007-03-24 01:42 +0000 [r59191-59196] Joshua Colp * /, channels/chan_sip.c: Merged revisions 59195 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r59195 | file | 2007-03-23 21:39:44 -0400 (Fri, 23 Mar 2007) | 10 lines Merged revisions 59194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59194 | file | 2007-03-23 21:35:49 -0400 (Fri, 23 Mar 2007) | 2 lines Only try to handle a response if it has a response code. (ASA-2007-011) ........ ................ * doc/modules.txt: Update modules.txt to new loader. (issue #9358 reported by eliel) 2007-03-23 16:17 +0000 [r59190] Steve Murphy * /, apps/app_macro.c: Merged revisions 59188 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r59188 | murf | 2007-03-23 10:09:01 -0600 (Fri, 23 Mar 2007) | 9 lines Merged revisions 59186 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59186 | murf | 2007-03-23 09:57:26 -0600 (Fri, 23 Mar 2007) | 1 line Added a few words in the Macro doc strings about the behavior of macros with hangups (et al.), as per 9337 ........ ................ 2007-03-22 23:41 +0000 [r59181-59183] Kevin P. Fleming * /, channels/chan_sip.c: Merged revisions 59182 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59182 | kpfleming | 2007-03-22 16:40:01 -0700 (Thu, 22 Mar 2007) | 2 lines don't allow string input to overrun the buffer to hold it (ASA-2007-010) ........ * channels/chan_misdn.c, /: Merged revisions 59180 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59180 | kpfleming | 2007-03-22 16:34:22 -0700 (Thu, 22 Mar 2007) | 2 lines remove variables that are no longer used (--enable-dev-mode is good, developers should be using it) ........ 2007-03-22 14:48 +0000 [r59146] Steve Murphy * utils/Makefile, /: Merged revisions 59145 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59145 | murf | 2007-03-22 08:40:53 -0600 (Thu, 22 Mar 2007) | 1 line The stuff in utils was compiling with -O6 even if DONT_OPTIMIZE is set in menuconfig. Added the include to fix that ........ 2007-03-21 18:10 +0000 [r59080-59090] Joshua Colp * /, main/http.c: Merged revisions 59089 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59089 | file | 2007-03-21 14:08:57 -0400 (Wed, 21 Mar 2007) | 2 lines Add svg mimetype for pari. ........ * /, res/res_monitor.c: Merged revisions 59087 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r59087 | file | 2007-03-21 14:04:58 -0400 (Wed, 21 Mar 2007) | 10 lines Merged revisions 59086 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59086 | file | 2007-03-21 14:03:20 -0400 (Wed, 21 Mar 2007) | 2 lines Indicate the filename changed when it is changed. (issue #9311 reported by jsmith) ........ ................ * channels/chan_sip.c: Minor tweak. Only queue up an unhold control frame if we are actually on hold. This would have shown itself when a call was initially being setup and the SDP data was being parsed in. * /, channels/chan_sip.c: Merged revisions 59081 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59081 | file | 2007-03-20 23:25:48 -0400 (Tue, 20 Mar 2007) | 2 lines Until we can do media level parsing for sendrecv/etc just use the first value found. This crept up when a phone was offered audio+video and returned an inactive video stream. chan_sip thought the phone said to put the person on hold but that was totally wrong. (issue #9319 reported by benbrown) ........ * main/db.c: Make the database show command spit out how many results it got. (issue #9332 reported by junky) 2007-03-20 21:06 +0000 [r59079] Tilghman Lesher * /, main/logger.c: Merged revisions 59078 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59078 | tilghman | 2007-03-20 16:04:52 -0500 (Tue, 20 Mar 2007) | 2 lines Fix defines for inline stack backtraces (only used by developers anyway) ........ 2007-03-20 20:44 +0000 [r59077] Joshua Colp * /, channels/iax2-parser.c: Merged revisions 59076 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59076 | file | 2007-03-20 16:42:46 -0400 (Tue, 20 Mar 2007) | 2 lines Copy len variable as well, should fix remaining IAX2 DTMF issues. ........ 2007-03-20 18:18 +0000 [r59071-59073] Steve Murphy * pbx/pbx_ael.c, include/asterisk/ael_structs.h: The fix for the AEL <> (bug 9316) is here... * /: blocking 59070... it was just a repair, doesn't need to be here * /: blocking 59069... will commit these changes with separate patch 2007-03-19 22:32 +0000 [r59051] Joshua Colp * main/loader.c: It is possible for mod to become invalid after we unload it (if it's a dynamic module) so move it around a bit. 2007-03-19 22:31 +0000 [r59050] Tilghman Lesher * /, funcs/func_strings.c: Merged revisions 59049 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59049 | tilghman | 2007-03-19 17:29:56 -0500 (Mon, 19 Mar 2007) | 2 lines Oops, this should have been a %d all along ........ 2007-03-19 15:43 +0000 [r59041] Tilghman Lesher * configs/sip_notify.conf.sample, /: Merged revisions 59040 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59040 | tilghman | 2007-03-19 10:42:26 -0500 (Mon, 19 Mar 2007) | 2 lines Fix unescaped semicolon (reported via -dev list) ........ 2007-03-18 20:39 +0000 [r59038] Olle Johansson * /, channels/chan_sip.c: Merged revisions 59037 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59037 | oej | 2007-03-18 21:37:06 +0100 (Sun, 18 Mar 2007) | 3 lines Issue #9313, Asterisk crash on SIP return code 0 (reported by qwerty1979) (ASA-2007-011) ........ 2007-03-18 16:59 +0000 [r59036] BJ Weschke * /, apps/app_followme.c: Merged revisions 59035 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59035 | bweschke | 2007-03-18 12:36:44 -0400 (Sun, 18 Mar 2007) | 3 lines Don't return a non-zero return code if the profile doesn't exist, to match what the documentation says it already does. (#9307 Reported by kkiely) ........ 2007-03-16 16:14 +0000 [r58995] Joshua Colp * /, apps/app_page.c: Merged revisions 58992 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58992 | file | 2007-03-16 12:12:28 -0400 (Fri, 16 Mar 2007) | 2 lines Wait for the async thread to exit when hanging up all of the paged phones under all circumstances. (issue #9181 reported by PhilSmith) ........ 2007-03-16 01:43 +0000 [r58954-58958] Russell Bryant * /, configs/sla.conf.sample: Merged revisions 58957 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58957 | russell | 2007-03-15 20:42:37 -0500 (Thu, 15 Mar 2007) | 1 line fix a couple SLA documentation references ........ * /, build_tools/prep_tarball: Merged revisions 58953 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58953 | russell | 2007-03-15 20:12:40 -0500 (Thu, 15 Mar 2007) | 2 lines Add the --pdf option to the usage of rubber in prep_tarball ........ 2007-03-16 00:04 +0000 [r58949-58950] Tilghman Lesher * main/pbx.c, /, doc/ast_appdocs.tex: Merged revisions 58946 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58946 | tilghman | 2007-03-15 18:52:48 -0500 (Thu, 15 Mar 2007) | 2 lines Refashion dump command to match common syntax and update the resulting appdocs TeX file ........ * main/pbx.c: Fix trunk so that it compiles again 2007-03-15 23:56 +0000 [r58942-58948] Russell Bryant * Makefile, /, build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Merged revisions 58947 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58947 | russell | 2007-03-15 18:53:26 -0500 (Thu, 15 Mar 2007) | 3 lines Add configure script checking for GTK2 and some additional Makefile targets to support gmenuselect ........ * /, doc/asterisk.tex: Merged revisions 58941 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58941 | russell | 2007-03-15 18:24:09 -0500 (Thu, 15 Mar 2007) | 1 line add a link to the rubber homepage ........ 2007-03-15 22:52 +0000 [r58936-58938] Russell Bryant * Makefile, /, doc/asterisk.tex: Merged revisions 58937 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58937 | russell | 2007-03-15 17:51:29 -0500 (Thu, 15 Mar 2007) | 2 lines Add Asterisk version information to the generated PDF ........ * /, build_tools/prep_tarball: Merged revisions 58935 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58935 | russell | 2007-03-15 17:35:52 -0500 (Thu, 15 Mar 2007) | 2 lines have prep_tarball attempt to build asterisk.pdf ........ 2007-03-15 22:33 +0000 [r58934] Tilghman Lesher * /, funcs/func_realtime.c: Merged revisions 58933 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58933 | tilghman | 2007-03-15 17:32:33 -0500 (Thu, 15 Mar 2007) | 2 lines Function works fine, but the documentation is backwards. ........ 2007-03-15 22:29 +0000 [r58932] Russell Bryant * doc/manager.txt (removed), doc/misdn.txt (removed), doc/jitterbuffer.tex (added), /, doc/billing.txt (removed), doc/extensions.tex (added), doc/queues-with-callback-members.tex (added), doc/localchannel.txt (removed), doc/cdrdriver.txt (removed), doc/00README.1st (removed), doc/ajam.tex (added), doc/manager.tex (added), doc/misdn.tex (added), doc/freetds.txt (removed), doc/odbcstorage.txt (removed), configure, doc/model.txt (removed), doc/cygwin.txt (removed), doc/sla.tex, doc/billing.tex (added), doc/ael.txt (removed), doc/channelvariables.txt (removed), doc/callingpres.txt (removed), doc/musiconhold-fpm.txt (removed), doc/localchannel.tex (added), doc/enum.txt (removed), doc/cdrdriver.tex (added), build_tools/make_buildopts_h, doc/security.txt (removed), doc/imapstorage.txt (removed), doc/PEERING, main/pbx.c, doc/freetds.tex (added), doc/odbcstorage.tex (added), doc/privacy.txt (removed), configure.ac, doc/iax.txt (removed), doc/channelvariables.tex (added), doc/ael.tex (added), doc/enum.tex (added), doc/security.tex (added), doc/math.txt (removed), Makefile, doc/imapstorage.tex (added), doc/privacy.tex (added), doc/realtime.txt (removed), doc/dundi.txt (removed), doc/mysql.txt (removed), apps/app_voicemail.c, doc/cliprompt.txt (removed), doc/chaniax.txt (removed), doc/app-sms.txt (removed), doc/ast_appdocs.tex (added), doc/realtime.tex (added), doc/ices.txt (removed), doc/dundi.tex (added), doc/queuelog.txt (removed), doc/extconfig.txt (removed), doc/radius.txt (removed), doc/cliprompt.tex (added), doc/chaniax.tex (added), doc/hardware.txt (removed), doc/mp3.txt (removed), doc/app-sms.tex (added), doc/ices.tex (added), doc/asterisk.tex (added), doc/configuration.txt (removed), doc/queuelog.tex (added), doc/asterisk-conf.txt (removed), doc/sla.pdf (removed), doc/ip-tos.txt (removed), doc/hardware.tex (added), doc/h323.txt (removed), doc/mp3.tex (added), doc/configuration.tex (added), doc/asterisk-conf.tex (added), doc/jitterbuffer.txt (removed), doc/channels.txt (removed), doc/ip-tos.tex (added), doc/extensions.txt (removed), doc/queues-with-callback-members.txt (removed), doc/apps.txt (removed), makeopts.in, doc/ajam.txt (removed): Merged revisions 58931 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58931 | russell | 2007-03-15 17:25:12 -0500 (Thu, 15 Mar 2007) | 13 lines Merge changes from svn/asterisk/team/russell/LaTeX_docs. * Convert most of the doc directory into a single LaTeX formatted document so that we can generate a PDF, HTML, or other formats from this information. * Add a CLI command to dump the application documentation into LaTeX format which will only be include if the configure script is run with --enable-dev-mode. * The PDF turned out to be close to 1 MB, so it is not included. However, you can simply run "make asterisk.pdf" to generate it yourself. We may include it in release tarballs or have automatically generated ones on the web site, but that has yet to be decided. ........ 2007-03-15 18:21 +0000 [r58924] Joshua Colp * /, channels/chan_iax2.c: Merged revisions 58923 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58923 | file | 2007-03-15 15:13:21 -0300 (Thu, 15 Mar 2007) | 2 lines Don't assume that the pvt structure will still exist after calling schedule_delivery as it may not. (issue #9278 reported by fmachado) ........ 2007-03-14 19:19 +0000 [r58904-58907] Russell Bryant * /, channels/chan_sip.c: Merged revisions 58906 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58906 | russell | 2007-03-14 14:18:08 -0500 (Wed, 14 Mar 2007) | 4 lines Some people like to put "limitonpeer" instead of "limitonpeers" in their configuration. While we're at it, support "limitonpeerz" and "limitonpeerssssss". (inspired by issue #9172) ........ * /, doc/sla.tex, doc/sla.pdf: Merged revisions 58902 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58902 | russell | 2007-03-14 12:04:38 -0500 (Wed, 14 Mar 2007) | 2 lines Add a more basic example setup to the examples section ........ 2007-03-14 17:01 +0000 [r58900-58901] Olle Johansson * cdr/cdr_radius.c: Correct reference to Radius library THanks Philippe - Greetings from Lisboa, Portugal * /, channels/chan_sip.c: Merged revisions 58848 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r58848 | oej | 2007-03-13 12:49:35 +0100 (Tue, 13 Mar 2007) | 10 lines Merged revisions 58847 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r58847 | oej | 2007-03-13 12:45:52 +0100 (Tue, 13 Mar 2007) | 2 lines Issue #9229 - No port in request URI on register to non default SIP ports (neelakantan) ........ ................ 2007-03-14 16:40 +0000 [r58895-58898] Russell Bryant * /, doc/security.txt: Merged revisions 58897 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r58897 | russell | 2007-03-14 11:40:22 -0500 (Wed, 14 Mar 2007) | 11 lines Merged revisions 58896 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r58896 | russell | 2007-03-14 11:38:48 -0500 (Wed, 14 Mar 2007) | 3 lines Add a note to the security file that the Asterisk CLI and log files may contain sensitive information, and that people should keep this in mind. ........ ................ * /, apps/app_meetme.c, configs/sla.conf.sample: Merged revisions 58894 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58894 | russell | 2007-03-14 11:33:01 -0500 (Wed, 14 Mar 2007) | 8 lines By default, don't attempt to do any CallerID handling at all with SLA because it is known to not work properly in some situations. However, add an option to enable it for those that would like to use it anyway. The short story behind this is that to properly handle CallerID with SLA, we need the ability to change the CallerID on an existing call, and we are not ready to handle that. ........ 2007-03-14 01:56 +0000 [r58881] Tilghman Lesher * /, funcs/func_strings.c: Merged revisions 58880 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58880 | tilghman | 2007-03-13 20:47:08 -0500 (Tue, 13 Mar 2007) | 3 lines Issue 9162 - pbx_substitute_variables_helper assumes the buffer is initialized to all zeroes. This fixes a case where it wasn't. ........ 2007-03-13 23:20 +0000 [r58866-58873] Russell Bryant * /, apps/app_meetme.c: Merged revisions 58872 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58872 | russell | 2007-03-13 18:19:51 -0500 (Tue, 13 Mar 2007) | 4 lines Ensure that the blinky lights show that the trunk stopped ringing when the trunk hangs up before a station has answered it. (issue #9234, reported by francesco_r) ........ * /, configs/sla.conf.sample: Merged revisions 58870 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58870 | russell | 2007-03-13 18:11:08 -0500 (Tue, 13 Mar 2007) | 1 line fix the reference to the SLA documentation ........ * cdr/cdr_sqlite3_custom.c (added), build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, configs/cdr_sqlite3_custom.conf (added), doc/res_config_sqlite.txt (added), cdr/cdr_sqlite.c, configs/extconfig.conf.sample, configure.ac, UPGRADE.txt, CHANGES, makeopts.in, res/res_config_sqlite.c (added), configs/res_config_sqlite.conf (added): Merge changes from team/russell/sqlite: * Add new module, cdr_sqlite3_custom which allows logging custom CDRs into a SQLite3 database. (issue #7149, alerios) * Add new module, res_config_sqlite, which adds realtime database configuration support for SQLite version 2. I decided that this was ok since we didn't have any realtime support for version 3. If someone ports this to version 3, then version 2 support can be removed or marked deprecated. (issue #7790, rbarun_proformatique) * Mark cdr_sqlite as deprecated in favor of cdr_sqlite3_custom. Also, note that there were other modules on the bug tracker that did not make the cut because they provided some duplicated functionality. Those are: * cdr_sqlite3 (issue #6754, moy) * cdr_sqlite3 (issue #8694, bsd) 2007-03-13 10:14 +0000 [r58822-58846] Olle Johansson * /, channels/chan_sip.c: Merged revisions 58845 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58845 | oej | 2007-03-13 11:03:03 +0100 (Tue, 13 Mar 2007) | 3 lines Don't hangup the call on OK or errors on MESSAGE and INFO inside of a dialog (like video update requests). ........ * /, channels/chan_sip.c: Merged revisions 58843 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58843 | oej | 2007-03-13 10:12:16 +0100 (Tue, 13 Mar 2007) | 2 lines Issue #9251 - Clear From URI from user attributes (tgrman) ........ * channels/chan_h323.c: Change URL to OpenH323 (thanks, Tzafrir!) 2007-03-12 01:22 +0000 [r58780-58784] Joshua Colp * /, main/rtp.c: Merged revisions 58783 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58783 | file | 2007-03-11 21:21:12 -0400 (Sun, 11 Mar 2007) | 2 lines Allow RFC2833 compensation to compensate for even stupider implementations by queueing up the end frame at the start, not the actual end. (issue #8963 reported by AndrewZ) ........ * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 58779 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58779 | file | 2007-03-11 20:51:16 -0400 (Sun, 11 Mar 2007) | 2 lines Add matchexterniplocally setting which only substitutes your externip/externhost setting if it matches the localnet setting. I know of at least two people who need opposite settings, so I made it an option! (issue #8821 reported by kokoskarokoska) ........ 2007-03-11 21:57 +0000 [r58761] Kevin P. Fleming * main/asterisk.c: grammatical errors are bad, mmmkay? 2007-03-11 16:43 +0000 [r58742] Jason Parker * build_tools/cflags.xml, main/asterisk.c: Add CLI command "marko show birthday" to show "birthday information" for Mark Spencers upcoming 30th birthday. To enable, run `make menuselect` and select the option MARKO_BDAY under Compiler Flags. 2007-03-10 18:15 +0000 [r58639-58706] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 58705 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58705 | russell | 2007-03-10 12:11:11 -0600 (Sat, 10 Mar 2007) | 6 lines Fix a few more places in chan_iax2 where the ast_frame used for receiving a frame was not properly initialized. - Interpolating a frame when the jitterbuffer is in use - decrypting a frame when IAX2 encryption is on - frames in an IAX2 trunk ........ * /, apps/app_meetme.c: Merged revisions 58669 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58669 | russell | 2007-03-09 21:58:27 -0600 (Fri, 09 Mar 2007) | 2 lines Make the compiler happy and initialize a variable. ........ * /, doc/sla.txt (removed), doc/sla.tex (added), doc/sla.pdf (added): Merged revisions 58638 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58638 | russell | 2007-03-09 17:59:10 -0600 (Fri, 09 Mar 2007) | 8 lines Merge some updates to the SLA documentation. I plan to keep working on this to explain all of the expected behavior with call handling, configuration details for specific phones, and other things. However, I got tired of doing it in plain text, so I switched to using LaTeX. I have included the PDF version. I haven't been able to get a nice looking plain text version out of it yet, but I'm not terribly concerned since this is supposed to be more of the manual, while the plain text sample configuration file is the reference. ........ 2007-03-09 21:10 +0000 [r58592-58605] Joshua Colp * /, apps/app_voicemail.c: Merged revisions 58604 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58604 | file | 2007-03-09 16:08:19 -0500 (Fri, 09 Mar 2007) | 2 lines Fix spelling of unavailable in voicemail documentation. (issue #9248 reported by tensai) ........ * /, channels/chan_sip.c: Merged revisions 58584 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r58584 | file | 2007-03-09 15:49:47 -0500 (Fri, 09 Mar 2007) | 10 lines Merged revisions 58579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r58579 | file | 2007-03-09 15:46:43 -0500 (Fri, 09 Mar 2007) | 2 lines If we are unable to lookup the host in a c line we have to abort, otherwise the previous data is gone and we will (potentially) have no data when all is said and done. ........ ................ 2007-03-08 23:21 +0000 [r58511-58541] Russell Bryant * /, apps/app_meetme.c: Merged revisions 58512 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58512 | russell | 2007-03-08 16:15:15 -0600 (Thu, 08 Mar 2007) | 5 lines Hang up the channel that put the call on hold in the event processing thread to avoid a race condition. Also, if the station originated the call that it is putting on hold, don't hang up the trunk if it was the only station on the call and it is hanging up due to hold and not a normal hangup. ........ * channels/chan_zap.c, /: Merged revisions 58510 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58510 | russell | 2007-03-08 16:06:54 -0600 (Thu, 08 Mar 2007) | 3 lines Add a missing break statement so that handling the above event does not incorrectly destroy the channel. (issue #9242, andrew) ........ 2007-03-08 21:34 +0000 [r58480] Tilghman Lesher * /, res/res_odbc.c: Merged revisions 58479 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58479 | tilghman | 2007-03-08 15:33:03 -0600 (Thu, 08 Mar 2007) | 2 lines Fix segfault (Issue 9236) ........ 2007-03-08 20:56 +0000 [r58475] Russell Bryant * /, apps/app_meetme.c: Merged revisions 58474 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58474 | russell | 2007-03-08 14:54:56 -0600 (Thu, 08 Mar 2007) | 3 lines Refactor hold handling a bit so that it does not require keeping the call up when a call is put on hold. ........ 2007-03-08 18:05 +0000 [r58390-58437] Joshua Colp * /, main/rtp.c: Merged revisions 58436 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58436 | file | 2007-03-08 13:01:00 -0500 (Thu, 08 Mar 2007) | 2 lines Make early SDP seeding even smarter! We have to check codecs in the make_compatible function too. (issue #9221 reported by marcelbarbulescu) ........ * /, main/dsp.c: Merged revisions 58389 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r58389 | file | 2007-03-08 11:07:10 -0500 (Thu, 08 Mar 2007) | 10 lines Merged revisions 58388 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r58388 | file | 2007-03-08 11:04:58 -0500 (Thu, 08 Mar 2007) | 2 lines Only print out debug message if the definition that makes the variables shows up was actually defined. (issue #9233 reported by serginuez) ........ ................ 2007-03-08 13:27 +0000 [r58353-58355] Kevin P. Fleming * /, main/http.c: Merged revisions 58354 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58354 | kpfleming | 2007-03-08 08:23:46 -0500 (Thu, 08 Mar 2007) | 2 lines this change was not needed; fclose() handles closing the file descriptor already ........ * /, apps/app_meetme.c, main/http.c: Merged revisions 58351-58352 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58351 | kpfleming | 2007-03-08 08:17:17 -0500 (Thu, 08 Mar 2007) | 2 lines fix two cases where HTTP session file descriptors would not be closed ........ r58352 | kpfleming | 2007-03-08 08:17:42 -0500 (Thu, 08 Mar 2007) | 2 lines fix a compiler warning, and overwriting 'res' value ........ 2007-03-08 01:06 +0000 [r58304-58321] Russell Bryant * channels/chan_zap.c, /, configure, configure.ac: Merged revisions 58320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58320 | russell | 2007-03-07 19:01:46 -0600 (Wed, 07 Mar 2007) | 6 lines If we receive ZT_EVENT_REMOVED, destroy the specified channel. (issue #7256, tzafrir) Also, update the configure script to make sure that we don't try to build chan_zap if the installed version of zaptel does not include ZT_EVENT_REMOVED. ........ * configs/dundi.conf.sample, pbx/pbx_dundi.c, CHANGES: Add the ability to dynamically specify weights for responses to DUNDi queries. This can be done using a global variable or a dialplan function. Using the SHELL() function will allow you to use an external script to determine what the weight in the response should be. This can be very useful in load balancing applications. (inspired by discussions with blitzrage and jsmith in #asterisk-bugs) 2007-03-07 20:05 +0000 [r58286] Joshua Colp * main/loader.c: Make the loader less noisy under valgrind. 2007-03-07 18:20 +0000 [r58244] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 58243 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r58243 | russell | 2007-03-07 12:19:19 -0600 (Wed, 07 Mar 2007) | 17 lines (This bug was reported to me by Kinsey Moore) Merged revisions 58242 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r58242 | russell | 2007-03-07 12:17:07 -0600 (Wed, 07 Mar 2007) | 7 lines Fix a problem where the Asterisk channel name could be that of the wrong IAX2 user for a call. This is because the first step of choosing this name is to look for an IAX2 peer that happens to have the same IP/port number that this call is coming from and assuming that is it. However, this is not always correct. So, I have made it change this name after authentication happens since at that point, we have an exact match. ........ ................ 2007-03-07 17:55 +0000 [r58241] Joshua Colp * /, channels/chan_sip.c, main/rtp.c: Merged revisions 58240 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58240 | file | 2007-03-07 12:52:58 -0500 (Wed, 07 Mar 2007) | 2 lines Ensure we have (or should have) at least one matching codec before attempting early bridge SDP seeding. (issue #9221 reported by marcelbarbulescu) ........ 2007-03-07 08:08 +0000 [r58224] Olle Johansson * apps/app_ices.c: Adding reference to ices home page. Anyone that has tested with ices2 ? 2007-03-07 01:07 +0000 [r58123-58208] Russell Bryant * main/file.c: Add the format of the file that is currently being played to the verbose message. (issue #9105, junky) * main/manager.c, /: Merged revisions 58165 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r58165 | russell | 2007-03-06 18:25:19 -0600 (Tue, 06 Mar 2007) | 12 lines Merged revisions 58164 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r58164 | russell | 2007-03-06 18:20:13 -0600 (Tue, 06 Mar 2007) | 4 lines If the channels acquired using the manager Redirect action are not up, then don't attempt to do anything with them. It could lead to weird behavior, including crashes. (issue #8977) ........ ................ * include/asterisk/utils.h: Add some documentation on the arguments to the base64 encode/decode functions. (inspired by issue #9215) * apps/app_queue.c: Send a manager AgentComplete event when the agent transfers the call, in addition to where it is already sent if either side hangs up. (issue #9219, rgollent) In passing, I put this code in a function so it would not be duplicated a third time. 2007-03-06 23:19 +0000 [r58122] Steve Murphy * /, channels/chan_sip.c: Merged revisions 58121 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r58121 | murf | 2007-03-06 16:10:14 -0700 (Tue, 06 Mar 2007) | 9 lines Merged revisions 58115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r58115 | murf | 2007-03-06 15:52:52 -0700 (Tue, 06 Mar 2007) | 1 line Fix for 9220: Eyebeam cannot renew subscriptions for presence info. Reason: re-SUBSCRIBE requests don't include Accept headers, which the rfc says are optional (to put it tersely), (it uses MAY), and luckily, the sip_pvt struct has the format info stored, so we simply leave it if the format is set, and the accept header null. ........ ................ 2007-03-06 23:01 +0000 [r58101-58120] Russell Bryant * /, configs/voicemail.conf.sample: Merged revisions 58119 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58119 | russell | 2007-03-06 17:00:57 -0600 (Tue, 06 Mar 2007) | 3 lines Clarify the documentation of the dialout and sendvoicemail options. (issue #9000, caio1982 and serge-v) ........ * codecs/codec_zap.c: Sync codec_zap with the one that is in the 1.4 branch so that it can actually build here, too. 2007-03-06 20:45 +0000 [r58054-58055] Olle Johansson * /, channels/chan_sip.c: Merged revisions 58053 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r58053 | oej | 2007-03-06 21:37:07 +0100 (Tue, 06 Mar 2007) | 10 lines Merged revisions 58052 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r58052 | oej | 2007-03-06 21:33:21 +0100 (Tue, 06 Mar 2007) | 2 lines Change error message to proper message ........ ................ * apps/app_stack.c: Debug control, debug control. 2007-03-06 18:02 +0000 [r58024-58025] Russell Bryant * /, channels/chan_skinny.c: Merged revisions 58023 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58023 | russell | 2007-03-06 12:01:20 -0600 (Tue, 06 Mar 2007) | 3 lines Return an error of transmit_response is called without a session. (issue #9002) ........ 2007-03-06 08:51 +0000 [r57979-57993] Luigi Rizzo * main/say.c: move declaration to the beginning of a block * apps/app_meetme.c: remove duplicate const 2007-03-05 20:13 +0000 [r57871-57943] Joshua Colp * channels/chan_zap.c, CHANGES: Add zap show version CLI command. This pulls the version/echo canceller in use directly using the ZT_GETVERSION ioctl. (issue #9094 reported by tootai) * /, channels/chan_iax2.c: Merged revisions 57914 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57914 | file | 2007-03-05 14:19:07 -0500 (Mon, 05 Mar 2007) | 2 lines Since chan_iax2 does not support reception of DTMF with duration ensure that it is set to 0 on the frame. (issue #8521 reported by gdhgdh) ........ * /, apps/app_meetme.c: Merged revisions 57872 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57872 | file | 2007-03-05 13:39:28 -0500 (Mon, 05 Mar 2007) | 2 lines Don't create a listen channel and record the conference unless the option is turned on. (issue #9204 reported by francesco_r) ........ * apps/app_meetme.c: I like it when app_meetme builds under dev mode, don't you? * /, apps/app_voicemail.c: Merged revisions 57870 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r57870 | file | 2007-03-05 12:52:03 -0500 (Mon, 05 Mar 2007) | 10 lines Merged revisions 57869 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r57869 | file | 2007-03-05 12:49:18 -0500 (Mon, 05 Mar 2007) | 2 lines Make create_dirpath use our standard for return values. -1 is failure, 0 is success. (issue #9205 reported by ballares) ........ ................ 2007-03-05 15:30 +0000 [r57827] Steve Murphy * main/pbx.c, /: Merged revisions 57826 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r57826 | murf | 2007-03-05 08:20:17 -0700 (Mon, 05 Mar 2007) | 9 lines Merged revisions 57825 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r57825 | murf | 2007-03-05 07:53:57 -0700 (Mon, 05 Mar 2007) | 1 line Fixed a typo introduced via 9156 (either the gotos or their doc strings are wrong) ........ ................ 2007-03-05 04:21 +0000 [r57769-57799] Joshua Colp * /, main/slinfactory.c: Merged revisions 57798 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57798 | file | 2007-03-04 23:19:53 -0500 (Sun, 04 Mar 2007) | 2 lines Don't allow a NULL pointer to reach ast_frdup. (issue #9155 reported by cmaj) ........ * configs/extensions.conf.sample: Remove no longer present CLI commands from sample extensions.conf. (issue #9193 reported by junky) * /, res/res_jabber.c: Merged revisions 57770 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57770 | file | 2007-03-04 22:35:03 -0500 (Sun, 04 Mar 2007) | 2 lines Don't reference a potentially NULL pointer. (issue #9199 reported by klolik) ........ * /, main/rtp.c: Merged revisions 57768 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57768 | file | 2007-03-04 22:22:17 -0500 (Sun, 04 Mar 2007) | 2 lines Preserve marker bit when P2P bridging. (issue #9198 reported by edgreenberg) ........ 2007-03-03 16:43 +0000 [r57736] Tilghman Lesher * apps/app_stack.c: Convert stack apps to use ast_storage channel structure 2007-03-03 15:35 +0000 [r57708] Steve Murphy * pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test6, pbx/ael/ael-test/ref.ael-test7, pbx/ael/ael-test/ref.ael-vtest13: updated the regression tests 2007-03-03 14:40 +0000 [r57651-57691] Tilghman Lesher * main/channel.c, include/asterisk/channel.h: Expand datastores to add the notion of inheritance. This will be needed for the conversion of IAX2 variables from the current custom method to ast_storage. * /, apps/app_voicemail.c: Merged revisions 57649 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r57649 | tilghman | 2007-03-03 00:45:00 -0600 (Sat, 03 Mar 2007) | 10 lines Merged revisions 57648 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r57648 | tilghman | 2007-03-03 00:36:55 -0600 (Sat, 03 Mar 2007) | 2 lines Memory leak of a list, if call recording was abandoned ........ ................ 2007-03-03 01:11 +0000 [r57621] Dwayne M. Hubbard * main/say.c: Merged revisions 57620 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57620 | dhubbard | 2007-03-02 18:59:24 -0600 (Fri, 02 Mar 2007) | 1 line submitted patch for Georgian language, issue 9010, submitted by Alexander Shaduri ........ 2007-03-03 00:01 +0000 [r57557-57590] Russell Bryant * configs/sla.conf.sample: Add the missing configuration template to the sample config file. Thanks to Lacy Moore on the asterisk-users list for pointing out that this was missing! * /, configure, configure.ac: Merged revisions 57556 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57556 | russell | 2007-03-02 17:03:01 -0600 (Fri, 02 Mar 2007) | 3 lines Update the check that is used to determine whether zaptel transcoder support is present. The interface has changed. ........ 2007-03-02 18:05 +0000 [r57478-57519] Joshua Colp * main/pbx.c: Don't try to do recursive locking/unlocking when it isn't supported. * /, channels/chan_sip.c: Merged revisions 57477 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r57477 | file | 2007-03-02 12:06:52 -0500 (Fri, 02 Mar 2007) | 10 lines Merged revisions 57475 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r57475 | file | 2007-03-02 12:02:46 -0500 (Fri, 02 Mar 2007) | 2 lines If a SIP message comes in and goes to a method handler that requires additional values that may not be present then send back an error. ........ ................ 2007-03-02 17:03 +0000 [r57476] Steve Murphy * main/pbx.c, /: Merged revisions 57473 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r57473 | murf | 2007-03-02 09:55:16 -0700 (Fri, 02 Mar 2007) | 9 lines Merged revisions 57458 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r57458 | murf | 2007-03-02 09:39:33 -0700 (Fri, 02 Mar 2007) | 1 line further refinement in wording of goto documentation, as per 9156, goto not proceeding to next instruction ........ ................ 2007-03-02 16:59 +0000 [r57474] Russell Bryant * apps/app_dumpchan.c, main/cli.c: Add the channel's Language to the "show channel" CLI command and the DumpChan application. (issue #9187, Junky) 2007-03-02 05:57 +0000 [r57438] Steve Murphy * /, pbx/pbx_ael.c, utils/ael_main.c: Merged revisions 57426 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57426 | murf | 2007-03-01 22:21:36 -0700 (Thu, 01 Mar 2007) | 1 line I almost had comma escapes right, but 9184 points out the problem-- the escape is removed by pbx_config, and pbx_ael should also, before sending it down into the pbx engine. Also, you have to insert it back in, if you are generating extensions.conf code from the AEL. ........ 2007-03-02 00:22 +0000 [r57365-57397] Russell Bryant * /, main/file.c: Merged revisions 57396 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57396 | russell | 2007-03-01 18:20:44 -0600 (Thu, 01 Mar 2007) | 4 lines Return the correct digit that interrupted the stream. This fixes exiting the Background application when using the m option. (issue #9176, mjagdis) ........ * /, apps/app_meetme.c, doc/sla.txt, include/asterisk/channel.h, configs/sla.conf.sample: Merged revisions 57364 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57364 | russell | 2007-03-01 17:42:53 -0600 (Thu, 01 Mar 2007) | 16 lines Merge changes from svn/asterisk/team/russell/sla_updates * Originally, I put in the documentation that only Zap interfaces would be supported on the trunk side. However, after a discussion with Qwell, we came up with a way to make IP trunks work as well, using some things already in Asterisk. So, here it is, this now officially supports IP trunks. * Update the SLA documentation to reflect how to setup IP trunks. * Add a section in sla.txt that describes how to set up an SLA system with voicemail. * Simplify the way DTMF passthrough is handled in MeetMe. * Fix a bug that exposed itself when using a Local channel on the trunk side in SLA. The station's channel needs to be passed to the dial API when dialing the trunk. * Change a WARNING message to DEBUG in channel.h. This message is of no use to users. ........ 2007-03-01 22:23 +0000 [r57319] Joshua Colp * channels/chan_local.c, /: Merged revisions 57318 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r57318 | file | 2007-03-01 17:21:44 -0500 (Thu, 01 Mar 2007) | 10 lines Merged revisions 57317 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r57317 | file | 2007-03-01 17:19:32 -0500 (Thu, 01 Mar 2007) | 2 lines Don't even attempt to optimize things when a proxy channel is involved. It will just explode in weird and unexplaineable ways. (issue #9175 reported by clegall_proformatique) ........ ................ 2007-03-01 20:24 +0000 [r57293] Russell Bryant * main/channel.c: Constify the list of codec preferences. 2007-03-01 03:01 +0000 [r57259] TransNexus OSP Development * doc/osp.txt: 1. Corrected a typo for www.etsi.org. Thank Patrick. 2007-03-01 00:08 +0000 [r57241] Joshua Colp * main/pbx.c: Minor code cleanup... nothing to write home about. 2007-02-28 23:02 +0000 [r57204-57209] Russell Bryant * /, doc/sla.txt, configs/sla.conf.sample: Merged revisions 57207 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57207 | russell | 2007-02-28 17:01:52 -0600 (Wed, 28 Feb 2007) | 2 lines minor tweaks to the sla docs ........ * /, apps/app_meetme.c, configs/sla.conf.sample: Merged revisions 57203 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57203 | russell | 2007-02-28 16:07:05 -0600 (Wed, 28 Feb 2007) | 7 lines Merge more changes from svn/asterisk/team/russell/sla_updates * Add support for private hold. By setting "hold=private" for a trunk, only the station that put the call on hold will be able to retrieve it from hold. Also, by setting "hold=private" for a station, any call that station puts on hold can only be retrieved by that station. ........ 2007-02-28 20:46 +0000 [r57184] Joshua Colp * main/pbx.c, pbx/pbx_dundi.c, include/asterisk/pbx.h, pbx/pbx_config.c, apps/app_while.c: Convert the PBX core to use read/write locks. This yields a nifty performance improvement when it comes to simultaneous calls going through the dialplan. Using murf's test the old mutex based core took an average of 57.3 seconds while the rwlock based core took 31.1 seconds. That's a nifty 26.2 seconds performance improvement. The other good part is that if we do need to switch back then we just have to change the lock/unlock API calls. I converted everywhere that used to touch the mutex locks directly to use them. 2007-02-28 19:59 +0000 [r57145-57147] Russell Bryant * /, apps/app_meetme.c: Merged revisions 57146 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57146 | russell | 2007-02-28 13:58:56 -0600 (Wed, 28 Feb 2007) | 2 lines Minor formatting change ........ * /, apps/app_meetme.c, configs/sla.conf.sample: Merged revisions 57144 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57144 | russell | 2007-02-28 13:56:20 -0600 (Wed, 28 Feb 2007) | 6 lines Merge changes from svn/asterisk/team/russell/sla_updates * Add support for the "barge=no" option for trunks. If this option is set, then stations will not be able to join in on a call that is on progress on this trunk. ........ 2007-02-28 19:30 +0000 [r57140] Steve Murphy * main/pbx.c, /: Merged revisions 57139 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r57139 | murf | 2007-02-28 12:23:05 -0700 (Wed, 28 Feb 2007) | 9 lines Merged revisions 57118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r57118 | murf | 2007-02-28 12:12:41 -0700 (Wed, 28 Feb 2007) | 1 line a small documentation update, to reflect reality in the goto doc strings, as per 9156, Goto does not proceed to next prio if jump fails ........ ................ 2007-02-28 19:00 +0000 [r57094] Joshua Colp * /, channels/chan_agent.c: Merged revisions 57093 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r57093 | file | 2007-02-28 13:57:52 -0500 (Wed, 28 Feb 2007) | 10 lines Merged revisions 57092 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r57092 | file | 2007-02-28 13:55:45 -0500 (Wed, 28 Feb 2007) | 2 lines Fix a few more issues with the agent logoff CLI command. (issue #9123 reported by arbrandes) ........ ................ 2007-02-28 18:21 +0000 [r57090] Russell Bryant * /, apps/app_meetme.c, configs/sla.conf.sample: Merged revisions 57089 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57089 | russell | 2007-02-28 12:20:05 -0600 (Wed, 28 Feb 2007) | 8 lines Merge current set of changes from svn/asterisk/team/russell/sla_updates * Add support for station ring delays. Ring delays can be set globally for a station or for specific trunks on the station. * Fix a few bugs in existing code. * Restructure and Reorganize code to improve readability and maintainability. * Improve formatting of the "sla show (trunks|stations)" CLI commands. ........ 2007-02-28 17:56 +0000 [r57054-57056] Joshua Colp * /: Merged revisions 57055 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57055 | file | 2007-02-28 12:55:03 -0500 (Wed, 28 Feb 2007) | 2 lines Picky compiler... ........ * /, apps/app_speech_utils.c: Merged revisions 57053 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57053 | file | 2007-02-28 12:45:50 -0500 (Wed, 28 Feb 2007) | 2 lines Better handle timeouts when the individual speaks after everything has been played but before the timeout ends. ........ 2007-02-28 17:22 +0000 [r57050] Steve Murphy * /, pbx/pbx_ael.c: Merged revisions 57049 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57049 | murf | 2007-02-28 10:15:27 -0700 (Wed, 28 Feb 2007) | 1 line I was surprised that I had not yet downgraded missing goto targets and macro call defs to a warning, in case they are in extensions.conf; I rectified this problem. Also, A goto in a macro to a target in a catch block was not being found; I fixed this too; the cause was that I needed to treat catch statements like an extension in the find_match code. ........ 2007-02-27 22:17 +0000 [r57011] Joshua Colp * apps/app_dial.c: Properly hangup the original dialed channel, not the new channel that appeared from the forwarding. (issue #9161 reported by PhilSmith) 2007-02-27 17:38 +0000 [r56976] Russell Bryant * /: (also issue #9159) Merged revisions 56975 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r56975 | russell | 2007-02-27 11:36:09 -0600 (Tue, 27 Feb 2007) | 4 lines Fix voicemail email attachments. I missed the conversion of one of the line endings and there was an extra one where it should not have been. (issue #9128) ........ 2007-02-27 00:11 +0000 [r56926-56952] Tilghman Lesher * channels/chan_zap.c, configs/zapata.conf.sample: Issue 7789 - some telcos want the TON set based on the number, but without the NANP prefix removed 2007-02-26 20:43 +0000 [r56889] Russell Bryant * /, channels/chan_alsa.c: Merged revisions 56888 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r56888 | russell | 2007-02-26 14:42:21 -0600 (Mon, 26 Feb 2007) | 4 lines Restore the behavior of Asterisk 1.2 where if a device was not specified in alsa.conf, then we just use the system default, instead of creating our own default of hw:0,0. (issue #9139) ........ 2007-02-26 20:09 +0000 [r56860] Joshua Colp * /, pbx/pbx_config.c: Merged revisions 56856 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r56856 | file | 2007-02-26 15:07:18 -0500 (Mon, 26 Feb 2007) | 10 lines Merged revisions 56850 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r56850 | file | 2007-02-26 15:05:02 -0500 (Mon, 26 Feb 2007) | 2 lines Obey the clearglobalvars option in extensions reload (or dialplan reload depending on your version). (issue #9146 reported by ramonpeek) ........ ................ 2007-02-26 20:04 +0000 [r56849] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 56847 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r56847 | russell | 2007-02-26 14:04:13 -0600 (Mon, 26 Feb 2007) | 2 lines Fix a crash in my last change to iax2_indicate(). (issue #9150) ........ 2007-02-26 19:34 +0000 [r56811-56840] Joshua Colp * /, apps/app_record.c: Merged revisions 56839 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r56839 | file | 2007-02-26 14:33:48 -0500 (Mon, 26 Feb 2007) | 2 lines Update app_record documentation to use new CLI command, core show file formats. (issue #9151 reported by junky) ........ * main/pbx.c, /: Merged revisions 56805 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r56805 | file | 2007-02-26 12:09:53 -0500 (Mon, 26 Feb 2007) | 2 lines Use ast_strlen_zero to see if the language and/or context argument is not present for Background instead of just checking if it is NULL. (issue #9141 reported by mjagdis) ........ 2007-02-26 16:54 +0000 [r56786] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 56785 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r56785 | russell | 2007-02-26 10:51:18 -0600 (Mon, 26 Feb 2007) | 3 lines Do more complete locking of the chan_iax2_pvt struct in the indicate callback. (Problem brought up by Ben Smithurst on the asterisk-dev list) ........ 2007-02-26 16:38 +0000 [r56784] Joshua Colp * /, main/asterisk.c: Merged revisions 56783 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r56783 | file | 2007-02-26 11:36:08 -0500 (Mon, 26 Feb 2007) | 2 lines Allow both of the show version files and core show file versions CLI commands to work. (issue #9135 reported by mvanbaak) ........ 2007-02-26 01:05 +0000 [r56731-56742] Russell Bryant * /, apps/app_meetme.c: Merged revisions 56740 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r56740 | russell | 2007-02-25 19:04:40 -0600 (Sun, 25 Feb 2007) | 2 lines Move a comment to be in the correct struct. ........ * main/asterisk.c: Remove redundant check to ensure that LOW_MEMORY is not defined. (issue #9136, mvanbaak) * channels/chan_iax2.c: There is no need to look in the iaxs array for the pvt struct when we already have a pointer to it. 2007-02-25 14:53 +0000 [r56686] Tilghman Lesher * main/channel.c, /: Merged revisions 56685 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r56685 | tilghman | 2007-02-25 08:46:41 -0600 (Sun, 25 Feb 2007) | 11 lines Merged revisions 56684 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r56684 | tilghman | 2007-02-25 08:38:03 -0600 (Sun, 25 Feb 2007) | 3 lines Issue 9130 - If prev is the last item on the channel list, then evaluating additional conditions (e.g. name prefix) will cause a NULL dereference. ........ ................ 2007-02-24 20:29 +0000 [r56623-56665] Olle Johansson * include/asterisk/http.h, main/channel.c, include/asterisk/doxyref.h, include/asterisk/utils.h, include/asterisk/zapata.h, apps/app_meetme.c, res/res_limit.c, include/asterisk/config.h, channels/chan_h323.c, pbx/pbx_ael.c, apps/app_amd.c, include/asterisk/ael_structs.h, include/asterisk/jingle.h, main/config.c, main/rtp.c: Doxygen additions, corrections * include/asterisk/doxyref.h, channels/chan_zap.c, main/manager.c, include/asterisk/frame.h: Doxygen updates and corrections * apps/app_osplookup.c, funcs/func_curl.c, res/res_snmp.c, apps/app_festival.c, cdr/cdr_sqlite.c, codecs/codec_speex.c, contrib/asterisk-ng-doxygen, include/asterisk/jabber.h, res/res_crypto.c, channels/chan_h323.c, cdr/cdr_pgsql.c, cdr/cdr_radius.c, apps/app_voicemail.c: Creating new doxygen macro "\extref" to create page that lists external libraries and URLs to these. Please help me add these references. We might want to create a similar macro "\linuxpackage" to list the needed Linux packages in popular distributions. * include/asterisk/jabber.h: Add some external references * include/asterisk/doxyref.h, include/asterisk/jabber.h: Doxygen updates for AJI - The Asterisk Jabber API 2007-02-24 02:23 +0000 [r56574-56594] Jason Parker * channels/chan_skinny.c, configs/skinny.conf.sample: Allow a Skinny device to monitor a dialplan hint (w00t!). See skinny.conf.sample for configuration example. Note: Some devices (seen on 12SP+/30VIP) will lock up if they monitor too many hints. This seems to be a hardware limitation - there isn't anything we can do about it. * channels/chan_skinny.c: Support devicestate requests. Now you should be able to subscribe to a Skinny device/line. * /, channels/chan_skinny.c: Merged revisions 56569 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r56569 | qwell | 2007-02-23 20:02:53 -0600 (Fri, 23 Feb 2007) | 4 lines Make sure to set a speeddials parent on creation. Don't crash if hold is pressed when no call is active. Don't return in places that we shouldn't.. Update softkey map when call is connected ........ 2007-02-24 01:56 +0000 [r56564] Joshua Colp * apps/app_meetme.c: Make Meetme build again under dev mode. 2007-02-23 23:25 +0000 [r56487-56506] Russell Bryant * /, main/asterisk.c: Merged revisions 56505 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r56505 | russell | 2007-02-23 17:24:18 -0600 (Fri, 23 Feb 2007) | 16 lines Merged revisions 56504 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r56504 | russell | 2007-02-23 17:20:55 -0600 (Fri, 23 Feb 2007) | 8 lines Fix up a couple more signal handlers to not do bad things that could cause various undesirable results. The other day, I made Asterisk deadlock by hitting Control-C because of a bad signal handler. Now, signal handlers just set a flag and write to an alert pipe for the flag to be handled. Then, there is another thread that is monitoring for these flags. If being run in console mode, it is just the main thread. If Asterisk is in the background, a thread is created to do it. ........ ................ * channels/chan_iax2.c: Make the hashing function calculate something that makes more sense. (Thanks to bmd on #asterisk-dev for pointing out my pointless math). 2007-02-23 21:57 +0000 [r56458] Joshua Colp * /, main/sched.c: Merged revisions 56457 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r56457 | file | 2007-02-23 16:53:41 -0500 (Fri, 23 Feb 2007) | 2 lines Change log notice to debug. It is possible for a scheduled item to execute and be deleted at close to the same time and unavoidable. If this happens this message creeps up. ........ 2007-02-23 21:20 +0000 [r56408-56447] Russell Bryant * channels/chan_iax2.c: Merge team/russell/iax2_performance. There is not a large amount of code here and the changes are not very invasive. However, they should significantly improve performance of chan_iax2 under load. IAX2 media frames only carry the *source* call number. So, when one arrives, the correct session that it is a part of has to be matched on IP address, port number, and call number, instead of just a call number. Had these frames carried the *destination* call number, this would not be an issue, because that would be a unique identifier that would make it easy to immediately identify the correct session. The way that chan_iax2 did this matching was extremely inefficient. It starts at the first available call number and traverses each call number sequentially, locking and unlocking a mutex for each one, to try to match against it. It would do this regardless of whether the call number was in use or not. So, for a call with a local call number of 25000, every single incoming media frame would require a traversal that required 25000 mutex lock and unlock operations. (Note that the max call number is about 32k). I have introduced a hash table of active IAX2 calls to improve this lookup process. The hash is done on the IP address, port number, and call number. So, for the previous example, a few lock/unlock operations may be done versus 25000 for each frame. * CHANGES: Note that the entries in the CHANGES file only list functionality changes * CHANGES: Add GetConfigJSON to the CHANGES file. * /, channels/chan_iax2.c: Merged revisions 56407 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r56407 | russell | 2007-02-23 14:20:00 -0600 (Fri, 23 Feb 2007) | 12 lines Merged revisions 56406 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r56406 | russell | 2007-02-23 14:17:56 -0600 (Fri, 23 Feb 2007) | 4 lines Don't destroy mutexes before unregistering all of the entry points from the core. Also, fix a potential memory leak from not destroying the locks for all of the possible call numbers (about 32k of them). ........ ................ 2007-02-23 19:00 +0000 [r56373] Kevin P. Fleming * /, build_tools/make_version_h: Merged revisions 56372 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r56372 | kpfleming | 2007-02-23 12:59:09 -0600 (Fri, 23 Feb 2007) | 2 lines build special version strings for AADK/S800i builds ........ 2007-02-23 18:01 +0000 [r56278-56342] Russell Bryant * /, apps/app_voicemail.c: Merged revisions 56341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r56341 | russell | 2007-02-23 11:58:57 -0600 (Fri, 23 Feb 2007) | 8 lines The IMAP storage code uses the same code to build the email that is used when voicemail is sent via email using something like sendmail. In the patch from bug 8033 to fix various IMAP storage problems, the line endings in the email file were changed in the code from "\n" to "\r\n". However, this breaks sending regular voicemail to email. So, this change conditionally sets line endings to "\r\n" only if IMAP_STORAGE is enabled. (issue #9128, patch by jarjarbinks, modified by me to not break IMAP storage) ........ * main/manager.c: Introduce a new manager action, GetConfigJSON, which is intended to improve performance of the GUI. This encodes the configuration into the JSON format in a manager header, "JSON: ". The encoded information can be directly used as a javascript object, so no parsing is needed. For large configuration files, this can greatly improve loading times in the GUI. Furthermore, the encoding takes up a lot less space when being transmitted than the other alternatives. (Inspired by discussion with Pari) Here is an example of what you get: http://localhost:8088/asterisk/rawman?action=getconfigjson&filename=users.conf Response: Success JSON: {"general":["hasvoicemail=yes"],"6000":["fullname=russell","secret=1234"]} * main/dial.c, /, apps/app_meetme.c, doc/sla.txt, configs/sla.conf.sample: Merged revisions 56277 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r56277 | russell | 2007-02-22 17:08:36 -0600 (Thu, 22 Feb 2007) | 18 lines Merge changes from team/russell/sla_updates. This batch of changes to the SLA code does a few different things. * I made the SLA code event driven instead of having to act in a lot of busy loops while dialing things to wait for state changes. This makes the code more efficient and readable at the same time. * I have implemented a couple of new features. The first is inbound trunk ringing timeouts. This is an option that defines how long to let an incoming call on a trunk to ring. * I have also implemented ring timeouts for stations. They may be specified for the entire station, meaning it is how long to let the station ring before giving up. You can also specify a ring timeout for a specific trunk on a station. So, you can say that you only want a specific station to ring 5 seconds if it is line1 ringing, but otherwise, there is no timeout. ........ 2007-02-22 18:53 +0000 [r56232] Joshua Colp * main/channel.c, /, channels/chan_sip.c: Merged revisions 56231 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r56231 | file | 2007-02-22 13:49:39 -0500 (Thu, 22 Feb 2007) | 10 lines Merged revisions 56230 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r56230 | file | 2007-02-22 13:44:24 -0500 (Thu, 22 Feb 2007) | 2 lines Only change the original or clone channel if it's the channel behind the proxy channel, not if it's just a regular bridged channel. ........ ................ 2007-02-22 17:36 +0000 [r56209] Kevin P. Fleming * include/asterisk/module.h: move the ast_module_info structure into the special section as well, otherwise when restore_globals() is called it will lose its pointer to the ast_module structure that the loader put there 2007-02-22 16:48 +0000 [r56188] Joshua Colp * .cleancount: Since I'm a nice guy... let's increment the clean count since last night's module changes require a rebuild of everything essentially. 2007-02-22 16:25 +0000 [r56187] Russell Bryant * apps/app_voicemail.c: Fix some compilation problems in app_voicemail. There was a parenthesis missing in a function prototype, and "#elifdef" is not a valid preprocessor directive. (issue #9122, akohlsmith) 2007-02-22 13:58 +0000 [r56156] TransNexus OSP Development * doc/osp.txt: Update OSP documention for v1.6. 2007-02-22 10:46 +0000 [r56126] Olle Johansson * /, channels/chan_sip.c: Merged revisions 56125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r56125 | oej | 2007-02-22 11:33:55 +0100 (Thu, 22 Feb 2007) | 2 lines Move message from verbose to debug ........ 2007-02-22 02:48 +0000 [r56095] Steve Murphy * /, sounds/Makefile: Merged revisions 56094 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r56094 | murf | 2007-02-21 19:39:58 -0700 (Wed, 21 Feb 2007) | 1 line updated the sound tarball versions in Makefile ........ 2007-02-22 02:36 +0000 [r56092] Kevin P. Fleming * funcs, codecs, apps, include/asterisk/module.h, Makefile.moddir_rules, Makefile.rules, build_tools/make_linker_eo_script (added), cdr, pbx, res, channels, formats, main/loader.c: give embedded modules a helping hand by backing up and restoring their global variables when they are loaded and unloaded 2007-02-22 01:26 +0000 [r56012-56056] Russell Bryant * /, channels/chan_sip.c: Merged revisions 56055 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r56055 | russell | 2007-02-21 19:24:10 -0600 (Wed, 21 Feb 2007) | 3 lines Restructure a little bit of code to reduce nesting. There is no functionality change here. ........ * /, channels/chan_sip.c: Merged revisions 56011 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r56011 | russell | 2007-02-21 18:57:36 -0600 (Wed, 21 Feb 2007) | 11 lines Merged revisions 56010 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r56010 | russell | 2007-02-21 18:53:25 -0600 (Wed, 21 Feb 2007) | 3 lines If we receive a frame that is not in any of the negotiated formats, then drop it. (potentially issue #8781 and SPD-12) ........ ................ 2007-02-22 00:38 +0000 [r56009] Joshua Colp * /, main/cli.c: Merged revisions 56008 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r56008 | file | 2007-02-21 19:35:55 -0500 (Wed, 21 Feb 2007) | 2 lines Print out deprecation notice on usage output of CLI commands. (issue #8925 reported by blitzrage) ........ 2007-02-22 00:05 +0000 [r55958-56005] Joshua Colp * apps/app_voicemail.c: Make filename on email follow subject message number, purely for cosmetic purposes for individuals like *cough* jsmith *cough*. (issue #9111 reported by sshah) * /, apps/app_meetme.c: Merged revisions 55957 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r55957 | file | 2007-02-21 15:35:40 -0500 (Wed, 21 Feb 2007) | 10 lines Merged revisions 55956 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r55956 | file | 2007-02-21 15:32:16 -0500 (Wed, 21 Feb 2007) | 2 lines Change naughty warning message to provide useful information. If a write now fails on a channel in meetme it will tell you the channel name instead of spitting out the wrong error message. ........ ................ 2007-02-21 20:30 +0000 [r55955] Jason Parker * channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged revisions 55954 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55954 | qwell | 2007-02-21 14:27:08 -0600 (Wed, 21 Feb 2007) | 4 lines Fix locking issue, and accept "transport-accept" as a valid accept message. This should solve issues 8970 and 8503. ........ 2007-02-21 20:26 +0000 [r55953] Joshua Colp * channels/chan_sip.c: Clarify in the doxygen docs abou RFC2833 compensation flag. 2007-02-21 20:23 +0000 [r55952] Russell Bryant * /, apps/app_meetme.c: Merged revisions 55951 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55951 | russell | 2007-02-21 14:22:33 -0600 (Wed, 21 Feb 2007) | 3 lines Simplify the last change to app_meetme, and move the call to dispose_conf() up into the block where we know a conf exists. ........ 2007-02-21 20:18 +0000 [r55915-55950] Joshua Colp * /, apps/app_meetme.c: Merged revisions 55949 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55949 | file | 2007-02-21 15:16:34 -0500 (Wed, 21 Feb 2007) | 2 lines Only dispose of the conference if one was created. ........ * /, apps/app_speech_utils.c: Merged revisions 55947 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55947 | file | 2007-02-21 15:03:38 -0500 (Wed, 21 Feb 2007) | 2 lines Only start playing the next file if we have not been quieted. ........ * /, channels/chan_sip.c: Merged revisions 55914 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55914 | file | 2007-02-21 12:18:19 -0500 (Wed, 21 Feb 2007) | 2 lines Add a flag that indicates whether a SIP dialog is an outgoing call or not. SIP_OUTGOING originally did it but it was repurposed to the direction of the last transaction, which can cause update_call_counter to falsely decrease the wrong counters. (please don't hurt me oej) (issue #8943 reported by mdu113) ........ 2007-02-21 14:07 +0000 [r55870] Kevin P. Fleming * /, build_tools/make_version: Merged revisions 55869 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r55869 | kpfleming | 2007-02-21 08:06:47 -0600 (Wed, 21 Feb 2007) | 10 lines Merged revisions 55868 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r55868 | kpfleming | 2007-02-21 08:03:11 -0600 (Wed, 21 Feb 2007) | 2 lines use new tag version script ........ ................ 2007-02-21 08:39 +0000 [r55835] Olle Johansson * /, channels/chan_sip.c: Merged revisions 55834 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55834 | oej | 2007-02-21 09:32:34 +0100 (Wed, 21 Feb 2007) | 2 lines Issue #8848 - Turn off lamp more quickly after transfer (decrement inuse early on transferer's call leg) ........ 2007-02-21 02:04 +0000 [r55805] Jason Parker * channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged revisions 55799 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55799 | qwell | 2007-02-20 20:01:36 -0600 (Tue, 20 Feb 2007) | 4 lines Fix segfault when buddy couldn't be found. Issue 7764, patch by sailer ........ 2007-02-21 01:05 +0000 [r55763] Joshua Colp * main/dns.c: Return trunk to a state where it compiles under Darwin. The byte order stuff is ugly, if anyone wants to clean it up... by all means do so. 2007-02-21 01:05 +0000 [r55762] Russell Bryant * /, apps/app_meetme.c: Merged revisions 55758 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55758 | russell | 2007-02-20 19:03:25 -0600 (Tue, 20 Feb 2007) | 4 lines Improve the reference counting to fix bugs where people report seeing conferences listed that have no members. (issue #9073) ........ 2007-02-21 00:14 +0000 [r55671-55748] Joshua Colp * /, apps/app_voicemail.c: Merged revisions 55741 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55741 | file | 2007-02-20 19:11:20 -0500 (Tue, 20 Feb 2007) | 2 lines Better handle dropped IMAP connections. (issue #9054 reported by bsmithurst) ........ * /, channels/chan_sip.c: Merged revisions 55717 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55717 | file | 2007-02-20 18:57:03 -0500 (Tue, 20 Feb 2007) | 2 lines Return behavior I removed. I did not remember that you could just add a localnet entry to make it work. ........ * main/logger.c: Flush out the file pointer. (issue #9115 reported by guthrie) * /, channels/chan_sip.c: Merged revisions 55688 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55688 | file | 2007-02-20 18:08:45 -0500 (Tue, 20 Feb 2007) | 2 lines Don't test our own address against the localnet settings. At least one person has had issues as a result of this from #7051 so I'm reversing it. (issue #8821 reported by kokoskarokoska) ........ * /, channels/chan_agent.c: Merged revisions 55670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r55670 | file | 2007-02-20 17:47:00 -0500 (Tue, 20 Feb 2007) | 10 lines Merged revisions 55669 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r55669 | file | 2007-02-20 17:39:14 -0500 (Tue, 20 Feb 2007) | 2 lines Defer clearing callback information if channels are up until they are hung up. This ensures the hangup process goes smoothly and no channels get hung in limbo. (issue #8088 reported by kebl0155) ........ ................ 2007-02-20 20:32 +0000 [r55591-55635] Russell Bryant * /, main/http.c: Merged revisions 55634 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55634 | russell | 2007-02-20 14:26:06 -0600 (Tue, 20 Feb 2007) | 3 lines Add the Asterisk version information to the Server header in HTTP responses. (requested by Pari) ........ * /, include/asterisk/manager.h: Merged revisions 55590 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55590 | russell | 2007-02-20 13:57:07 -0600 (Tue, 20 Feb 2007) | 2 lines Increase the maximum number of manager headers to 128, at the request of Pari. ........ 2007-02-20 16:56 +0000 [r55556] Jason Parker * channels/chan_jingle.c, /, channels/chan_gtalk.c, res/res_jabber.c: Merged revisions 55555 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55555 | qwell | 2007-02-20 10:53:45 -0600 (Tue, 20 Feb 2007) | 4 lines No need to cast nor free with strdupa (thanks file) 55555! ........ 2007-02-20 16:42 +0000 [r55554] Russell Bryant * /, configs/sla.conf.sample: Merged revisions 55553 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55553 | russell | 2007-02-20 10:41:57 -0600 (Tue, 20 Feb 2007) | 3 lines Change the formatting of sla.conf.sample to make it more readable. (issue #9112, blitzrage) ........ 2007-02-20 15:19 +0000 [r55534] Joshua Colp * res/res_jabber.c: I like it when trunk builds, so let's make res_jabber compile again! 2007-02-20 07:48 +0000 [r55514] Olle Johansson * /, res/res_jabber.c: Merged revisions 55483 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55483 | oej | 2007-02-19 22:12:55 +0100 (Mon, 19 Feb 2007) | 3 lines - Not sending arguments to an application is not "out of memory" - Making error messages a bit more clear ........ 2007-02-19 23:27 +0000 [r55495] Jason Parker * .cleancount: We need to bump the cleancount when we make API changes... 2007-02-19 18:15 +0000 [r55436] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 55435 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r55435 | tilghman | 2007-02-19 12:11:48 -0600 (Mon, 19 Feb 2007) | 10 lines Merged revisions 55434 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r55434 | tilghman | 2007-02-19 12:09:09 -0600 (Mon, 19 Feb 2007) | 2 lines forcename and forcegreetings options should check to see if the recording already exists ........ ................ 2007-02-19 16:01 +0000 [r55410-55414] Joshua Colp * CHANGES: Clarify last change for SMDI in CHANGES file. * configs/voicemail.conf.sample, apps/app_voicemail.c: Allow both an external application and SMDI to do voicemail notification at the same time. (issue #8625 reported by lters) 2007-02-19 15:24 +0000 [r55409] Doug Bailey * /, channels/chan_iax2.c: Merged revisions 55397 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55397 | dbailey | 2007-02-19 08:52:59 -0600 (Mon, 19 Feb 2007) | 3 lines Changed iax2 process thread to detached to correct memory leak due to left over thread context on thread exit. Modified module unload process to avoid deadlocks on pthread cancels ........ 2007-02-18 22:07 +0000 [r55375] Olle Johansson * apps/app_voicemail.c: Formatting changes. 2007-02-18 19:13 +0000 [r55351-55352] Joshua Colp * codecs/gsm/inc/proto.h: Return GSM to a state where it actually builds under dev mode. * channels/chan_h323.c: Update chan_h323 to new set_rtp_peer definition. 2007-02-18 15:11 +0000 [r55330] Olle Johansson * res/res_features.c: Being picky... 2007-02-18 15:03 +0000 [r55329] Kevin P. Fleming * Makefile, channels/chan_misdn.c, main/srv.c, main/editline/refresh.c, pbx/ael/ael.tab.c, channels/misdn/isdn_msg_parser.c, channels/chan_oss.c, main/enum.c, apps/app_voicemail.c, main/ast_expr2.c: add -Wundef to the --enable-dev-mode flags, so that mistyped macro names in #if expressions will be caught convert various #if expressions to #ifdef for macros that may not be defined (and where the value is not important) Note: two of these changes are in bison generated files which is going to be inconvenient when they are regenerated 2007-02-18 15:01 +0000 [r55279-55323] Olle Johansson * res/res_features.c: Simplify post_manager_event() * /, apps/app_record.c: Merged revisions 55278 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r55278 | oej | 2007-02-18 13:35:54 +0100 (Sun, 18 Feb 2007) | 10 lines Merged revisions 55277 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r55277 | oej | 2007-02-18 13:32:13 +0100 (Sun, 18 Feb 2007) | 2 lines Documentation update (#9053, jsmith) ........ ................ 2007-02-17 17:41 +0000 [r55220] Joshua Colp * /, apps/app_queue.c: Merged revisions 55219 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55219 | file | 2007-02-17 12:39:32 -0500 (Sat, 17 Feb 2007) | 2 lines Add missing membername option to AddQueueMember documentation. (issue #9088 reported by seanbright) ........ 2007-02-17 17:11 +0000 [r55218] Jason Parker * /, channels/chan_skinny.c: Merged revisions 55217 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55217 | qwell | 2007-02-17 11:10:09 -0600 (Sat, 17 Feb 2007) | 4 lines Fix an issue where callerid would not be displayed on some phones. Issue 8995, initial patch and research done by wedhorn ........ 2007-02-17 16:48 +0000 [r55087-55198] Joshua Colp * apps/app_queue.c: We want to skip the queue if the name doesn't match the specified one, not if they *do*. * apps/app_queue.c: Increase "queue show" buffer size from 80 to 240. This should be more then enough for most cases. (issue #9089 reported by mvanbaak) * apps/app_dial.c, /: Merged revisions 55154 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r55154 | file | 2007-02-16 22:55:30 -0500 (Fri, 16 Feb 2007) | 10 lines Merged revisions 55153 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r55153 | file | 2007-02-16 22:53:45 -0500 (Fri, 16 Feb 2007) | 2 lines Answer the channel before recording privacy information. (issue #8926 reported by lmamane) ........ ................ * /, apps/app_queue.c: Merged revisions 55129 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55129 | file | 2007-02-16 21:59:50 -0500 (Fri, 16 Feb 2007) | 2 lines Make the 'i' option of Queue actually work. (issue #8986 reported by utis) ........ * channels/chan_jingle.c: Update chan_jingle to new definition of set_rtp_peer. * /, channels/chan_sip.c: Merged revisions 55086 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r55086 | file | 2007-02-16 20:16:59 -0500 (Fri, 16 Feb 2007) | 10 lines Merged revisions 55073 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r55073 | file | 2007-02-16 20:09:50 -0500 (Fri, 16 Feb 2007) | 2 lines Allow chan_sip to handle attended transfers from a SIP phone that is sitting behind chan_agent. Yes folks, all it took was one line of code. (issue #8784 reported by pzieba) ........ ................ 2007-02-17 01:11 +0000 [r55004-55077] Russell Bryant * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 55052 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55052 | russell | 2007-02-16 18:40:34 -0600 (Fri, 16 Feb 2007) | 3 lines If the pg_config application is found, but there is probably executing it, then consider postgres unavailable. (issue #8637) ........ * /, codecs/gsm/Makefile: Merged revisions 55050 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55050 | russell | 2007-02-16 18:31:42 -0600 (Fri, 16 Feb 2007) | 3 lines Filter out yet another architecture that does not work with the optimizations in the built-in libgsm. (issue 8637, ovi) ........ * /, apps/app_meetme.c, configs/meetme.conf.sample: Merged revisions 55006 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r55006 | russell | 2007-02-16 16:49:42 -0600 (Fri, 16 Feb 2007) | 17 lines Merged revisions 55005 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r55005 | russell | 2007-02-16 16:48:22 -0600 (Fri, 16 Feb 2007) | 9 lines Revert the change I did in revisions 54955, 54969, and 54970, in 1.2, 1.4, and trunk. I decided that once a conference is created from meetme.conf, it is acceptable behavior that the pin can not be changed until the conference goes away. I also added a note in meetme.conf to describe this behavior. We still have another issue in 1.4 and trunk where some conferences with no users don't go away. That is the real bug that needs to be addressed here. ........ ................ * apps/app_dumpchan.c: Print the raw read/write formats in the DumpChan application. (issue #9083, junky) 2007-02-16 22:20 +0000 [r55003] Joshua Colp * /, channels/chan_agent.c: Merged revisions 55002 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r55002 | file | 2007-02-16 17:18:46 -0500 (Fri, 16 Feb 2007) | 10 lines Merged revisions 54999 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r54999 | file | 2007-02-16 17:13:45 -0500 (Fri, 16 Feb 2007) | 2 lines Do not send indications through ast_indicate in chan_agent but instead go directly to the technology. This way when indications are emulated they happen on the Agent channel and do not screw up formats on the channels. (issue #8439 reported by punkgode) ........ ................ 2007-02-16 21:13 +0000 [r54970] Russell Bryant * /, apps/app_meetme.c: Merged revisions 54969 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r54969 | russell | 2007-02-16 15:12:18 -0600 (Fri, 16 Feb 2007) | 13 lines Merged revisions 54955 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r54955 | russell | 2007-02-16 14:56:58 -0600 (Fri, 16 Feb 2007) | 5 lines For conferences that are configured in meetme.conf, check the configuration file every time someone joins the conference instead of only when the conference is first created. This is to ensure that changes to the pin numbers in the config file are always honored. (issue #9073) ........ ................ 2007-02-16 18:53 +0000 [r54910-54925] Joshua Colp * apps/app_dial.c, /: Merged revisions 54924 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r54924 | file | 2007-02-16 13:51:15 -0500 (Fri, 16 Feb 2007) | 2 lines Need to check macro extension as well as macro context for directed pickup. ........ * res/res_features.c, configs/features.conf.sample: Allow the user to specify where to enable the respective features for when a parked call is picked up. (ie: transfers and parking) 2007-02-16 18:04 +0000 [r54890-54901] Russell Bryant * /, pbx/pbx_config.c: Merged revisions 54898 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r54898 | russell | 2007-02-16 12:03:41 -0600 (Fri, 16 Feb 2007) | 4 lines Fix setting "autofallthrough" to yes by default. It was set to enabled in pbx.c. However, if the option was not present in extensions.conf, then pbx_config.c would set it back to disabled. ........ * /, res/res_features.c: Merged revisions 54888 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r54888 | russell | 2007-02-16 11:40:38 -0600 (Fri, 16 Feb 2007) | 3 lines Clean up a few coding guidelines issues - spaces to tabs, use sizeof() to pass the size of a static buffer, add spaces ... ........ 2007-02-16 17:41 +0000 [r54889] Joshua Colp * res/res_features.c, CHANGES, configs/features.conf.sample: Add option to features.conf that enables parking via DTMF on picked up parked calls. (issue #9082 reported by francesco_r) 2007-02-16 17:26 +0000 [r54887] Jason Parker * /, main/asterisk.c: Merged revisions 54886 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r54886 | qwell | 2007-02-16 11:25:21 -0600 (Fri, 16 Feb 2007) | 4 lines Clarify a restart message. It's silly, but the reporter had a very valid point. Issue 9079 ........ 2007-02-16 17:07 +0000 [r54885] Joshua Colp * apps/app_dial.c, /: Merged revisions 54884 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r54884 | file | 2007-02-16 12:02:35 -0500 (Fri, 16 Feb 2007) | 2 lines Allow directed pickup to pick up the real context instead of the macro context if a Macro is used. (issue #8984 reported by jamesb63) ........ 2007-02-16 14:31 +0000 [r54773-54862] Olle Johansson * channels/chan_sip.c: Formatting, whitespace fixes * apps/app_voicemail.c: More cleanups of app_voicemail * CREDITS, main/channel.c, channels/chan_sip.c, channels/chan_skinny.c, include/asterisk/rtp.h, include/asterisk/channel.h, channels/chan_gtalk.c, CHANGES, include/asterisk/frame.h, main/rtp.c, channels/chan_mgcp.c: Adding Realtime Text support (T.140) to Asterisk T.140/RFC 2793 is a live communication channel, originally created for IP based text phones for hearing impaired. Feels very much like the old Unix talk application. This code is developed and disclaimed by John Martin of Aupix, UK. Tested for interoperability by myself and Omnitor in Sweden, the company that wrote most of the specifications. A big thank you to everyone involved in this. * /, channels/chan_sip.c: Merged revisions 54787 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r54787 | oej | 2007-02-16 13:06:23 +0100 (Fri, 16 Feb 2007) | 2 lines Issue #7541 - Handle multipart attachments to SIP messages - even if boundary is quoted. ........ * res/res_agi.c: Issue #9068 - make sure we quote HTML characters correctly too (seanbright) * /, res/res_agi.c: Merged revisions 54772 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r54772 | oej | 2007-02-16 12:39:55 +0100 (Fri, 16 Feb 2007) | 10 lines Merged revisions 54771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r54771 | oej | 2007-02-16 12:38:03 +0100 (Fri, 16 Feb 2007) | 2 lines Issue #9069 - If we open with TH we should not close with /TD. (seanbright) ........ ................ 2007-02-16 01:36 +0000 [r54711-54749] Joshua Colp * main/acl.c: Rely on ast_gethostbyname to handle IP addresses, not inet_aton. (issue #9056 reported by pj) * CHANGES, apps/app_chanspy.c: Add 'o' option to Chanspy which causes it to only listen to audio coming from the channel, and the 'X' option which allows the user to exit to a valid single digit extension. (issue #8137 reported by mnicholson) * /, apps/app_speech_utils.c: Merged revisions 54714 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r54714 | file | 2007-02-15 19:48:48 -0500 (Thu, 15 Feb 2007) | 2 lines Don't let dtmf leak over into the engine and let it skew the results... also give DTMF results priority. (issue #9014 reported by surftek) ........ * main/manager.c: Properly handle an error result from a manager action. This could have left the action list permanently locked for reading. 2007-02-15 20:29 +0000 [r54654-54686] Olle Johansson * apps/app_voicemail.c: - add some notes, asking for help - insert a few ast_strlen_zero - Doxygen additions - A few more spaces * main/io.c: Make file's new comment doxygenified 2007-02-15 16:24 +0000 [r54624] Joshua Colp * apps/app_dial.c, /: Merged revisions 54623 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r54623 | file | 2007-02-15 11:19:39 -0500 (Thu, 15 Feb 2007) | 10 lines Merged revisions 54622 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r54622 | file | 2007-02-15 11:14:40 -0500 (Thu, 15 Feb 2007) | 2 lines Use a separate variable to indicate execution should continue instead of the return value. (issue #8842 reported by pluto70) ........ ................ 2007-02-15 15:53 +0000 [r54574-54599] Olle Johansson * CHANGES: ...and don't forget to update CHANGES * channels/chan_sip.c: Add callgroup and pickupgroup to SIPPEER function. (thanks ramon) * CHANGES: Update CHANGES * channels/chan_sip.c, configs/extconfig.conf.sample, doc/realtime.txt: Issue #7443 - amdtech - Optionally SIP registrations in another realtime family. 2007-02-15 02:11 +0000 [r54489-54552] Joshua Colp * main/io.c: Clean up the I/O context handler. * apps/app_flash.c, apps/app_image.c, apps/app_exec.c: Few more code clean ups. * apps/app_milliwatt.c: Clean up app_milliwatt code. * apps/app_dial.c, /: Merged revisions 54481 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r54481 | file | 2007-02-14 16:07:23 -0500 (Wed, 14 Feb 2007) | 2 lines Forward begin DTMF frames as well as end. (issue #9068 reported by mhardeman) ........ 2007-02-14 20:45 +0000 [r54464-54466] Olle Johansson * main/asterisk.c: Show version in "core show settings" * CHANGES: Updates and re-organization to make it easier to digest this information * main/cdr.c, main/manager.c, include/asterisk/config.h, include/asterisk/cdr.h, include/asterisk/manager.h, main/asterisk.c, main/config.c: New CLI command "Core show settings" to list some core settings 2007-02-14 17:14 +0000 [r54404] Matthew Fredrickson * channels/chan_zap.c, /: Merged revisions 54375 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r54375 | mattf | 2007-02-14 10:56:40 -0600 (Wed, 14 Feb 2007) | 10 lines Merged revisions 54373 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r54373 | mattf | 2007-02-14 10:25:49 -0600 (Wed, 14 Feb 2007) | 2 lines When handling glare on a PRI, move the requested channel rather than hang up the old one. Fix for 8957 and 9011. ........ ................ 2007-02-14 17:02 +0000 [r54348-54379] Olle Johansson * configs/sip.conf.sample: Make documentation match the source code. * channels/chan_sip.c: Issue #9060 - host= parameter in sip.conf stopped working caused by outbound proxy patch. * channels/chan_sip.c: Add port number to SIPPEER dialplan function 2007-02-14 08:34 +0000 [r54325] Paul Cadach * codecs/codec_g722.c: I don't know how it worked earlier, but valgrind produces core every time you try to load codec_g722. Fixed. ;-) 2007-02-14 01:12 +0000 [r54291] Joshua Colp * main/channel.c, /: Merged revisions 54290 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r54290 | file | 2007-02-13 20:09:40 -0500 (Tue, 13 Feb 2007) | 2 lines Add G722 to ast_best_codec. If anyone disagrees with it's placement, feel free to change it. (issue #9045 reported by gork) ........ 2007-02-13 22:02 +0000 [r54067-54261] Russell Bryant * include/asterisk/devicestate.h, apps/app_meetme.c, res/res_features.c, include/asterisk/cli.h, main/devicestate.c, CHANGES, apps/app_queue.c, funcs/func_devstate.c (added), main/cli.c: This introduces a new dialplan function, DEVSTATE, which allows you to do some pretty cool things. First, you can get the device state of anything in the dialplan: NoOp(SIP/mypeer has state ${DEVSTATE(SIP/mypeer)}) NoOp(The conference room 1234 has state ${DEVSTATE(MeetMe:1234)}) Most importantly, this allows you to create custom device states so you can control phone lamps directly from the dialplan. Set(DEVSTATE(Custom:mycustomlamp)=BUSY) ... exten => mycustomlamp,hint,Custom:mycustomlamp * /, channels/chan_sip.c: Merged revisions 54204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r54204 | russell | 2007-02-13 13:42:00 -0600 (Tue, 13 Feb 2007) | 5 lines If we fail to create the SIP socket, then return -1 from reload_config() so that load_module() will return AST_MODULE_LOAD_DECLINE. Otherwise, the console will just get spammed with error messages every time chan_sip tries to send a message. ........ * /, channels/chan_sip.c: Merged revisions 54235 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r54235 | russell | 2007-02-13 15:31:22 -0600 (Tue, 13 Feb 2007) | 2 lines Remove a couple of leftover debug messages ........ * include/asterisk/devicestate.h, /: Merged revisions 54218 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r54218 | russell | 2007-02-13 14:56:50 -0600 (Tue, 13 Feb 2007) | 3 lines Fix the documentation on the return values from device state provider registration and deletion. ........ * main/asterisk.c: Use spaces instead of tabs in the help text for a CLI command * main/asterisk.c: Simplify WELCOME_MESSAGE to be a single function call instead of one for each line. * include/asterisk/cli.h, main/asterisk.c, main/cli.c: - Constify the format string passed to ast_cli() - Simplify printing out the warranty and license * main/dial.c, /, include/asterisk/dial.h: Merged revisions 54103 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r54103 | russell | 2007-02-12 13:17:08 -0600 (Mon, 12 Feb 2007) | 2 lines Change ast_set_state_callback() to ast_dial_set_state_callback() ........ * main/dial.c, /, apps/app_meetme.c, apps/app_page.c, include/asterisk/dial.h: Merged revisions 54066 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r54066 | russell | 2007-02-12 11:58:43 -0600 (Mon, 12 Feb 2007) | 4 lines - Add the ability to register a callback to monitor state changes in an asynchronous dial operation. - Rename the various references to "status" to "state" in the dial API ........ 2007-02-12 15:48 +0000 [r54003-54004] Russell Bryant * configs/users.conf.sample, /: Merged revisions 54002 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r54002 | russell | 2007-02-12 10:38:39 -0500 (Mon, 12 Feb 2007) | 2 lines Fix a typo where "vmpassword" should be "vmsecret" ........ * main/channel.c: Simplify a small bit of logic. 2007-02-12 02:44 +0000 [r53980] Tilghman Lesher * funcs/func_realtime.c: Formatting fixes 2007-02-11 20:49 +0000 [r53914-53953] Olle Johansson * channels/chan_sip.c: Be careful with debug messages in trunk, they tend to stay around for release.... * channels/chan_sip.c: Small fix in outbound proxy support. * channels/chan_sip.c, configs/sip.conf.sample: Add support for outbound proxy for peers and [general] This replaces the older, broken, implementation where a setting in [general] did not do anything and the [peer] part was broken. * main/acl.c: Fix debug handling in acl.c 2007-02-10 09:23 +0000 [r53882-53885] Paul Cadach * /, channels/chan_h323.c: Merged revisions 53881 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53881 | pcadach | 2007-02-10 01:09:49 -0800 (Сбт, 10 Фев 2007) | 1 line Fix VLDTMF reception ........ * /, apps/app_echo.c: Merged revisions 53880 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53880 | pcadach | 2007-02-10 01:08:55 -0800 (Сбт, 10 Фев 2007) | 1 line Much simpler than previous one ;-) ........ * main/channel.c, /: Merged revisions 53879 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53879 | pcadach | 2007-02-10 01:07:11 -0800 (Сбт, 10 Фев 2007) | 1 line Provide correct DTMF duration ........ 2007-02-10 06:14 +0000 [r53851] Kevin P. Fleming * /, configure, configure.ac: Merged revisions 53850 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53850 | kpfleming | 2007-02-10 00:06:08 -0600 (Sat, 10 Feb 2007) | 3 lines don't display the --with-imap message unless --with-imap was specified without a path use '-n' instead of '! -z' for tests ........ 2007-02-10 00:42 +0000 [r53784-53819] Russell Bryant * include/asterisk/app.h, include/asterisk/utils.h, main/dial.c, /, apps/app_meetme.c, channels/chan_sip.c, doc/sla.txt (added), include/asterisk/dial.h, configs/sla.conf.sample: Merged revisions 53810 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53810 | russell | 2007-02-09 18:35:09 -0600 (Fri, 09 Feb 2007) | 24 lines Merge team/russell/sla_rewrite This is a completely new implementation of the SLA functionality introduced in Asterisk 1.4. It is now functional and ready for testing. However, I will be adding some additional features over the next week, as well. For information on how to set this up, see configs/sla.conf.sample and doc/sla.txt. In addition to the changes in app_meetme.c for the SLA implementation itself, this merge brings in various other changes: chan_sip: - Add the ability to indicate HOLD state in NOTIFY messages. - Queue HOLD and UNHOLD control frames even if the channel is not bridged to another channel. linkedlists.h: - Add support for rwlock based linked lists. dial.c: - Add the ability to run ast_dial_start() without a reference channel to inherit information from. ........ * channels/chan_jingle.c: add another dependency * /, apps/app_echo.c: Merged revisions 53783 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53783 | russell | 2007-02-09 18:15:50 -0600 (Fri, 09 Feb 2007) | 4 lines When the Echo() application receives the digit '#', echo that back as well. Since we already sent the BEGIN frame for that digit, it makes sense to send the END as well. ........ 2007-02-09 23:53 +0000 [r53782] Kevin P. Fleming * build_tools/get_moduleinfo, res/res_config_odbc.c, /, build_tools/get_makeopts, funcs/func_odbc.c, res/res_adsi.c, channels/chan_gtalk.c, apps/app_adsiprog.c, apps/app_voicemail.c: Merged revisions 53779-53781 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53779 | kpfleming | 2007-02-09 17:51:29 -0600 (Fri, 09 Feb 2007) | 2 lines fix awk scripts to work when both MODULEINFO and MAKEOPTS are present in a source file ........ r53780 | kpfleming | 2007-02-09 17:51:41 -0600 (Fri, 09 Feb 2007) | 2 lines add some inter-module dependencies ........ r53781 | kpfleming | 2007-02-09 17:52:44 -0600 (Fri, 09 Feb 2007) | 2 lines another dependency ........ 2007-02-09 19:39 +0000 [r53717-53750] Joshua Colp * apps/app_dial.c, /: Merged revisions 53749 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53749 | file | 2007-02-09 14:33:31 -0500 (Fri, 09 Feb 2007) | 2 lines Temporarily change musicclass on channel to one specified in Dial so that the 'm' option functions properly. (issue #8969 reported by christianbee) ........ * apps/app_queue.c: Clean up documentation of Queue application. (issue #9022 reported by seanbright) 2007-02-09 16:43 +0000 [r53716] Kevin P. Fleming * doc/imapstorage.txt, /, configure, configure.ac: Merged revisions 53715 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53715 | kpfleming | 2007-02-09 10:42:22 -0600 (Fri, 09 Feb 2007) | 2 lines clarify the fact that voicemail IMAP storage cannot be built against a distro's binary c-client library package (at least not at this time) ........ 2007-02-09 01:57 +0000 [r53602-53691] Joshua Colp * res/res_musiconhold.c: I'm crazy so I think I'll change the musiconhold classes linked list to read/write as well! * main/manager.c: It is with pleasure that I announce the return of rawman support through the HTTP server. (issue #9013 reported by Jynger) * /, apps/app_speech_utils.c: Merged revisions 53601 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53601 | file | 2007-02-08 12:54:32 -0500 (Thu, 08 Feb 2007) | 2 lines Fix timeout issue when utterance is longer then timeout itself. ........ 2007-02-08 17:19 +0000 [r53580] Jason Parker * channels/chan_sip.c: Rename this instance of "busy limit" to "busy level" as well 2007-02-08 16:41 +0000 [r53577] Kevin P. Fleming * channels/chan_sip.c, configs/sip.conf.sample: rename busy-limit to busy-level, since it is not a limit actually parse the busy-limit option from sip.conf, instead of ignoring it 2007-02-08 13:50 +0000 [r53531-53533] Tilghman Lesher * /, main/loader.c: Merged revisions 53532 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53532 | tilghman | 2007-02-08 07:47:54 -0600 (Thu, 08 Feb 2007) | 2 lines Issue 9007 - Mutex not released on early return ........ * /, apps/app_voicemail.c: Merged revisions 53530 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r53530 | tilghman | 2007-02-08 07:40:02 -0600 (Thu, 08 Feb 2007) | 10 lines Merged revisions 53529 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r53529 | tilghman | 2007-02-08 07:36:10 -0600 (Thu, 08 Feb 2007) | 2 lines Issue 9003 - If fullname is empty, quote() passes back "\"" ........ ................ 2007-02-07 23:56 +0000 [r53465-53498] Russell Bryant * /, main/db1-ast/Makefile: Merged revisions 53497 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53497 | russell | 2007-02-07 17:52:45 -0600 (Wed, 07 Feb 2007) | 6 lines When building libdb1.a, put the additional flags needed at the beginning of ASTCFLAGS, instead of at the end. This way, we ensure that we find the local headers first before accidentally trying to use headers that exist in locations specified in the ASTCFLAGS passed from the main Makefile. (issue #8637, ovi) ........ * /, main/Makefile: Merged revisions 53464 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53464 | russell | 2007-02-07 14:07:39 -0600 (Wed, 07 Feb 2007) | 4 lines The clean target actually needs to run "distclean" on editline. This is because we need to make sure that its configure script gets executed again, because the CFLAGS we want to pass to editline may have changed. ........ 2007-02-07 17:57 +0000 [r53435] Joshua Colp * /, main/rtp.c: Merged revisions 53434 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53434 | file | 2007-02-07 12:53:03 -0500 (Wed, 07 Feb 2007) | 2 lines We can not reliably do P2P bridging with DTMF passing back with compensation if we need to listen for DTMF frames. (issue #8962 reported by caio1982) ........ 2007-02-07 17:46 +0000 [r53431] Russell Bryant * /, main/rtp.c: Merged revisions 53429 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53429 | russell | 2007-02-07 11:39:31 -0600 (Wed, 07 Feb 2007) | 7 lines When parsing the NTP timestamp in a sender report message, you are supposed to take the low 16 bits of the integer part, and the high 16 bits of the fractional part. However, the code here was erroneously taking the low 16 bits of the fractional part. It then shifted the result 16 bits down, so the result was always zero. This fix makes it grab the appropriate high 16 bits, instead. (issue #8991, pointed out by andre_abrantes) ........ 2007-02-07 17:06 +0000 [r53359-53400] Joshua Colp * /, apps/app_playback.c: Merged revisions 53399 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53399 | file | 2007-02-07 12:04:44 -0500 (Wed, 07 Feb 2007) | 2 lines Directly load say.conf in load_module instead of calling the reload function. (issue #8946 reported by junky) ........ * /, channels/chan_iax2.c: Merged revisions 53358 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r53358 | file | 2007-02-07 10:43:39 -0500 (Wed, 07 Feb 2007) | 10 lines Merged revisions 53357 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r53357 | file | 2007-02-07 10:38:48 -0500 (Wed, 07 Feb 2007) | 2 lines Fix a few potential memory leaks with realtime users and peers. (issue #8999 reported by bsmithurst) ........ ................ 2007-02-07 15:35 +0000 [r53356] Tilghman Lesher * /, apps/app_macro.c: Merged revisions 53355 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r53355 | tilghman | 2007-02-07 09:33:51 -0600 (Wed, 07 Feb 2007) | 10 lines Merged revisions 53354 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r53354 | tilghman | 2007-02-07 09:30:02 -0600 (Wed, 07 Feb 2007) | 2 lines Issue 7440 - Macro called from Macro from the h extension exits prematurely ........ ................ 2007-02-07 09:51 +0000 [r53334] Christian Richter * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged revisions 53324 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r53324 | crichter | 2007-02-07 10:22:44 +0100 (Mi, 07 Feb 2007) | 9 lines Merged revisions 52843 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r52843 | crichter | 2007-01-30 15:38:08 +0100 (Di, 30 Jan 2007) | 1 line fixed some possible segfaults. also fixed an very important bug which occurs on high load (when calls are very fast generated) ........ ................ 2007-02-07 05:25 +0000 [r53247-53297] Tilghman Lesher * /, res/res_jabber.c: Merged revisions 53294 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53294 | tilghman | 2007-02-06 23:24:31 -0600 (Tue, 06 Feb 2007) | 2 lines Text fix for jabber reload command (reported by bkruse via IRC) ........ * main/manager.c, /: Merged revisions 53246 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r53246 | tilghman | 2007-02-06 01:00:52 -0600 (Tue, 06 Feb 2007) | 10 lines Merged revisions 53245 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r53245 | tilghman | 2007-02-06 00:58:28 -0600 (Tue, 06 Feb 2007) | 2 lines Issue 8987 - Status could return two responses (mnicholson) ........ ................ 2007-02-05 21:55 +0000 [r53200] Olle Johansson * main/io.c: Doxygen formatting changes 2007-02-05 17:06 +0000 [r53151-53153] Joshua Colp * /, apps/app_playback.c: Merged revisions 53152 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53152 | file | 2007-02-05 11:06:18 -0600 (Mon, 05 Feb 2007) | 2 lines Ensure say_cfg is NULL when the module is loaded. (issue #8946 reported by junky) ........ * /, apps/app_playback.c: Merged revisions 53150 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53150 | file | 2007-02-05 10:02:00 -0600 (Mon, 05 Feb 2007) | 2 lines Unregister Playback CLI commands as well as dialplan application. (issue #8946 reported by junky) ........ 2007-02-05 00:30 +0000 [r53144] Olle Johansson * /, channels/chan_sip.c: Merged revisions 53143 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53143 | oej | 2007-02-05 01:18:34 +0100 (Mon, 05 Feb 2007) | 3 lines Add some comments on queue system behaviour and how it affects the SIP channel ........ 2007-02-03 22:06 +0000 [r53140-53142] Tilghman Lesher * UPGRADE.txt: Deprecate SetCallerPres application * apps/app_setcallerid.c, funcs/func_callerid.c: Add CALLERPRES dialplan function and deprecate SetCallerPres application * funcs/func_odbc.c: Fix compiler warnings 2007-02-03 21:06 +0000 [r53139] Joshua Colp * /, channels/chan_sip.c: Merged revisions 53138 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53138 | file | 2007-02-03 15:05:02 -0600 (Sat, 03 Feb 2007) | 2 lines Make SIPDtmfMode application work with recent capability changes, and also fix an RTP stack issue when the auto option was used. (issue #8972 reported by mdu113) ........ 2007-02-03 20:46 +0000 [r53137] Russell Bryant * apps/app_dial.c, /: Merged revisions 53136 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r53136 | russell | 2007-02-03 14:44:20 -0600 (Sat, 03 Feb 2007) | 12 lines Merged revisions 53133 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r53133 | russell | 2007-02-03 14:38:13 -0600 (Sat, 03 Feb 2007) | 4 lines set the DIALSTATUS variable to contain "INVALIDARGS" when the dial application exits early because of invalid arguments instead of just leaving it empty. (issue #8975) ........ ................ 2007-02-03 10:12 +0000 [r53132] Paul Cadach * /, channels/h323/ast_h323.cxx: Merged revisions 53131 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53131 | pcadach | 2007-02-03 02:02:55 -0800 (Сбт, 03 Фев 2007) | 1 line Remove quote from H.323 vendor string because due to compatibilities with Nortel Meridian CS1000 reported at www.voip-info.org ........ 2007-02-02 20:05 +0000 [r53126-53127] Olle Johansson * doc/queue.txt: Update with info about SIP channels and queues * doc/queue.txt (added): Adding a template for documentation on call queues. Please help us add to this! Thanks /OEJ and BJ 2007-02-02 18:21 +0000 [r53111-53125] Joshua Colp * channels/chan_sip.c: Add onHold value to sip show inuse as well. * /, main/rtp.c: Merged revisions 53120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53120 | file | 2007-02-02 11:15:22 -0600 (Fri, 02 Feb 2007) | 2 lines Correct a copy/pasted error message line for RTCP. ........ * /, main/config.c: Merged revisions 53118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r53118 | file | 2007-02-02 10:59:53 -0600 (Fri, 02 Feb 2007) | 10 lines Merged revisions 53117 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r53117 | file | 2007-02-02 10:58:09 -0600 (Fri, 02 Feb 2007) | 2 lines Pass the glob expanded filename to process_text_line so that error messages contain the actual filename, not the original include one. (issue #8959 reported by tzafrir) ........ ................ * Makefile, /: Merged revisions 53114 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53114 | file | 2007-02-02 09:29:35 -0600 (Fri, 02 Feb 2007) | 2 lines Add systemname to asterisk.conf generation per recent discussions about it. (issue #8968 reported by blitzrage) ........ * main/devicestate.c: Clean up ast_device_state. It's pretty now! * main/devicestate.c: Switch the devicestate thread to operate the same way as the logging thread. Pops all entries off the list to be processed, resets the list back to a clean state, and processes each entry. The thread won't have to acquire the list lock again until it checks to see if there are more to process. * main/devicestate.c: Read/write lockify the devicestate stuff a bit. 2007-02-02 00:26 +0000 [r53110] Olle Johansson * /, channels/chan_sip.c, configs/sip.conf.sample: Patch based on this patch with small changes for trunk... Merged revisions 53109 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53109 | oej | 2007-02-02 01:24:03 +0100 (Fri, 02 Feb 2007) | 4 lines Disable the direct p2p RTP call setup in SIP. You can enable it in sip.conf, but it is now considered experimental until we solve the AST_CONTROL_ANSWER with payload and videocaps stuff. ........ 2007-02-01 22:26 +0000 [r53098-53105] Joshua Colp * /, channels/chan_sip.c: Merged revisions 53104 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r53104 | file | 2007-02-01 16:24:32 -0600 (Thu, 01 Feb 2007) | 10 lines Merged revisions 53103 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r53103 | file | 2007-02-01 16:21:56 -0600 (Thu, 01 Feb 2007) | 2 lines Copy noncodeccapability over to the joint variable so that telephone-event will get transmitted in the sent INVITE. ........ ................ * /, channels/chan_sip.c: Merged revisions 53097 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r53097 | file | 2007-02-01 15:54:28 -0600 (Thu, 01 Feb 2007) | 10 lines Merged revisions 53095 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r53095 | file | 2007-02-01 15:47:11 -0600 (Thu, 01 Feb 2007) | 2 lines Don't negotiate RFC2833 when not configured to do so. (issue #8799 reported by mdu113) ........ ................ 2007-02-01 21:27 +0000 [r53094] Russell Bryant * /, funcs/func_strings.c: Merged revisions 53093 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53093 | russell | 2007-02-01 15:24:52 -0600 (Thu, 01 Feb 2007) | 2 lines Fix the FIELDQTY function to not crash. (reported by blitzrage and Corydon on IRC) ........ 2007-02-01 21:17 +0000 [r53092] Olle Johansson * /, channels/chan_sip.c: Merged revisions 53085 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53085 | oej | 2007-02-01 22:05:34 +0100 (Thu, 01 Feb 2007) | 4 lines - Clean INC_COUNT flag when we decrement call counter - If it's still set at time of dialog destruction, make sure we decrement the device call counter properly before we destroy the dialog ........ 2007-02-01 21:12 +0000 [r53087-53089] Joshua Colp * /, res/res_musiconhold.c: Merged revisions 53088 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r53088 | file | 2007-02-01 15:11:28 -0600 (Thu, 01 Feb 2007) | 10 lines Merged revisions 53084 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r53084 | file | 2007-02-01 15:03:10 -0600 (Thu, 01 Feb 2007) | 2 lines Return previous behavior of having MOH pick up where it was left off. (issue #8672 reported by sinistermidget) ........ ................ 2007-02-01 20:44 +0000 [r53080-53083] Olle Johansson * /, apps/app_queue.c: Merged revisions 53081 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53081 | oej | 2007-02-01 21:38:58 +0100 (Thu, 01 Feb 2007) | 2 lines Change debug level for state change message that is not really informative when debugging app_queue ........ * channels/chan_sip.c, configs/sip.conf.sample: Implementing "busy-limit". If you set call limit and busy limit, chan_sip will indicate BUSY for a device that has reached the busy limit and allow calls up to the call limit, allowing for call transfers (that generate a new call). If you only set call limit, chan_sip will not indicate BUSY until that limit is filled. This affects SIP subscriptions, call queues and manager applications. * /, channels/chan_sip.c: Merged revisions 53079 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53079 | oej | 2007-02-01 21:28:54 +0100 (Thu, 01 Feb 2007) | 2 lines Cleaning up the devicestate callback function ........ 2007-02-01 20:14 +0000 [r53076-53078] Tilghman Lesher * /, funcs/func_strings.c: Merged revisions 53075 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r53075 | tilghman | 2007-02-01 14:09:52 -0600 (Thu, 01 Feb 2007) | 10 lines Merged revisions 53074 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r53074 | tilghman | 2007-02-01 14:07:35 -0600 (Thu, 01 Feb 2007) | 2 lines Bug 8965 - Allow FIELDQTY to work with both variables and dialplan functions ........ ................ 2007-02-01 19:34 +0000 [r53073] Joshua Colp * /, main/asterisk.c: Merged revisions 53072 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53072 | file | 2007-02-01 13:33:33 -0600 (Thu, 01 Feb 2007) | 2 lines Add missing 'F' letter to getopt so it magically becomes a valid option. (issue #8960 reported by tzafrir) ........ 2007-02-01 19:27 +0000 [r53071] Tilghman Lesher * main/pbx.c, /, funcs/func_strings.c: Merged revisions 53070 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r53070 | tilghman | 2007-02-01 13:21:20 -0600 (Thu, 01 Feb 2007) | 10 lines Merged revisions 53069 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r53069 | tilghman | 2007-02-01 13:13:53 -0600 (Thu, 01 Feb 2007) | 2 lines No wonder FIELDQTY doesn't work with functions... the documentation in pbx.c was wrong ........ ................ 2007-02-01 19:04 +0000 [r53067] Olle Johansson * channels/chan_sip.c: Signal HOLD status to phones that subscribe for status. 2007-02-01 17:42 +0000 [r53065-53066] Joshua Colp * /, channels/chan_sip.c: Merged revisions 53064 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53064 | file | 2007-02-01 11:37:44 -0600 (Thu, 01 Feb 2007) | 2 lines Fix silly logic. We really want to write UDPTL frames out when the call is up. ........ * main/db1-ast/hash/hash.c: Make trunk compile under dev mode. 2007-02-01 16:42 +0000 [r53063] Olle Johansson * /, configs/sip.conf.sample: Merged revisions 53062 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53062 | oej | 2007-02-01 17:35:12 +0100 (Thu, 01 Feb 2007) | 2 lines Add explanation of port= in combination with defaultip= (thanks jsmith) ........ 2007-02-01 14:43 +0000 [r53061] Russell Bryant * apps/app_rpt.c: Remove duplicate calls to pthread_attr_destroy() that I put in yesterday by accident. 2007-02-01 11:16 +0000 [r53058-53059] Paul Cadach * /, channels/chan_h323.c: Oops -- Merged revisions 53057 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53057 | pcadach | 2007-02-01 03:07:41 -0800 (Чтв, 01 Фев 2007) | 1 line chan_h323 is very stable, so let it built by default ........ 2007-02-01 00:38 +0000 [r53054] Olle Johansson * res/res_features.c: Formatting changes 2007-02-01 00:24 +0000 [r53051-53053] Joshua Colp * /, main/rtp.c: Merged revisions 53052 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53052 | file | 2007-01-31 18:24:20 -0600 (Wed, 31 Jan 2007) | 2 lines When going on hold have the side that was put on hold reinvite back to Asterisk. When going off hold have the side that was taken off hold reinvited back to the other party. ........ * /, main/rtp.c: Merged revisions 53050 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53050 | file | 2007-01-31 18:19:48 -0600 (Wed, 31 Jan 2007) | 2 lines Add more frame types to forward in the RTP bridge loops. ........ 2007-01-31 21:35 +0000 [r52905-53047] Russell Bryant * main/pbx.c, channels/chan_zap.c, /, apps/app_meetme.c, channels/chan_sip.c, apps/app_queue.c, channels/chan_iax2.c, main/cdr.c, main/manager.c, pbx/pbx_spool.c, channels/chan_skinny.c, channels/chan_h323.c, main/http.c, pbx/pbx_dundi.c, apps/app_rpt.c, channels/chan_mgcp.c: Merged revisions 53046 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r53046 | russell | 2007-01-31 15:32:08 -0600 (Wed, 31 Jan 2007) | 11 lines Merged revisions 53045 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r53045 | russell | 2007-01-31 15:25:11 -0600 (Wed, 31 Jan 2007) | 3 lines Fix a bunch of places where pthread_attr_init() was called, but pthread_attr_destroy() was not. ........ ................ * /, apps/app_userevent.c: Merged revisions 53042 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53042 | russell | 2007-01-31 12:18:25 -0600 (Wed, 31 Jan 2007) | 2 lines Remove an extra \r\n from manager user events. (issue #8955, mnicholson) ........ * /, main/rtp.c: Merged revisions 53040 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r53040 | russell | 2007-01-31 11:45:05 -0600 (Wed, 31 Jan 2007) | 11 lines Merged revisions 53039 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r53039 | russell | 2007-01-31 11:41:51 -0600 (Wed, 31 Jan 2007) | 3 lines Use the proper format string to print unsigned values in the rtp debug output. (issue #8954, wmis) ........ ................ * /, apps/app_queue.c: Merged revisions 53037 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53037 | russell | 2007-01-31 11:39:28 -0600 (Wed, 31 Jan 2007) | 3 lines Only changed the paused status in an existing queue member if the paused column exists. ........ * /, apps/app_queue.c: Merged revisions 53035 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53035 | russell | 2007-01-31 11:34:22 -0600 (Wed, 31 Jan 2007) | 4 lines Instead of always creating a realtime queue member as unpaused, read the "paused" column and use that value for the paused status of the member. (issue #8949, jmls) ........ * /, contrib/init.d/rc.suse.asterisk: Merged revisions 53001 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53001 | russell | 2007-01-30 17:38:42 -0600 (Tue, 30 Jan 2007) | 2 lines Update init script for SuSE 10. (issue #8363, johnlange) ........ * /, doc/cdrdriver.txt: Merged revisions 52999 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52999 | russell | 2007-01-30 17:30:34 -0600 (Tue, 30 Jan 2007) | 2 lines Add documentation for using cdr_pgsql. (issue #8942, lters) ........ * /, configure, include/asterisk/autoconfig.h.in, configure.ac, codecs/codec_gsm.c: Merged revisions 52997 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52997 | russell | 2007-01-30 17:23:24 -0600 (Tue, 30 Jan 2007) | 5 lines When we are checking for a system installed version of libgsm, we need to check for gsm.h as well. Furthermore, when checking for this header, it may be located in a gsm/ sub directory, so check for that, as well. (issue #8773) ........ * /, channels/chan_sip.c: Merged revisions 52952 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52952 | russell | 2007-01-30 13:33:12 -0600 (Tue, 30 Jan 2007) | 5 lines Only set the DTMF flag on the rtp structure if the DTMF mode is actually RFC2833, not just that it is not INFO. This makes it get set for inband DTMF as well, which is not valid. (issue #8936) ........ * /, main/asterisk.c: Merged revisions 52904 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r52904 | russell | 2007-01-30 11:19:39 -0600 (Tue, 30 Jan 2007) | 17 lines Merged revisions 52903 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r52903 | russell | 2007-01-30 11:12:04 -0600 (Tue, 30 Jan 2007) | 9 lines The SIGHUP handler was implemented to allow admins to send SIGHUP to a running Asterisk process to reload the configuration. However, doing the actual reload in the signal handler itself is a very bad thing to do, because the reload process includes calling non-reentrant functions such as malloc/calloc/etc. If Asterisk is running in the background, then the reload will happen immediately. However, if running in console mode, the reload doesn't work until something is typed at the console. That sort of defeats the purpose, but I don't see an easy way to get around it at this point. ........ ................ 2007-01-30 15:39 +0000 [r52858-52860] Joshua Colp * channels/chan_sip.c: Use provided variable for name instead of one in the structure since the structure was just allocated and will be NULL. (issue #8938 reported by st41ker) 2007-01-30 09:13 +0000 [r52818-52820] Paul Cadach * /, res/res_odbc.c: Merged revisions 52808 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52808 | pcadach | 2007-01-30 00:34:26 -0800 (Втр, 30 Янв 2007) | 1 line Don't play with free()'d pointers ........ * /, configure, acinclude.m4: Merged revisions 52807 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52807 | pcadach | 2007-01-30 00:33:22 -0800 (Втр, 30 Янв 2007) | 1 line Handle non-standard OpenH323/PWLib library names ........ 2007-01-30 00:16 +0000 [r52764] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 52763 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r52763 | russell | 2007-01-29 18:15:50 -0600 (Mon, 29 Jan 2007) | 13 lines Merged revisions 52762 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r52762 | russell | 2007-01-29 18:15:06 -0600 (Mon, 29 Jan 2007) | 5 lines Fix the extraction of the timestamp from video frames. It was using the mapping for a mini-frame instead of a video-frame, which caused it to get invalid data. (issue #8795, mihai) ........ ................ 2007-01-29 23:45 +0000 [r52718] Joshua Colp * /, apps/app_mixmonitor.c: Merged revisions 52717 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r52717 | file | 2007-01-29 18:43:40 -0500 (Mon, 29 Jan 2007) | 10 lines Merged revisions 52716 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r52716 | file | 2007-01-29 18:39:39 -0500 (Mon, 29 Jan 2007) | 2 lines Now that filename is part of the structure and since it comes before postprocess... we have to add it to our postprocess line. (reported on asterisk-dev by Boris Bakchiev) ........ ................ 2007-01-29 22:58 +0000 [r52692-52696] Russell Bryant * /, main/Makefile: Merged revisions 52695 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52695 | russell | 2007-01-29 16:58:09 -0600 (Mon, 29 Jan 2007) | 2 lines Add a missing quotation mark. This was pointed out by jcmoore on #asterisk-dev. ........ * main/manager.c, /: Merged revisions 52688 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52688 | russell | 2007-01-29 16:55:41 -0600 (Mon, 29 Jan 2007) | 3 lines Remove a recursive lock of the manager session. This was pointed out by zandbelt in issue #8711. ........ 2007-01-29 22:13 +0000 [r52680] Tilghman Lesher * /, pbx/pbx_config.c: Merged revisions 52679 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52679 | tilghman | 2007-01-29 16:12:12 -0600 (Mon, 29 Jan 2007) | 2 lines Argument number correction ........ 2007-01-29 21:37 +0000 [r52646-52648] Russell Bryant * /, main/Makefile: Merged revisions 52647 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52647 | russell | 2007-01-29 15:36:56 -0600 (Mon, 29 Jan 2007) | 3 lines ASTLDFLAGS needs to be passed to the editline configure script as LDFLAGS. (issue #8928, zandbelt) ........ * /, main/rtp.c: Merged revisions 52645 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52645 | russell | 2007-01-29 15:26:27 -0600 (Mon, 29 Jan 2007) | 6 lines Fix a problem with packet-to-packet bridging and DTMF mode translation. P2P bridging can only be used when the DTMF modes don't match if the core is monitoring DTMF in both directions. Then, the core will handle the translation. Otherwise, this bridging method can not be used. (issue #8936) ........ 2007-01-29 21:03 +0000 [r52635] Joshua Colp * main/rtp.c: Only use locking for bridge information if intense P2P bridging is enabled. 2007-01-29 20:51 +0000 [r52612-52613] Russell Bryant * main/manager.c, /: The changes for trunk are less extensive, but include - changing the actionlock to a rwlock - not locking the session before doing the action callback The crash issue in 8711 should not be an issue here. Merged revisions 52611 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52611 | russell | 2007-01-29 14:39:20 -0600 (Mon, 29 Jan 2007) | 10 lines The session lock can not be held while calling action callbacks. If so, then when the WaitEvent callback gets called, then no event can happen because the session can't be locked by another thread. Also, the session needs to be locked in the HTTP callback when it reads out the output string. This fixes the deadlock reported in both 8711 and 8934. Regarding issue 8711, there still may be an issue. If there is a second action requested before the processing of the first action is finished, there could still be some corruption of the output string buffer used to build the result. (issue #8711, #8934) ........ * apps/app_voicemail.c: Resolve some warnings when not building with IMAP_STORAGE 2007-01-29 20:22 +0000 [r52580-52610] Joshua Colp * apps/app_voicemail.c: Change vmstates list to use linked list macros. * apps/app_voicemail.c: Code cleanup of IMAP storage support in app_voicemail. * /, apps/app_voicemail.c: Merged revisions 52572 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52572 | file | 2007-01-29 13:59:41 -0500 (Mon, 29 Jan 2007) | 2 lines Use ast_calloc instead of malloc. ........ 2007-01-29 17:49 +0000 [r52524-52525] Joshua Colp * CHANGES, main/cli.c: Add core show channels count CLI command. (issue #8932 reported by mr_mehul_shah) * /, apps/app_voicemail.c: Merged revisions 52523 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52523 | file | 2007-01-29 12:33:19 -0500 (Mon, 29 Jan 2007) | 2 lines Set quota information to 0 when creating a vm_state. (issue #8924 reported by neutrino88) ........ 2007-01-29 17:03 +0000 [r52522] Russell Bryant * /, main/jitterbuf.c, include/jitterbuf.h: Merged revisions 52494,52506 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52494 | jdixon | 2007-01-28 22:18:36 -0600 (Sun, 28 Jan 2007) | 4 lines Fixed problem with jitterbuf, whereas it would not complain about, and would allow itself to be overfilled (per the max_jitterbuf parameter). Now it rejects any data over and above that size, and complains about it. ........ r52506 | russell | 2007-01-29 10:54:27 -0600 (Mon, 29 Jan 2007) | 5 lines Clean up a few things in the last commit to the adaptive jitterbuffer code. - Specifically indicate to the compiler that the "dropem" variable only needs one but. - Change formatting to conform to coding guidelines. ........ 2007-01-28 05:18 +0000 [r52463] Tilghman Lesher * /, configure, configure.ac: Merged revisions 52462 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52462 | tilghman | 2007-01-27 23:15:07 -0600 (Sat, 27 Jan 2007) | 2 lines Suggested change to fix normal usage of --with-tds=/usr/local (Sean Bright, via asterisk-dev mailing list) ........ 2007-01-27 02:15 +0000 [r52332-52417] Joshua Colp * /, apps/app_queue.c: Merged revisions 52416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r52416 | file | 2007-01-26 21:13:41 -0500 (Fri, 26 Jan 2007) | 10 lines Merged revisions 52415 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r52415 | file | 2007-01-26 21:09:10 -0500 (Fri, 26 Jan 2007) | 2 lines Make COMPLETECALLER and COMPLETEAGENT output to queue_log follow documentation. (issue #7677 reported by amilcar) ........ ................ * /, channels/chan_iax2.c: Merged revisions 52370 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r52370 | file | 2007-01-26 19:08:18 -0500 (Fri, 26 Jan 2007) | 10 lines Merged revisions 52360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r52360 | file | 2007-01-26 19:03:23 -0500 (Fri, 26 Jan 2007) | 2 lines Make the last context entry read in the dominant one. (issue #8918 reported by pj) ........ ................ * /, main/file.c: Merged revisions 52335 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52335 | file | 2007-01-26 18:46:47 -0500 (Fri, 26 Jan 2007) | 2 lines Fix core show file formats CLI command. ........ * main/file.c, main/image.c: Convert some more stuff to read/write lists. 2007-01-25 22:49 +0000 [r52168-52308] Joshua Colp * CHANGES, main/db.c: Add DBDel and DBDelTree manager commands. (issue #8516 reported by dprado) * /, main/jitterbuf.c: Merged revisions 52265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r52265 | file | 2007-01-25 14:18:33 -0500 (Thu, 25 Jan 2007) | 10 lines Merged revisions 52264 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r52264 | file | 2007-01-25 14:15:29 -0500 (Thu, 25 Jan 2007) | 2 lines Allow dequeueing of frames with negative timestamp by moving jitterbuffer frames check to jb_next. (issue #8546 reported by harmen) ........ ................ * channels/chan_sip.c: Use atomic operation functions for use/ringing/hold manipulation. * /, channels/chan_sip.c: Merged revisions 52210 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52210 | file | 2007-01-25 12:49:39 -0500 (Thu, 25 Jan 2007) | 2 lines Drop out variables I accidentally put in. ........ * /, channels/chan_sip.c: Merged revisions 52208 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52208 | file | 2007-01-25 12:14:53 -0500 (Thu, 25 Jan 2007) | 2 lines Decrement onHold count if we are hung up on and still on hold. (issue #8909 reported by alexh42) ........ * /, apps/app_mixmonitor.c: Merged revisions 52163 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r52163 | file | 2007-01-24 20:51:35 -0500 (Wed, 24 Jan 2007) | 10 lines Merged revisions 52162 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r52162 | file | 2007-01-24 20:48:52 -0500 (Wed, 24 Jan 2007) | 2 lines Add another note about audio files being played back to each bridged party. (issue #8718 reported by ppyy) ........ ................ 2007-01-25 01:38 +0000 [r52108-52161] Russell Bryant * configs/users.conf.sample, /, apps/app_voicemail.c: Merged revisions 52160 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52160 | russell | 2007-01-24 19:37:16 -0600 (Wed, 24 Jan 2007) | 2 lines By suggestion from kpfleming last week, change "vmpassword" to "vmsecret". ........ * /, include/asterisk/dial.h: Merged revisions 52107 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52107 | russell | 2007-01-24 15:42:47 -0600 (Wed, 24 Jan 2007) | 3 lines Fix the formatting of doxygen comments to properly indicate that the comment documents the previous entity, as opposed to the next one. ........ 2007-01-24 20:35 +0000 [r52053-52086] Steve Murphy * UPGRADE.txt, apps/app_chanisavail.c: As per bug 8859 (Add option to revert old ChanIsAvail() with 's' option behavior), this update makes the 't' option available, which calls ast_parse_device_state instead of ast_device_state. This option will not dive into the channel driver to find the status of the device (which could be good if sip devicestate isn't returning full status, for various reasons). * utils/Makefile, /, utils/check_expr.c: Merged revisions 52052 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r52052 | murf | 2007-01-24 11:26:22 -0700 (Wed, 24 Jan 2007) | 9 lines Merged revisions 52002 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r52002 | murf | 2007-01-24 10:43:50 -0700 (Wed, 24 Jan 2007) | 1 line updated check_expr via 8322 (refactoring of expression checking impl); elfring contributed a nice code reorg, I contributed some time to get it working again, better messages ........ ................ 2007-01-24 18:23 +0000 [r52025-52050] Joshua Colp * main/dial.c (added), /, apps/app_page.c, main/Makefile, include/asterisk/dial.h (added): Merged revisions 52049 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52049 | file | 2007-01-24 13:20:05 -0500 (Wed, 24 Jan 2007) | 2 lines Merge in dialing API and the app_page that uses it. (issue #BE-118) ........ * /, channels/chan_sip.c: Merged revisions 52016 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52016 | file | 2007-01-24 12:59:55 -0500 (Wed, 24 Jan 2007) | 2 lines Fix changing channel formats when joint capability changes and there are no audio formats... I didn't break it originally! (issue #8535 reported by ivoc) ........ 2007-01-24 09:42 +0000 [r51905-51933] Olle Johansson * /, channels/chan_sip.c: Merged revisions 51931 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51931 | oej | 2007-01-24 10:30:21 +0100 (Wed, 24 Jan 2007) | 3 lines Show capabilities *and* preference in general settings in "sip show settings" (reported by Clona/Telio - Thanks!) ........ * include/asterisk/http.h, main/http.c: Doxygen updates * funcs/func_rand.c, funcs/func_base64.c, funcs/func_module.c, funcs/func_md5.c, funcs/func_db.c, funcs/func_version.c, funcs/func_timeout.c, funcs/func_env.c, funcs/func_math.c, funcs/func_strings.c, funcs/func_sha1.c, funcs/func_logic.c, funcs/func_uri.c, funcs/func_global.c, funcs/func_enum.c, funcs/func_groupcount.c, funcs/func_odbc.c, funcs/func_shell.c, funcs/func_channel.c, funcs/func_cdr.c, funcs/func_callerid.c: Doxygen update * main/udptl.c: Adding some doxygen for udptl.c 2007-01-24 01:00 +0000 [r51850] Russell Bryant * main/channel.c, /: Merged revisions 51848 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r51848 | russell | 2007-01-23 18:59:58 -0600 (Tue, 23 Jan 2007) | 14 lines Merged revisions 51843 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51843 | russell | 2007-01-23 18:57:28 -0600 (Tue, 23 Jan 2007) | 6 lines Fix an issue related to synchronization of recordings when using Monitor(). The bug is a miscalculation of the amount to seek the stream for writing to disk when the number of samples coming in and out of a channel do not match up. (issue #8298, #8887, report and patch by guillecabeza, patch files created and testing done by whoiswes) ........ ................ 2007-01-24 00:22 +0000 [r51831] Joshua Colp * main/manager.c: Close file after we do the translation, and map memory for both reading/writing. (issue #8886 reported by cwegener) 2007-01-24 00:21 +0000 [r51830] Russell Bryant * /, apps/app_while.c: Merged revisions 51829 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r51829 | russell | 2007-01-23 18:19:55 -0600 (Tue, 23 Jan 2007) | 12 lines Merged revisions 51828 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51828 | russell | 2007-01-23 18:17:50 -0600 (Tue, 23 Jan 2007) | 4 lines Don't set a new value for the END_ variable on the channel before using the old value. If you do, it will lead to accessing a memory address that has been free()'d. (issue #8895, arkadia) ........ ................ 2007-01-23 22:59 +0000 [r51801] Joshua Colp * channels/chan_phone.c, channels/chan_zap.c, /, channels/chan_sip.c, channels/chan_skinny.c, channels/chan_features.c, channels/chan_alsa.c, channels/chan_gtalk.c, channels/chan_iax2.c, channels/chan_oss.c: Merged revisions 51788 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51788 | file | 2007-01-23 17:46:31 -0500 (Tue, 23 Jan 2007) | 2 lines Update channel drivers to use module referencing so that unloading them while in use will not result in crashes. (issue #8897 reported by junky) ........ 2007-01-23 22:09 +0000 [r51751-51787] Russell Bryant * main/manager.c, /: Merged revisions 51781 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51781 | russell | 2007-01-23 16:04:01 -0600 (Tue, 23 Jan 2007) | 6 lines Fix some bugs in process_message(). The manager session lock needs to be held when sending some sort of response, or calling one of the manager action callbacks. This resolves an issue where people using the GUI would get random crashes when they start clicking around a lot. (issue #8711, reported and debugged by zandbelt) ........ * main/manager.c, /: Merged revisions 51750 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51750 | russell | 2007-01-23 15:33:15 -0600 (Tue, 23 Jan 2007) | 4 lines When traversing the list of manager actions, the iterator needs to be initialized to the list head *after* locking the list. Also, lock the actions list in one place it is being accessed where it was not being done. ........ 2007-01-23 20:36 +0000 [r51684-51717] Steve Murphy * /, res/res_features.c: Merged revisions 51716 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51716 | murf | 2007-01-23 13:32:54 -0700 (Tue, 23 Jan 2007) | 1 line this mod from 8593 (dstchannel in cdr is empty when transfer call). ........ * /, main/callerid.c: Merged revisions 51683 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51683 | murf | 2007-01-23 11:58:27 -0700 (Tue, 23 Jan 2007) | 1 line via 8748 (callerid.c loses name when returning PRIVATE_NUMBER flag), the user suggested this mod, saying it would allow 'WITHHELD' to appear in the name field, which would be useful ........ 2007-01-23 15:36 +0000 [r51659] Olle Johansson * channels/chan_sip.c: Issue #8817 - Registry corruption when packet retransmits fail. (tootai, patchy by oej) 2007-01-23 06:56 +0000 [r51623] Paul Cadach * /, channels/chan_h323.c, channels/Makefile: Merged revisions 51615 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51615 | pcadach | 2007-01-22 22:51:51 -0800 (Пнд, 22 Янв 2007) | 1 line Do not abort Asterisk startup if h323 configuration file not found (reported by mithraen) ........ 2007-01-23 04:45 +0000 [r51463-51592] Joshua Colp * doc/externalivr.txt, apps/app_externalivr.c, CHANGES: Make 'H' command do as advertised and add 'E' and 'V' commands to ExternalIVR. (issue #8165 reported by mnicholson) * configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: Add SRV Lookup support on outbound calls to chan_iax2. It's listed in the RFC so we might want to support it and please don't hurt me Marko ... (issue #7812 reported by drorlb) * /, channels/chan_sip.c: Merged revisions 51558 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51558 | file | 2007-01-22 22:00:12 -0500 (Mon, 22 Jan 2007) | 2 lines Only change audio formats on the channel if we have an audio format to change to. (issue #8535 reported by ivoc) ........ * /: No more conflicts on properties! svnmerge-block be gone! * /, res/res_musiconhold.c: Merged revisions 51513 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r51513 | file | 2007-01-22 20:45:04 -0500 (Mon, 22 Jan 2007) | 10 lines Merged revisions 51512 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51512 | file | 2007-01-22 20:41:35 -0500 (Mon, 22 Jan 2007) | 2 lines Yield before reading from zaptel timing source under Solaris so that other threads get a chance to do things. (issue #7875 reported by bob) ........ ................ * main/autoservice.c: Might as well go crazy here too and make the autoservice list read/write. * main/pbx.c, main/autoservice.c, main/frame.c, main/say.c, main/jitterbuf.c, main/devicestate.c, main/utils.c, main/enum.c, main/fskmodem.c, main/config.c, main/cli.c, main/io.c, main/channel.c, main/cdr.c, main/abstract_jb.c, main/logger.c, main/callerid.c, main/file.c, main/app.c, main/image.c, main/alaw.c, main/asterisk.c, main/dsp.c: Cosmetic changes. Make main source files better conform to coding guidelines and standards. (issue #8679 reported by johann8384) * main/rtp.c: Change RTP protos list to be read/write. Most of the time it's only going to be read so making it use mutex locks was a waste. * main/rtp.c: Make the RTP stack better conform to coding guidelines. (issue #8679 reported by johann8384) 2007-01-22 19:42 +0000 [r51413] Steve Murphy * /, pbx/pbx_ael.c: Merged revisions 51409 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51409 | murf | 2007-01-22 12:28:51 -0700 (Mon, 22 Jan 2007) | 1 line This fixes 8836, according to dnatural ........ 2007-01-22 19:22 +0000 [r51408] Joshua Colp * /, apps/app_mixmonitor.c: Merged revisions 51407 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r51407 | file | 2007-01-22 14:13:44 -0500 (Mon, 22 Jan 2007) | 10 lines Merged revisions 51406 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51406 | file | 2007-01-22 14:08:52 -0500 (Mon, 22 Jan 2007) | 2 lines Move filestream creation to Mixmonitor loop. This will prevent a blank file from being created if no frames ever pass through to be recorded. (issue #7589 reported by steve_mcneil) ........ ................ 2007-01-22 19:00 +0000 [r51405] Olle Johansson * channels/chan_sip.c: Remove (to quote Rizzo) "useless" variable. 2007-01-21 03:25 +0000 [r51353] Tilghman Lesher * main/pbx.c: Fix bug introduced during constification (reported by tzanger via IRC) 2007-01-20 18:27 +0000 [r51352] Russell Bryant * include/asterisk/frame.h: Add a comment that the frame type constants are transmitted directly over IAX2. 2007-01-20 06:54 +0000 [r51349-51351] Jason Parker * /, configs/say.conf.sample: Merged revisions 51350 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51350 | qwell | 2007-01-20 00:53:49 -0600 (Sat, 20 Jan 2007) | 5 lines Fix Italian numeral support in say.conf for "_[2-9]00" case. "2131" would've translated to something along the lines of (pardon my..Italian {or lack thereof}) "duecentocentotrentuno", which makes no sense at all. ........ * /, configs/say.conf.sample: Merged revisions 51348 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51348 | qwell | 2007-01-20 00:16:06 -0600 (Sat, 20 Jan 2007) | 8 lines Fix German language support in say.conf Properly support 21, 31, 41, 51, 61, 71, 81, and 91. einundzwanzig has the same format as zweiundzwanzig (as do all other "_ZX" spoken numerals) Fix support for numbers in the 10,000,000 to 99,999,999 range. Add support for numbers in the 100,000,000 to 999,999,999 range. ........ 2007-01-20 00:13 +0000 [r51314-51344] Russell Bryant * /, apps/app_meetme.c: Merged revisions 51343 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51343 | russell | 2007-01-19 18:13:06 -0600 (Fri, 19 Jan 2007) | 2 lines Remove an unused instance of an unnamed enum. ........ * /, apps/app_meetme.c: Merged revisions 51341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51341 | russell | 2007-01-19 16:19:10 -0600 (Fri, 19 Jan 2007) | 2 lines Remove another duplicated definition ........ * /, apps/app_meetme.c: Merged revisions 51339 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51339 | russell | 2007-01-19 15:20:20 -0600 (Fri, 19 Jan 2007) | 2 lines Remove a variable that was declared twice. ........ * /, codecs/gsm/Makefile: Merged revisions 51331 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51331 | russell | 2007-01-19 13:30:54 -0600 (Fri, 19 Jan 2007) | 3 lines Add a couple more processors that need optimizations excluded. (issue #8637) ........ * /, channels/chan_gtalk.c: Merged revisions 51328 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51328 | russell | 2007-01-19 13:08:25 -0600 (Fri, 19 Jan 2007) | 5 lines Fix VLDTMF support in chan_gtalk. AST_FRAME_DTMF and AST_FRAME_DTMF_END are actually the same thing. So, a digit would have been interpreted incorrectly here. Since the channel driver will always have the begin and end callbacks called for a digit, only support the button-down and button-up messages. ........ * /, .cleancount: Merged revisions 51326 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51326 | russell | 2007-01-19 13:02:55 -0600 (Fri, 19 Jan 2007) | 2 lines Bump the cleancount since my last commit changed the channel structure. ........ * channels/chan_zap.c, channels/chan_local.c, main/frame.c, /, channels/chan_sip.c, channels/chan_agent.c, include/asterisk/channel.h, channels/chan_gtalk.c, channels/chan_iax2.c, channels/chan_oss.c, main/rtp.c, main/channel.c, channels/chan_jingle.c, channels/chan_phone.c, channels/chan_misdn.c, channels/chan_skinny.c, channels/chan_features.c, channels/chan_h323.c, channels/chan_alsa.c, channels/chan_mgcp.c: Merged revisions 51311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines Merge the changes from the /team/group/vldtmf_fixup branch. The main bug being addressed here is a problem introduced when two SIP channels using SIP INFO dtmf have their media directly bridged. So, when a DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk would try to emulate a digit of some length by first sending a DTMF BEGIN frame and sending a DTMF END later timed off of incoming audio. However, since there was no audio coming in, the DTMF_END was never generated. This caused DTMF based features to no longer work. To fix this, the core now knows when a channel doesn't care about DTMF BEGIN frames (such as a SIP channel sending INFO dtmf). If this is the case, then Asterisk will not emulate a digit of some length, and will instead just pass through the single DTMF END event. Channel drivers also now get passed the length of the digit to their digit_end callback. This improves SIP INFO support even further by enabling us to put the real digit duration in the INFO message instead of a hard coded 250ms. Also, for an incoming INFO message, the duration is read from the frame and passed into the core instead of just getting ignored. (issue #8597, maybe others...) ........ 2007-01-19 18:00 +0000 [r51308-51312] Luigi Rizzo * include/asterisk/strings.h: As the comment in the diff says: AST_INLINE_API() is a macro that takes a block of code as an argument. Using preprocessor #directives in the argument is not supported by all compilers, and it is a bit of an obfuscation anyways, so avoid it. As a workaround, define a macro that produces either its argument or nothing, and use that instead of #ifdef/#endif within the argument to AST_INLINE_API(). * main/rtp.c: in the interest of portability, avoid using %zd when all we need is to print is an integer that fits in 16 bits. * channels/chan_iax2.c: sizeof() is compatible with format %d so don't be too picky on printf formats. * channels/chan_zap.c: remove variable declaration in the middle of a block 2007-01-19 17:19 +0000 [r51303-51305] Russell Bryant * configure, include/asterisk/autoconfig.h.in: Regenerate configure script to reflect recent zaptel changes * include/asterisk/zapata.h: Include tonezone.h for linux, too * main/asterisk.c: Merged revisions 51302 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r51302 | russell | 2007-01-19 10:56:17 -0600 (Fri, 19 Jan 2007) | 12 lines Merged revisions 51300 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51300 | russell | 2007-01-19 10:44:09 -0600 (Fri, 19 Jan 2007) | 4 lines Fix a memory leak on command line tab completion. The container for the matches was freed, but the individual matches themselves were not. (issue #8851, arkadia) ........ ................ 2007-01-19 16:51 +0000 [r51297-51301] Luigi Rizzo * main/Makefile: forgot to add AST_LIBS += $(BKTR_LIB) * main/channel.c: include "asterisk/zapata.h" to get the zaptel headers. this should be the last one left around... * channels/chan_zap.c: whoops, fix a cut&paste error... * channels/chan_zap.c: slight change to the initialization of a structure, also using '\0' to make it clear we need a (char)0 2007-01-19 16:30 +0000 [r51296] Russell Bryant * main/manager.c: Break out of the config processing loop for manager.conf once the correct user has been found so that 'cat' is non-NULL. This way, users.conf is only checked when necessary. (issue #8852, akohlsmith, committed patch a bit different) 2007-01-19 16:28 +0000 [r51285-51295] Luigi Rizzo * channels/chan_zap.c: include "asterisk/zapata.h" to get the zaptel headers. * codecs/codec_zap.c: include "asterisk/zapata.h" to get the zaptel headers * apps/app_meetme.c: include "asterisk/zapata.h" instead of testing for the location of the header files. On passing, add a cast to insure -Werror clean compilation on FreeBSD 6.x, where time_t does not match %ld * apps/app_zapbarge.c, apps/app_flash.c, apps/app_zapscan.c, apps/app_zapras.c, res/res_musiconhold.c, channels/chan_iax2.c, apps/app_rpt.c: include "asterisk/zapata.h" instead of looking directly for the zaptel.h and tonezone.h * configure.ac: another freebsd-specific check for zaptel compatibility * include/asterisk/zapata.h (added): Add a stub file to find the zaptel headers in the right place, rather than repeating the check on every single file. Changes to the individual files are coming. The header file name has been suggested by kevin. Approved by: kpfleming * makeopts.in: forgot to add BKTR_INCLUDE and BKTR_LIB in makeopts.in * configure.ac: add comments that AC_USE_SYSTEM_EXTENSIONS and AST_PROG_LD do not work on FreeBSD - presumably they depend on some auto* feature that is not installed by default. I am not sure on what is a proper fix. In my local copy i simply comment them out. The AST_PROG_LD is a long standing isse, there were attempts to fix it in the past but probably not enough has been copied to acinclude.m4, and i had forgotten about it because i commented out this call in configure.ac long ago * configure.ac: Add check for backtrace support on platforms that do not have it natively. Part of it leaked in in a previous commit. * configure.ac: remove a useless (and harmful on some platforms) -lnsl from IKSEMEL_LIB. Actually i am not even sure whether -lgcrypt -lgpg-error are needed. * configure.ac: simplify checking for zaptel version and location (for linux, this is functionally equivalent to the previous method; for FreeBSD, it re-adds inspection in $PREFIX/zaptel.h). Please wait to regenerate the "configure" file as i have another few pending changes to configure.ac Not applicable to 1.4 until acinclude.m4 is also updated. 2007-01-19 00:28 +0000 [r51273-51275] Dwayne M. Hubbard * channels/chan_zap.c, /: Merged revisions 51274 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51274 | dhubbard | 2007-01-18 18:17:32 -0600 (Thu, 18 Jan 2007) | 3 lines chan_zap compiles without libpri after committing 7877 patch ........ * channels/chan_zap.c, /: Merged revisions 51272 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r51272 | dhubbard | 2007-01-18 17:56:49 -0600 (Thu, 18 Jan 2007) | 11 lines Merged revisions 51271 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51271 | dhubbard | 2007-01-18 17:47:10 -0600 (Thu, 18 Jan 2007) | 3 lines issue 7877: chan_zap module reload does not use default/initialized values on subsequent loads. Reset configuration variables to default values prior to parsing configuration file. ........ ................ 2007-01-18 22:56 +0000 [r51266] Jason Parker * main/pbx.c, /, funcs/func_strings.c, apps/app_voicemail.c: Merged revisions 51265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51265 | qwell | 2007-01-18 16:50:23 -0600 (Thu, 18 Jan 2007) | 4 lines Add some more checks for option_debug before ast_log(LOG_DEBUG, ...) calls. Issue 8832, patch(es) by tgrman ........ 2007-01-18 21:57 +0000 [r51263] Russell Bryant * Makefile, /, configure, main/Makefile, acinclude.m4, makeopts.in: Merged revisions 51262 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51262 | russell | 2007-01-18 15:54:23 -0600 (Thu, 18 Jan 2007) | 5 lines Ensure that the locations given to the Asterisk configure script for ncurses, curses, termcap, or tinfo are further passed along to the editline configure script. This fixes some cross-compilation environments. (issue #8637, reported by ovi, patch by me) ........ 2007-01-18 21:15 +0000 [r51257] Tilghman Lesher * /, main/stdtime/localtime.c: Merged revisions 51256 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r51256 | tilghman | 2007-01-18 15:14:24 -0600 (Thu, 18 Jan 2007) | 10 lines Merged revisions 51255 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51255 | tilghman | 2007-01-18 15:11:34 -0600 (Thu, 18 Jan 2007) | 2 lines If a timezone is not specified, assume localtime (instead of gmtime) (Issue #7748) ........ ................ 2007-01-18 19:19 +0000 [r51252] Joshua Colp * /, apps/app_speech_utils.c: Merged revisions 51251 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51251 | file | 2007-01-18 14:17:34 -0500 (Thu, 18 Jan 2007) | 2 lines Only start timeout once we reach the end of the files to play back. ........ 2007-01-18 19:03 +0000 [r51249] Jason Parker * main/cli.c: Fix filename completion for "module load" and "load" CLI commands. Issue 8846 2007-01-18 18:54 +0000 [r51247] Russell Bryant * main/manager.c: Fix trunk version of manager support for users.conf. Now it actually pays attention to the "hasmanager" option. (Thanks to Anthony L. for pointing out that this was broken!) 2007-01-18 18:39 +0000 [r51244] Joshua Colp * /, channels/chan_sip.c: Merged revisions 51243 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51243 | file | 2007-01-18 13:36:35 -0500 (Thu, 18 Jan 2007) | 2 lines Copy MOH settings when calling a peer so that if they put someone on hold or get put on hold themselves they get the right music class. (issue #8840 reported by mdu113) ........ 2007-01-18 18:36 +0000 [r51242] Jason Parker * main/channel.c, /: Merged revisions 51241 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51241 | qwell | 2007-01-18 12:28:29 -0600 (Thu, 18 Jan 2007) | 2 lines Fix an issue with deprecated commands ........ 2007-01-18 17:52 +0000 [r51237] Tilghman Lesher * contrib/scripts/vmdb.sql, /: Merged revisions 51236 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r51236 | tilghman | 2007-01-18 11:49:41 -0600 (Thu, 18 Jan 2007) | 10 lines Merged revisions 51235 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51235 | tilghman | 2007-01-18 11:42:17 -0600 (Thu, 18 Jan 2007) | 2 lines Document all the fields, including the indication that "uniqueid" should not be renamed. ........ ................ 2007-01-18 17:33 +0000 [r51234] Russell Bryant * main/manager.c, /: Merged revisions 51233 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51233 | russell | 2007-01-18 11:18:43 -0600 (Thu, 18 Jan 2007) | 3 lines Make the "hasmanager" option in users.conf actually have an effect. (issue #8740, LnxPrgr3) ........ 2007-01-18 06:59 +0000 [r51221] Paul Cadach * channels/chan_h323.c: Update ast_append_ha() usage 2007-01-18 05:24 +0000 [r51212-51215] Joshua Colp * apps/app_page.c, CHANGES: Add 's' option to Page application which checks devicestate before dialing. (issue #8673 reported by sunder) * /, apps/app_voicemail.c: Merged revisions 51213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51213 | file | 2007-01-17 19:48:55 -0500 (Wed, 17 Jan 2007) | 2 lines Build the IMAP remote directory string better and properly. Fix an issue with encoding the GSM voicemail when attaching to the voicemail. (issue #8808 reported by akohlsmith) ........ * /, main/rtp.c: Merged revisions 51211 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51211 | file | 2007-01-17 19:18:44 -0500 (Wed, 17 Jan 2007) | 2 lines Pass data as well for hold/unhold/vidupdate frames. (issue #8840 reported by mdu113) ........ 2007-01-17 23:35 +0000 [r51199-51207] Russell Bryant * /, funcs/func_odbc.c: Merged revisions 51205 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51205 | russell | 2007-01-17 17:31:11 -0600 (Wed, 17 Jan 2007) | 5 lines Fix some instances where when loading func_odbc, a double-free could occur. Also, remove an unneeded error message. If the failure condition is actually a memory allocation failure, a log message will already be generated automatically. ........ * channels/chan_zap.c, /: Merged revisions 51204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51204 | russell | 2007-01-17 16:09:52 -0600 (Wed, 17 Jan 2007) | 4 lines Instead of dividing the offset by 2 directly, make it more clear that the offset is being scaled by the size of the elements in the buffer. (Inspired by a discussing on the asterisk-dev list about this code) ........ * /, channels/chan_sip.c: Merged revisions 51198 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r51198 | russell | 2007-01-17 15:18:35 -0600 (Wed, 17 Jan 2007) | 11 lines Merged revisions 51197 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51197 | russell | 2007-01-17 15:17:21 -0600 (Wed, 17 Jan 2007) | 3 lines Move the check for a failure of ast_channel_alloc() to before locking the pvt structure again. Otherwise, on a failure, this will cause a deadlock. ........ ................ 2007-01-17 20:57 +0000 [r51196] Tilghman Lesher * /, main/utils.c: Merged revisions 51195 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r51195 | tilghman | 2007-01-17 14:56:15 -0600 (Wed, 17 Jan 2007) | 12 lines Merged revisions 51194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51194 | tilghman | 2007-01-17 14:52:21 -0600 (Wed, 17 Jan 2007) | 4 lines When ast_strip_quoted was called with a zero-length string, it would treat a NULL as if it were the quoting character (and would thus return the string in memory immediately following the passed-in string). ........ ................ 2007-01-17 19:43 +0000 [r51193] Joshua Colp * main/channel.c: Don't hold channel lock while sleeping/waiting for audio stream to get setup. (issue #8834 reported by phsultan) 2007-01-17 17:37 +0000 [r51189] Jason Parker * /, apps/app_voicemail.c: Merged revisions 51186 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51186 | qwell | 2007-01-17 11:36:53 -0600 (Wed, 17 Jan 2007) | 2 lines re-add "password" for realtime voicemail ........ 2007-01-17 06:37 +0000 [r51183] Joshua Colp * /, main/rtp.c: Merged revisions 51182 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51182 | file | 2007-01-17 01:36:41 -0500 (Wed, 17 Jan 2007) | 2 lines Return the correct result when directly writing out a packet so that the core doesn't then decide to handle it the regular way again. (issue #8833 reported by rcourtna) ........ 2007-01-17 01:30 +0000 [r51177] Kevin P. Fleming * /, apps/app_voicemail.c: Merged revisions 51176 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51176 | kpfleming | 2007-01-16 19:29:12 -0600 (Tue, 16 Jan 2007) | 2 lines a few more coding style cleanups and one bug fix (from AnthonyL) ........ 2007-01-17 00:50 +0000 [r51173] Joshua Colp * /, channels/chan_iax2.c: Merged revisions 51172 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51172 | file | 2007-01-16 19:46:29 -0500 (Tue, 16 Jan 2007) | 2 lines Move rescheduling of lagrq/pings into the scheduler callback. ........ 2007-01-17 00:22 +0000 [r51166-51171] Jason Parker * /, main/rtp.c: Merged revisions 51170 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51170 | qwell | 2007-01-16 18:20:56 -0600 (Tue, 16 Jan 2007) | 4 lines Fix issue with dtmf continuation packets when the dtmf digit is 0... Issue 8831 ........ * contrib/scripts/vmdb.sql, /, apps/app_voicemail.c: Merged revisions 51167 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51167 | qwell | 2007-01-16 16:50:19 -0600 (Tue, 16 Jan 2007) | 6 lines Fix an issue with IMAP storage and realtime voicemail. Also update the vmdb sql script for IMAP specific options. Issue 8819, initial patches by bsmithurst (slightly modified by me) ........ * /, doc/voicemail_odbc_postgresql.txt: Merged revisions 51165 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51165 | qwell | 2007-01-16 16:07:53 -0600 (Tue, 16 Jan 2007) | 2 lines change documentation to reflect new procedure in 1.4/trunk ........ 2007-01-16 21:52 +0000 [r51160-51163] Tilghman Lesher * /, doc/voicemail_odbc_postgresql.txt (added): Merged revisions 51162 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r51162 | tilghman | 2007-01-16 15:51:15 -0600 (Tue, 16 Jan 2007) | 10 lines Merged revisions 51161 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51161 | tilghman | 2007-01-16 15:50:04 -0600 (Tue, 16 Jan 2007) | 2 lines Add documentation walkthrough on getting Postgres to work with voicemail (from Issue 8513) ........ ................ * /, apps/app_voicemail.c: Merged revisions 51159 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r51159 | tilghman | 2007-01-16 15:28:39 -0600 (Tue, 16 Jan 2007) | 10 lines Merged revisions 51158 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51158 | tilghman | 2007-01-16 15:26:06 -0600 (Tue, 16 Jan 2007) | 2 lines Postgres driver doesn't like a NULL pointer when retrieving the length (Bug 8513) ........ ................ 2007-01-16 19:01 +0000 [r51155] Kevin P. Fleming * apps/app_voicemail.c: remove pointless DEBUG message (watch those patch merges, people!) 2007-01-16 17:50 +0000 [r51152] Joshua Colp * res/res_features.c, CHANGES, configs/features.conf.sample: Add parkedcalltransfers option for res_features. This basically enables/disables DTMF based transfers. If you want to get former behavior you will have to make sure it is enabled. 2007-01-16 17:47 +0000 [r51151] Matt O'Gorman * /, apps/app_voicemail.c: Merged revisions 51150 via svnmerge from https://svn.digium.com/svn/asterisk/branches/1.4 ........ r51150 | mogorman | 2007-01-16 11:46:12 -0600 (Tue, 16 Jan 2007) | 2 lines minor things i missed before i get jumped on ........ 2007-01-16 17:42 +0000 [r51149] Joshua Colp * /, res/res_features.c: Merged revisions 51148 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r51148 | file | 2007-01-16 12:39:50 -0500 (Tue, 16 Jan 2007) | 10 lines Merged revisions 51145 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51145 | file | 2007-01-16 12:36:50 -0500 (Tue, 16 Jan 2007) | 2 lines Return previous behavior. ParkedCalls will be able to do DTMF based transfers again. trunk however will get an option to allow this to be set on/off. (issue #8804 reported by nortex) ........ ................ 2007-01-16 17:39 +0000 [r51147] Jason Parker * /, main/file.c: Merged revisions 51146 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51146 | qwell | 2007-01-16 11:36:53 -0600 (Tue, 16 Jan 2007) | 6 lines Display more useful output when streaming files. Include the channel name to which the file is being played. Issue 8828, patch by junky. ........ 2007-01-16 17:23 +0000 [r51144] Joshua Colp * channels/chan_phone.c, configs/phone.conf.sample, CHANGES: Add support for G729 passthrough with Sigma Designs boards. (issue #8829 reported by ywalther) 2007-01-16 08:38 +0000 [r51123] Tilghman Lesher * channels/iax2-parser.h, channels/iax2.h, channels/chan_iax2.c, channels/iax2-parser.c: IAX2 remote variables - Bug 7619 2007-01-16 05:56 +0000 [r51090] Joshua Colp * channels/chan_zap.c, /: Merged revisions 51087 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r51087 | file | 2007-01-16 00:55:23 -0500 (Tue, 16 Jan 2007) | 10 lines Merged revisions 51085 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51085 | file | 2007-01-16 00:53:31 -0500 (Tue, 16 Jan 2007) | 2 lines Add none as a valid callgroup/pickupgroup option. I consider it a bug that it would inherit it all the way down and not have any way to reset it to nothing - so that's why it is in 1.2. (issue #8296 reported by gkloepfer) ........ ................ 2007-01-16 01:20 +0000 [r51058-51060] Russell Bryant * configs/osp.conf.sample: Fix a couple of typos in the sample osp.conf. * /, main/config.c: Merged revisions 51057 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51057 | russell | 2007-01-15 19:15:44 -0600 (Mon, 15 Jan 2007) | 3 lines It is possible for the config pointer to be NULL here, so it needs to be checked before dereferencing it. ........ 2007-01-16 00:29 +0000 [r51031] Matt O'Gorman * configs/users.conf.sample, /, apps/app_voicemail.c: Patch allows for changing voicemail password in users.conf from voicemail main, written by AnthonyL bug #8436 2007-01-15 23:51 +0000 [r50995] Russell Bryant * /, Makefile.rules: Merged revisions 50994 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r50994 | russell | 2007-01-15 17:49:48 -0600 (Mon, 15 Jan 2007) | 2 lines Filter out a few CFLAGS that are not valid CXXFLAGS. ........ 2007-01-15 21:12 +0000 [r50958] Matt O'Gorman * /, apps/app_voicemail.c: Merged revisions 50957 via svnmerge from https://svn.digium.com/svn/asterisk/branches/1.4 ................ r50957 | mogorman | 2007-01-15 15:08:07 -0600 (Mon, 15 Jan 2007) | 12 lines Merged revisions 50946 via svnmerge from https://svn.digium.com/svn/asterisk/branches/1.2 ........ r50946 | mogorman | 2007-01-15 14:44:53 -0600 (Mon, 15 Jan 2007) | 4 lines Solves issue with forwarding voicemails from folders other than inbox. patch by anthonyl. ........ ................ 2007-01-15 18:24 +0000 [r50922] Jason Parker * /: These deprecated functions were removed in trunk on purpose. No need to re-add. 2007-01-15 16:40 +0000 [r50896] Joshua Colp * main/manager.c, /: Merged revisions 50895 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r50895 | file | 2007-01-15 11:36:07 -0500 (Mon, 15 Jan 2007) | 2 lines Move event processing into do_message so that it gets executed again when events are tripped. ........ 2007-01-15 15:08 +0000 [r50868-50869] Kevin P. Fleming * /, configure, include/asterisk/autoconfig.h.in, main/Makefile, configure.ac, Makefile.rules, acinclude.m4, makeopts.in: Merged revisions 50867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r50867 | kpfleming | 2007-01-15 09:03:06 -0600 (Mon, 15 Jan 2007) | 2 lines use the ACX_PTHREAD macro from the Autoconf macro archive for setting up compiler pthreads support... should improve portability to platforms with unusual pthreads requirements ........ * codecs/g722: ignore dependency files in this directory 2007-01-15 02:28 +0000 [r50847] Tilghman Lesher * channels/chan_oss.c: Feature: allow soundcard to be used in both modes (autoanswer and not), selectable by how it is called in the dialplan. This allows a speaker system hooked up to the soundcard to be used for both ring notification, as well as paging. 2007-01-14 22:00 +0000 [r50821] Joshua Colp * /, main/astmm.c: Merged revisions 50820 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r50820 | file | 2007-01-14 16:59:05 -0500 (Sun, 14 Jan 2007) | 2 lines Add missing newlines for two memory CLI commands. ........ 2007-01-14 05:34 +0000 [r50783-50784] Tilghman Lesher * main/config.c: Bug 8803 - Fix crash in API * /, main/db1-ast/hash/hsearch.c, main/db1-ast/btree/bt_page.c, main/db1-ast/recno/rec_search.c, main/db1-ast/btree/bt_get.c, main/db1-ast/hash/hash.c, main/db1-ast/db/db.c, main/db1-ast/recno/rec_get.c, main/db1-ast/btree/bt_seq.c, main/db1-ast/hash/hash_func.c, main/db1-ast/btree/bt_utils.c, main/db1-ast/recno/rec_seq.c, main/db1-ast/btree/bt_overflow.c, main/db1-ast/btree/bt_search.c, main/db1-ast/btree/bt_conv.c, main/db1-ast/btree/bt_close.c, main/db1-ast/btree/bt_put.c, main/db1-ast/recno/rec_utils.c, main/db1-ast/hash/hash_bigkey.c, main/db1-ast/recno/rec_open.c, main/db1-ast/recno/rec_delete.c, main/db1-ast/hash/hash_buf.c, main/db1-ast/hash/hash_page.c, main/db1-ast/recno/rec_close.c, main/db1-ast/recno/rec_put.c, main/db1-ast/include/ndbm.h, main/db1-ast/btree/bt_debug.c, main/db1-ast/mpool/mpool.c, main/db1-ast/btree/bt_split.c, main/db1-ast/btree/bt_open.c, main/db1-ast/btree/bt_delete.c, main/db1-ast/hash/hash_log2.c: Merged revisions 50782 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r50782 | tilghman | 2007-01-13 23:13:47 -0600 (Sat, 13 Jan 2007) | 10 lines Merged revisions 50781 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r50781 | tilghman | 2007-01-13 23:01:16 -0600 (Sat, 13 Jan 2007) | 2 lines Bug 8814 - db should look for its header using a relative path, instead of the system path (Fixes FreeWRT) ........ ................ 2007-01-13 16:47 +0000 [r50755] Kevin P. Fleming * Makefile, /, build_tools/make_sample_voicemail (added): Merged revisions 50754 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r50754 | kpfleming | 2007-01-13 10:45:37 -0600 (Sat, 13 Jan 2007) | 2 lines when building the sample greetings for maibox 1234@default during 'make samples', build a greeting for each language and file format the user selected to install with menuselect (reported by Brian Capouch on asterisk-dev) ........ 2007-01-13 06:01 +0000 [r50675-50728] Joshua Colp * main/channel.c, /: Merged revisions 50727 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r50727 | file | 2007-01-13 01:00:24 -0500 (Sat, 13 Jan 2007) | 2 lines Only write a frame out to the channel if one exists. There are cases where one may not and would therefore cause the channel driver to segfault. (issue #8434 reported by slimey) ........ * channels/chan_sip.c: Get rid of unneeded code, fix a spelling mistake, and use registry state a bit more. (issue #8402 reported by rizzo) * configs/iax.conf.sample: Clarify what the trunkmaxsize value is in (bytes). * configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: Drop trunkrealloc option and just have the maximum size be a configurable option. This is per Kevin's comments on -dev and my own thoughts after I put the previous option in. * channels/chan_sip.c: Ensure error variable is set to 0 or else we might get false error messages. (issue #8798 reported by tootai, fix by anthonyl) * configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: Merge in trunkrealloc option for chan_iax2. (issue #8267 reported by marcodmb, branch by anthonyl) * /, res/res_snmp.c: Merged revisions 50674 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r50674 | file | 2007-01-12 22:04:55 -0500 (Fri, 12 Jan 2007) | 2 lines Only join the snmp thread on an unload if the thread is actually running. (issue #8810 reported by junky) ........ 2007-01-12 19:25 +0000 [r50648] Jason Parker * /, configs/voicemail.conf.sample: Merged revisions 50647 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r50647 | qwell | 2007-01-12 13:24:40 -0600 (Fri, 12 Jan 2007) | 2 lines Update documentation to state that you shouldn't use realtime static with voicemail.conf ........ 2007-01-12 18:13 +0000 [r50603-50629] Joshua Colp * main/manager.c: Exit from session loop upon error (ie: they disconnected) and don't do any buffer manipulation in do_message. get_input will handle it. * main/manager.c, /: Merged revisions 50602 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r50602 | file | 2007-01-12 11:42:33 -0500 (Fri, 12 Jan 2007) | 2 lines We need to check for res being 0 in do_message itself, otherwise our headers will get lost. ........ 2007-01-12 15:01 +0000 [r50538-50571] Kevin P. Fleming * main/channel.c, main/pbx.c, include/asterisk/channel.h: make the automatic post-answer delay happen only when the answer is 'automatic' (not done by the Answer() dialplan application) * main/pbx.c, /: Merged revisions 50562 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r50562 | kpfleming | 2007-01-12 08:42:24 -0600 (Fri, 12 Jan 2007) | 10 lines Merged revisions 50561 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r50561 | kpfleming | 2007-01-12 08:34:15 -0600 (Fri, 12 Jan 2007) | 2 lines minor documentation clarification ........ ................ * main/channel.c: when a channel gets automatically answered by an application, sleep a bit to give the audio path (for VOIP channels) time to be setup 2007-01-11 05:54 +0000 [r50378-50469] Joshua Colp * /, channels/chan_sip.c: Merged revisions 50468 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r50468 | file | 2007-01-11 00:53:09 -0500 (Thu, 11 Jan 2007) | 2 lines Remove check for channel state as it can definitely be something other then ring, and also clean up the code a bit. This should solve the parking issues and maybe some attended transfer issues people have been seeing. ........ * /, channels/chan_sip.c, include/asterisk/rtp.h, main/rtp.c: Merged revisions 50466 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r50466 | file | 2007-01-11 00:19:39 -0500 (Thu, 11 Jan 2007) | 2 lines Add support to see whether NAT was detected (yay symmetric RTP) and also add a check in chan_sip so that if NAT has been detected and the reinvite behind nat option has been turned off, then just do partial bridge. (issue #8655 reported by mnicholson) ........ * /, apps/app_speech_utils.c: Merged revisions 50433 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r50433 | file | 2007-01-10 15:25:44 -0500 (Wed, 10 Jan 2007) | 2 lines Merge speech-multi branch which adds support for joining multiple sound files together to be played one after another in SpeechBackground. ........ * /, main/config.c: Merged revisions 50405 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r50405 | file | 2007-01-10 14:46:29 -0500 (Wed, 10 Jan 2007) | 2 lines Fix parsing when using something like ldap settings. (done by anthonyl) ........ * include/asterisk/strings.h: Return the useless casts that ensure this file is C++ clean. (issue #8602 reported by mikma) * /, channels/chan_sip.c: Merged revisions 50377 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r50377 | file | 2007-01-10 13:32:29 -0500 (Wed, 10 Jan 2007) | 2 lines Fix chan_sip not working issue. Let's not prematurely return 0. (issue #8783 reported by st41ker) ........ 2007-01-10 16:47 +0000 [r50347] Jason Parker * /, cdr/cdr_manager.c: Merged revisions 50346 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r50346 | qwell | 2007-01-10 10:45:36 -0600 (Wed, 10 Jan 2007) | 4 lines Reverse some logic in cdr_manager, which made it fail to load if the config file existed. Issue 8777 ........ 2007-01-10 04:56 +0000 [r50267-50302] Joshua Colp * apps/app_dial.c, /: Merged revisions 50298 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r50298 | file | 2007-01-09 23:55:13 -0500 (Tue, 09 Jan 2007) | 10 lines Merged revisions 50295 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r50295 | file | 2007-01-09 23:51:06 -0500 (Tue, 09 Jan 2007) | 2 lines Add another return value to dial_exec_full that indicates execution is going to continuing at a new extension/context/priority and to just let it slide. (issue #8598 reported by jon) ........ ................ * channels/chan_zap.c: Allow usedistinctiveringdetection and distinctiveringaftercid to be reset during a reload. (issue #8739 reported by tzafrir) * main/pbx.c, /: Merged revisions 50266 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r50266 | file | 2007-01-09 22:51:29 -0500 (Tue, 09 Jan 2007) | 2 lines Ensure data's existence before trying to access it. (issue #8774 reported by rcourtna) ........ 2007-01-10 02:50 +0000 [r50229-50230] Russell Bryant * channels/chan_iax2.c: Covert some spaces to tabs, and put a list of defines in an enum. * Makefile, /: Merged revisions 50228 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r50228 | russell | 2007-01-09 21:17:46 -0500 (Tue, 09 Jan 2007) | 14 lines Merged revisions 50227 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r50227 | russell | 2007-01-09 21:16:45 -0500 (Tue, 09 Jan 2007) | 6 lines Make the number that represents the major version number a single digit instead of 2. Using two digits makes it an octal number when put into version.h, which breaks the compilation of any out of tree module that checks the version for any version after 1.2.7 (reported by Matteo Brancaleoni on the asterisk-dev mailing list, who gave credit to vihai for pointing it out) ........ ................ 2007-01-09 13:45 +0000 [r50152] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 50151 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r50151 | tilghman | 2007-01-09 07:40:45 -0600 (Tue, 09 Jan 2007) | 12 lines Merged revisions 50150 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r50150 | tilghman | 2007-01-09 07:30:04 -0600 (Tue, 09 Jan 2007) | 4 lines The advent of realtime has enabled people to use commas in the fullname field. This could cause an issue with sending voicemails, when the field is unquoted. (Issue 8595) ........ ................ 2007-01-09 12:25 +0000 [r50132] Olle Johansson * /, channels/chan_sip.c: Based on the following patch, changed for trunk... Merged revisions 50124 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r50124 | oej | 2007-01-09 12:25:20 +0100 (Tue, 09 Jan 2007) | 3 lines - handle re-invites properly in sip_hangup() - Add some invitestate status changes just to be sure ........ 2007-01-08 23:40 +0000 [r50099] Jason Parker * /, apps/app_voicemail.c: Merged revisions 50098 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r50098 | qwell | 2007-01-08 17:39:12 -0600 (Mon, 08 Jan 2007) | 4 lines Fix an issue with voicemail and users.conf, where it wouldn't ever parse a password, since it was using "secret" instead of "password" Issue 8761, reported by and patch suggestion from ssokol. ........ 2007-01-08 21:40 +0000 [r50075] Joshua Colp * codecs/codec_zap.c: Move channel acquisition to when the translation path is setup, and clean up. 2007-01-08 21:17 +0000 [r50074] Matt O'Gorman * /, apps/app_senddtmf.c: Merged revisions 50073 via svnmerge from https://svn.digium.com/svn/asterisk/branches/1.4 ........ r50073 | mogorman | 2007-01-08 15:11:16 -0600 (Mon, 08 Jan 2007) | 1 line we can't unlock a channel if we cant find it. - AnthonyL bug #8741 ........ 2007-01-08 20:10 +0000 [r50033-50056] Joshua Colp * main/rtp.c: Make callback declaration match one used in trunk. * include/asterisk/lock.h: Change trylock output for what already has the lock from an error to a warning. * /, main/rtp.c: Merged revisions 50032 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r50032 | file | 2007-01-08 13:21:31 -0500 (Mon, 08 Jan 2007) | 2 lines Disable the more intense packet2packet bridging until the bugs can be worked out. ........ 2007-01-08 14:31 +0000 [r49931-50007] Olle Johansson * /, channels/chan_sip.c: Merged revisions 50006 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r50006 | oej | 2007-01-08 15:26:14 +0100 (Mon, 08 Jan 2007) | 11 lines Issue #8677 - Handle failure of T.38 re-invite This is not a fix, but adding an error message to tell the admin that we have a bad configuration. We should not send T.38 re-invites to devices that can't handle it (with the current architecture where you have to hard-code t.38 support per device). To really fix this, we need to figure out a way to tell the incoming call that the re-invite failed, so we can signal failure on that end and go back to the original call. ........ * /, channels/chan_sip.c: Merged revisions 49983 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49983 | oej | 2007-01-08 14:28:18 +0100 (Mon, 08 Jan 2007) | 3 lines Issue #8524, support multiple via header values (tardieu) Thanks! ........ * main/frame.c, include/asterisk/frame.h, main/rtp.c: Issue #8663 - Add passthrough support for MPEG4 (neutrino88). * /, channels/chan_sip.c: Merged revisions 49945 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49945 | oej | 2007-01-08 10:08:10 +0100 (Mon, 08 Jan 2007) | 2 lines We only need one forward declaration ........ * /, channels/chan_sip.c: Merged revisions 49925 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49925 | oej | 2007-01-08 09:55:03 +0100 (Mon, 08 Jan 2007) | 2 lines Issue 8735: Terminate state when extension is unavailable for subscription ........ 2007-01-08 05:13 +0000 [r49891] Joshua Colp * /, channels/chan_iax2.c: Merged revisions 49890 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r49890 | file | 2007-01-08 00:11:54 -0500 (Mon, 08 Jan 2007) | 10 lines Merged revisions 49889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r49889 | file | 2007-01-08 00:10:07 -0500 (Mon, 08 Jan 2007) | 2 lines Ensure we use the default refresh value of 60 if the remote server does not send one. (issue #8746 reported by maethor) ........ ................ 2007-01-08 03:56 +0000 [r49870] Kevin P. Fleming * /, configure, configure.ac: Merged revisions 49866 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49866 | kpfleming | 2007-01-07 21:53:53 -0600 (Sun, 07 Jan 2007) | 2 lines since we use AC_PATH_TOOL to find tools, we should use the results it provides for us (reported by Brian Capouch on the asterisk-dev list) ........ 2007-01-07 21:46 +0000 [r49832-49835] Tilghman Lesher * /, apps/app_dictate.c: Merged revisions 49834 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r49834 | tilghman | 2007-01-07 15:44:52 -0600 (Sun, 07 Jan 2007) | 10 lines Merged revisions 49833 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r49833 | tilghman | 2007-01-07 15:43:10 -0600 (Sun, 07 Jan 2007) | 2 lines If openstream fails, then we crash (Issue 8564) ........ ................ * /, channels/chan_sip.c: Merged revisions 49831 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49831 | tilghman | 2007-01-07 15:24:04 -0600 (Sun, 07 Jan 2007) | 2 lines Second condition was a subset of the first, so hold was never decremented, thus hint stayed stuck (Issue 8747) ........ 2007-01-07 19:00 +0000 [r49816] Joshua Colp * funcs/func_base64.c, funcs/func_blacklist.c, funcs/func_callerid.c: One const, two const. Let's stick with everything else - one const. Plus older versions of GCC don't support double const either. 2007-01-07 16:21 +0000 [r49784-49801] Tilghman Lesher * res/res_config_odbc.c, include/asterisk/config.h, res/res_realtime.c, main/config.c, funcs/func_realtime.c: When calling the Realtime app more than once, unset fields which were previously set are erroneously still set (Bug 6701). After discussion, it was determined this should only be changed in trunk. * funcs/func_shell.c, funcs/func_strings.c, funcs/func_cut.c: Modifications to allow the output of SHELL() to be split per line (Issue 8676) * funcs/func_shell.c (added): Add function to execute a shell command and return the output (Issue 8676) * main/channel.c: Reduce duplication of code (Issue 6542) 2007-01-07 07:43 +0000 [r49769] Jason Parker * main/indications.c: Fix a segfault when using "countries" that don't have a matching zone. 2007-01-06 00:28 +0000 [r49743] Jason Parker * main/pbx.c, /, res/res_features.c, pbx/pbx_config.c: Merged revisions 49742 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49742 | qwell | 2007-01-05 18:24:38 -0600 (Fri, 05 Jan 2007) | 7 lines Save 1 whopping byte of allocated memory! This looks like it may have been a chicken/egg scenario.. You had to call a cleanup func, because everything was allocated. Then since you had to call a cleanup func, you were forced to allocate - ie; strdup(""). ........ 2007-01-06 00:13 +0000 [r49727-49741] Kevin P. Fleming * funcs/func_base64.c, funcs/func_rand.c, funcs/func_md5.c, funcs/func_db.c, channels/chan_zap.c, funcs/func_module.c, funcs/func_version.c, funcs/func_timeout.c, funcs/func_env.c, funcs/func_strings.c, funcs/func_math.c, funcs/func_vmcount.c, funcs/func_cut.c, include/asterisk/channel.h, funcs/func_sha1.c, funcs/func_logic.c, funcs/func_uri.c, funcs/func_global.c, funcs/func_realtime.c, funcs/func_enum.c, funcs/func_curl.c, funcs/func_groupcount.c, funcs/func_odbc.c, funcs/func_blacklist.c, funcs/func_cdr.c, funcs/func_channel.c, funcs/func_callerid.c: finish const-ifying ast_func_read() * main/manager.c: probably shouldn't leave the mmap'ed file hanging around in memory * /, configure, acinclude.m4: Merged revisions 49714-49715 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49714 | kpfleming | 2007-01-05 17:49:52 -0600 (Fri, 05 Jan 2007) | 2 lines proper fix for r49712 ........ r49715 | kpfleming | 2007-01-05 17:51:31 -0600 (Fri, 05 Jan 2007) | 2 lines one more time... ........ * main/manager.c, include/asterisk/config.h, main/config.c: a little more const-ification 2007-01-05 23:51 +0000 [r49716] Joshua Colp * codecs/codec_zap.c: It is possible for framein to get called and no channel be available, so do a check before we increment the count. 2007-01-05 23:41 +0000 [r49711-49713] Kevin P. Fleming * /, configure, acinclude.m4: Merged revisions 49712 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49712 | kpfleming | 2007-01-05 17:40:29 -0600 (Fri, 05 Jan 2007) | 2 lines if --with-foo= is specific for a configure option, ensure that it is used for header file checking as well ........ * main/pbx.c, /, channels/chan_sip.c, channels/chan_agent.c, pbx/pbx_dundi.c, include/asterisk/pbx.h, apps/app_queue.c, channels/chan_iax2.c, main/db.c, apps/app_speech_utils.c, include/asterisk/astdb.h, apps/app_voicemail.c: const-ify some more APIs, and fix rev 49710 from branch-1.4 in a better way here 2007-01-05 23:31 +0000 [r49709] Matt O'Gorman * codecs/codec_zap.c: no need to spam everyone with show transcoder messages 2007-01-05 23:17 +0000 [r49706] Jason Parker * channels/chan_zap.c, /, codecs/codec_zap.c: Merged revisions 49705 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49705 | qwell | 2007-01-05 17:16:16 -0600 (Fri, 05 Jan 2007) | 4 lines Make codec_zap and chan_zap also depend on zaptel. This fixes an issue (8727) with zaptel being in a different directory, using --with-zaptel. ........ 2007-01-05 22:53 +0000 [r49678-49681] Kevin P. Fleming * main/manager.c, /: Merged revisions 49680 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49680 | kpfleming | 2007-01-05 16:52:37 -0600 (Fri, 05 Jan 2007) | 2 lines don't 'consume' the params list before we try to use it again ........ * main/manager.c: use mmap() to read in the results of the manager action for an HTTP request, instead of reading it into a buffer * main/pbx.c, channels/chan_zap.c, /, channels/chan_sip.c, apps/app_meetme.c, res/res_features.c, channels/chan_agent.c, utils/astman.c, res/res_jabber.c, include/asterisk/manager.h, channels/chan_iax2.c, apps/app_queue.c, main/config.c, res/res_monitor.c, main/manager.c, include/asterisk/jabber.h, apps/app_senddtmf.c, main/db.c: Merged revisions 49676 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49676 | kpfleming | 2007-01-05 16:16:33 -0600 (Fri, 05 Jan 2007) | 2 lines reduce stack consumption for AMI and AMI/HTTP requests by nearly 20K in most cases ........ 2007-01-05 22:18 +0000 [r49677] Joshua Colp * main/channel.c, /: Merged revisions 49675 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49675 | file | 2007-01-05 17:14:47 -0500 (Fri, 05 Jan 2007) | 2 lines Don't keep repeating the warning over and over when the end of the call is reached. (issue #8724 reported by xrg) ........ 2007-01-05 17:10 +0000 [r49578-49637] Kevin P. Fleming * /, channels/chan_sip.c, channels/chan_skinny.c, channels/chan_iax2.c: Merged revisions 49636 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r49636 | kpfleming | 2007-01-05 11:09:00 -0600 (Fri, 05 Jan 2007) | 10 lines Merged revisions 49635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r49635 | kpfleming | 2007-01-05 10:56:40 -0600 (Fri, 05 Jan 2007) | 2 lines ensure that threads which are supposed to be detached (because we aren't going to wait on them) are created properly ........ ................ * main/threadstorage.c: use a rwlock-list for the list of TLS objects * /, channels/chan_iax2.c: Merged revisions 49600 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49600 | kpfleming | 2007-01-04 18:01:40 -0600 (Thu, 04 Jan 2007) | 2 lines revert the dynamic_list insertion change... that was not the right thing to do ........ * /, channels/chan_iax2.c: Merged revisions 49581 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49581 | kpfleming | 2007-01-04 17:50:15 -0600 (Thu, 04 Jan 2007) | 3 lines create the IAX2 processing threads as background threads so they will use smaller stacks when we create a dynamic thread, put it on the dynamic_list right away so we don't lose track of it ........ * include/asterisk/strings.h: ensure that the proper file/function/line shows up for dynamic string threadstorage objects remove pointless casts * include/asterisk/threadstorage.h: yeah... so... compiling before committing seems like it might be a good idea * build_tools/cflags.xml, include/asterisk.h, /, main/threadstorage.c (added), main/Makefile, include/asterisk/strings.h, include/asterisk/threadstorage.h, main/asterisk.c: Merged revisions 49553 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49553 | kpfleming | 2007-01-04 16:51:01 -0600 (Thu, 04 Jan 2007) | 2 lines add support for tracking thread-local-storage objects that exist via 'threadstorage' CLI commands ........ 2007-01-04 23:02 +0000 [r49552-49573] Joshua Colp * /, channels/chan_iax2.c: Merged revisions 49568 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49568 | file | 2007-01-04 18:00:50 -0500 (Thu, 04 Jan 2007) | 2 lines It's possible for the iax2 pvt to disappear, so if it has... don't bother looking for dpentries. ........ * /, main/config.c: Merged revisions 49551 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49551 | file | 2007-01-04 17:28:29 -0500 (Thu, 04 Jan 2007) | 2 lines Only free comments and line buffer once we reach the first level. (issue #8678 reported by ssokol, fixed by anthonyl) ........ 2007-01-04 21:59 +0000 [r49538] Kevin P. Fleming * main/frame.c, /, channels/iax2-parser.c: Merged revisions 49536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49536 | kpfleming | 2007-01-04 15:58:42 -0600 (Thu, 04 Jan 2007) | 2 lines don't mark these allocations as 'cache' allocations when caching has been disabled ........ 2007-01-04 21:40 +0000 [r49525] Joshua Colp * main/manager.c: It's pretty difficult to pthread_kill a thread that doesn't exist. (issue #8681 reported by bkruse) 2007-01-04 21:06 +0000 [r49524] Kevin P. Fleming * /, channels/iax2-parser.c: Merged revisions 49523 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49523 | kpfleming | 2007-01-04 15:06:02 -0600 (Thu, 04 Jan 2007) | 2 lines if we're going to decrement the frame count when we free a frame, we should inrement it when we create one :-) ........ 2007-01-04 20:27 +0000 [r49491-49507] TransNexus OSP Development * doc/osp.txt: 1. Update osp guide. * configs/osp.conf.sample: 1. Update osp module configuration file. 2007-01-04 18:32 +0000 [r49466] Kevin P. Fleming * channels/iax2-parser.h, /, channels/chan_iax2.c, channels/iax2-parser.c: Merged revisions 49465 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49465 | kpfleming | 2007-01-04 12:31:55 -0600 (Thu, 04 Jan 2007) | 2 lines only do IAX2 frame caching for voice and video frames ........ 2007-01-04 18:28 +0000 [r49464] Matt O'Gorman * /, apps/app_voicemail.c: Merged revisions 49459 via svnmerge from https://svn.digium.com/svn/asterisk/branches/1.4 ................ r49459 | mogorman | 2007-01-04 12:11:19 -0600 (Thu, 04 Jan 2007) | 10 lines Merged revisions 49447 via svnmerge from https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49447 | mogorman | 2007-01-04 11:45:16 -0600 (Thu, 04 Jan 2007) | 2 lines converted a lot of 256 to PATH_MAX and some white space fixes. ........ ................ 2007-01-04 18:19 +0000 [r49463] Kevin P. Fleming * codecs/Makefile, main/frame.c, /, channels/iax2-parser.c: Merged revisions 49457,49460-49461 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49457 | kpfleming | 2007-01-04 12:05:47 -0600 (Thu, 04 Jan 2007) | 2 lines make building of codec_gsm against the system GSM library actually work ........ r49460 | kpfleming | 2007-01-04 12:16:40 -0600 (Thu, 04 Jan 2007) | 2 lines don't define this type either if LOW_MEMORY is enabled ........ r49461 | kpfleming | 2007-01-04 12:17:01 -0600 (Thu, 04 Jan 2007) | 2 lines don't do frame header caching in the core if LOW_MEMORY is defined ........ 2007-01-04 18:17 +0000 [r49414-49462] Matt O'Gorman * /, channels/iax2-parser.c: Merged revisions 49458 via svnmerge from https://svn.digium.com/svn/asterisk/branches/1.4 ........ r49458 | kpfleming | 2007-01-04 12:06:51 -0600 (Thu, 04 Jan 2007) | 2 lines don't do frame caching in LOW_MEMORY mode ........ * /, apps/app_voicemail.c: Merged revisions 49413 via svnmerge from https://svn.digium.com/svn/asterisk/branches/1.4 ................ r49413 | mogorman | 2007-01-04 10:50:56 -0600 (Thu, 04 Jan 2007) | 11 lines Merged revisions 49412 via svnmerge from https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49412 | mogorman | 2007-01-04 10:48:43 -0600 (Thu, 04 Jan 2007) | 3 lines good catch russell sorry i missed that. fix magic number with proper sizeof ........ ................ 2007-01-03 23:41 +0000 [r49356] Matt O'Gorman * /, apps/app_voicemail.c: Merged revisions 49355 via svnmerge from https://svn.digium.com/svn/asterisk/branches/1.4 ................ r49355 | mogorman | 2007-01-03 17:32:03 -0600 (Wed, 03 Jan 2007) | 14 lines Merged revisions 49354 via svnmerge from https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49354 | mogorman | 2007-01-03 17:22:47 -0600 (Wed, 03 Jan 2007) | 6 lines When using ODBC_STORAGE VoicemailMain doesn't create the subdirectories for a mailbox such as the INBOX directory. this patch solves that problem, was written by anthony be-125 ........ ................ 2007-01-03 11:15 +0000 [r49320-49321] Christian Richter * doc/misdn.txt, channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c, configs/misdn.conf.sample, channels/misdn/isdn_lib.c, channels/misdn_config.c: Merged revisions 49313 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r49313 | crichter | 2007-01-03 10:06:50 +0100 (Mi, 03 Jan 2007) | 41 lines Merged revisions 48319,48321,48467,48552,48576,49135,49303 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) | 1 line changed a few debugs to higher debug levels ........ r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) | 1 line added the export and import of the MISDN_ADDRESS_COMPLETE Variable to inidcate wether the extension is already completely dialed or if there might come additional digits by information elements. also added some docs for that. ........ r48467 | crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line removed FIXUP state. added check for channel allocation conflict when we create a setup while the other site creates a setup on the same channel, besides the check we resolve this conflict. ........ r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18 Dez 2006) | 1 line when our PTP Partner sends us a SETUP with a preselected channel we just accept it, even when we're NT. added some checks for segfaults. ........ r48576 | crichter | 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line when we reject a channel, because it's in use already, we shouldn't process the setup anymore. made the channel allocation a bit easier and more understandable, removed a few unused lines ........ r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02 Jan 2007) | 1 line added check for channel ranges in the set/empty channel functions. set pmp_l1_check default to no. added misdn restart pid cli command. added cleaning of channel when we send a RELEASE_COMPLETE. ........ r49303 | crichter | 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines * Added check for bridging in misdn_call to avoid setting echocancellation when 2 mISDN channels are involved and when bridging is set. That lead to a kernel panic before under different situations, because we switched about 2 times between hardware bridging and echocancelation * readded MISDN_URATE variable which got lost before, this should make app_v110 work again * fixed typo ........ ................ * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c, channels/misdn_config.c: Merged revisions 47989 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r47989 | crichter | 2006-11-24 16:46:13 +0100 (Fr, 24 Nov 2006) | 9 lines Merged revisions 47968 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47968 | crichter | 2006-11-23 17:10:23 +0100 (Do, 23 Nov 2006) | 1 line fixed a litle bug regarding HOLD/RETRIEVE. beatufied some logs, changed some loglevels. changed the default value of block_on_alarm ........ ................ 2007-01-03 03:28 +0000 [r49283] Kevin P. Fleming * Makefile, /, Makefile.rules: Merged revisions 49282 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49282 | kpfleming | 2007-01-02 21:21:25 -0600 (Tue, 02 Jan 2007) | 2 lines various Makefile improvements to get chan_vpb (and any other C++ modules) to build properly ........ 2007-01-03 01:21 +0000 [r49260] Joshua Colp * /, channels/chan_iax2.c: Merged revisions 49259 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49259 | file | 2007-01-02 20:19:53 -0500 (Tue, 02 Jan 2007) | 2 lines Check pvt structure presence before passing to send_command. This gets rid of the irritating message about a packet without pvt structure. This happens because the scheduled item is getting cancelled at almost the exact moment it is getting executed. ........ 2007-01-02 22:43 +0000 [r49238] Steve Murphy * /, main/ast_expr2f.c, pbx/ael/ael_lex.c, pbx/ael/ael.flex, main/ast_expr2.fl: Merged revisions 49237 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49237 | murf | 2007-01-02 15:30:53 -0700 (Tue, 02 Jan 2007) | 1 line This is a slight modification to Josh's edits for #8579; both files edited were the produced by flex; so the source files need to be changed instead, and the generated files regenerated. ........ 2007-01-02 20:02 +0000 [r49214-49215] Olle Johansson * channels/chan_sip.c: Removing propably accidentally added debug messages sent to verbose channel * /, channels/chan_sip.c: Merged revisions 49212 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49212 | oej | 2007-01-02 20:58:45 +0100 (Tue, 02 Jan 2007) | 2 lines Small cleanup of add_t38sdp - it's always enabled at that point in the code ........ 2007-01-02 17:04 +0000 [r49187] Tilghman Lesher * funcs/func_math.c: Tweak description text to match new functionality (Issue 7959) 2007-01-02 14:01 +0000 [r49166] Kevin P. Fleming * channels/chan_zap.c, /: Merged revisions 49165 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49165 | kpfleming | 2007-01-02 07:59:44 -0600 (Tue, 02 Jan 2007) | 2 lines remove comment that is unrelated to this function ........ 2007-01-02 13:50 +0000 [r49152] Olle Johansson * /, configs/features.conf.sample: Update sample config 2007-01-01 23:43 +0000 [r49100-49103] Kevin P. Fleming * channels/chan_zap.c, /, build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, configure.ac, codecs/codec_zap.c: Merged revisions 49102 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49102 | kpfleming | 2007-01-01 17:34:35 -0600 (Mon, 01 Jan 2007) | 2 lines check specifically for VLDTMF and transcoding support in the system's Zaptel installation, and make only the modules that need those features dependent on them (this will allow building the other Zaptel-using parts of Asterisk against older versions of Zaptel or those on other platforms that haven't caught up yet to the Linux version) ........ * Makefile, sounds/Makefile: GNU make already knows what the current directory is, there is no need to use 'pwd' * Makefile, /: Merged revisions 49098-49099 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49098 | kpfleming | 2007-01-01 16:08:24 -0600 (Mon, 01 Jan 2007) | 2 lines revert this change until a better solution can be found... 'env -i' was not being used properly, but even when changed to do so, this process fails during cross-compilation because the menuselect build still sees 'CC' as set to the cross-compiler ........ r49099 | kpfleming | 2007-01-01 16:48:03 -0600 (Mon, 01 Jan 2007) | 2 lines use a simpler (and portable) method to ensure that menuselect is built as a host binary ........ 2007-01-01 20:16 +0000 [r49092-49097] Olle Johansson * /: Block cleanup of release branch * include/asterisk/indications.h: Doxygen documentationification * main/manager.c: Fix manager too. * main/frame.c, channels/chan_sip.c, include/asterisk/frame.h: - Add error handling to ast_parse_allow_disallow - Use this in chan_sip configuration parsing * include/asterisk/acl.h, channels/chan_sip.c, channels/chan_skinny.c, channels/chan_h323.c, main/acl.c, channels/chan_iax2.c, channels/chan_mgcp.c: - Implement error handling in ast_append_ha - Use this in chan_sip - Document ha functions in acl.c 2006-12-31 19:15 +0000 [r49089] Joshua Colp * channels/chan_iax2.c: count is no longer used in the iaxq structure really so let's just make this a statically declared linked list. 2006-12-31 09:38 +0000 [r49080-49082] Olle Johansson * CHANGES: Update CHANGES, make section about SIP. This might be a good way to handle other parts of this file too, as it grows. * configs/sip.conf.sample: Added some docs * channels/chan_sip.c: Add version number to useragent string - Issue #8700, blanchet - THANKS! 2006-12-31 05:20 +0000 [r49075-49076] Tilghman Lesher * funcs/func_math.c: Add power and right/left shift functions (Issue 7959) * configs/voicemail.conf.sample, UPGRADE.txt, apps/app_voicemail.c: 1. Rename 'maxmessage' to 'maxsecs' to differentiate from 'maxmsg'. 2. Rename 'minmessage' to 'minsecs' for parity. 3. Make 'maxsecs' a per-user option, in addition to global. (Issue # 8624) 2006-12-30 18:32 +0000 [r49071-49074] Joshua Colp * /, pbx/pbx_config.c: Merged revisions 49073 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49073 | file | 2006-12-30 13:31:17 -0500 (Sat, 30 Dec 2006) | 2 lines IAX has been deprecated for quite some time so we had better use IAX2 when creating the dial string for users. (issue #8697 reported by ssokol) ........ * main/rtp.c: Clarify why we are reading in a frame in the Packet2Packet bridge. 2006-12-30 13:27 +0000 [r49068-49069] Kevin P. Fleming * sounds/Makefile: now that the 'languageprefix' option defaults to 'on', and all channels have a default language of 'en', let's install the English sound files into /var/lib/asterisk/sounds/en, just like the other languages * main/channel.c: small formatting fix 2006-12-30 05:49 +0000 [r49064-49067] Joshua Colp * /, main/rtp.c: Merged revisions 49066 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49066 | file | 2006-12-30 00:46:57 -0500 (Sat, 30 Dec 2006) | 2 lines If the Packet2Packet bridge is being broken because of a masquerade then attempt to read a frame in so the masquerade actually happens. Otherwise weirdness will occur. (issue #8696 reported by kjotte) ........ * funcs/func_odbc.c: Initialize obj pointers to NULL. Gets rid of two compiler warnings. * /, channels/chan_iax2.c: Merged revisions 49063 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49063 | file | 2006-12-29 22:37:22 -0500 (Fri, 29 Dec 2006) | 2 lines Initialize the packet queue in load_module instead of just declaring the list with the default value. (issue #8695 reported by ssokol) ........ 2006-12-30 00:51 +0000 [r49062] Steve Murphy * /, pbx/pbx_ael.c: Merged revisions 49061 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49061 | murf | 2006-12-29 17:40:37 -0700 (Fri, 29 Dec 2006) | 1 line A fix for 8661, where the CUT func needed to have comma args converted to vertical bars. I hope this change does little harm. ........ 2006-12-29 13:25 +0000 [r49056] Russell Bryant * channels/chan_oss.c: Convert various comments to doxygen format. 2006-12-29 11:02 +0000 [r49054] Olle Johansson * channels/chan_sip.c: Removing extra output 2006-12-29 06:26 +0000 [r49053] Russell Bryant * include/asterisk/smdi.h: Fix a spelling mistake in a comment. 2006-12-29 00:33 +0000 [r49047] Kevin P. Fleming * /, BUGS: Merged revisions 49046 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r49046 | kpfleming | 2006-12-28 18:32:59 -0600 (Thu, 28 Dec 2006) | 10 lines Merged revisions 49045 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r49045 | kpfleming | 2006-12-28 18:32:32 -0600 (Thu, 28 Dec 2006) | 2 lines location of the bug posting guidelines has changed ........ ................ 2006-12-28 20:13 +0000 [r49030] Tilghman Lesher * configs/func_odbc.conf.sample, funcs/func_odbc.c, funcs/func_strings.c: Integrate functionality tested on svncommunity users back into trunk 2006-12-28 20:10 +0000 [r49029] Kevin P. Fleming * /, sounds/Makefile: Merged revisions 49028 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49028 | kpfleming | 2006-12-28 14:08:59 -0600 (Thu, 28 Dec 2006) | 2 lines new versions of sounds ........ 2006-12-28 20:05 +0000 [r49026-49027] Joshua Colp * main/http.c: Convert uri_redirects list to read/write locks. * /, main/http.c: Merged revisions 49024 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49024 | qwell | 2006-12-28 14:52:46 -0500 (Thu, 28 Dec 2006) | 2 lines make the uris_lock a rwlock instead of a mutex lock - needs to be forward ported to trunk ........ 2006-12-28 17:56 +0000 [r49019] Steve Murphy * pbx/ael/ael.tab.c, main/ast_expr2.y, main/ast_expr2f.c, pbx/ael/ael_lex.c, include/asterisk/ael_structs.h, pbx/ael/ael.tab.h, utils/ael_main.c, main/ast_expr2.fl, main/ast_expr2.c: Jason is having problems with the inclusion of ; it appears to be unnecessary for sucessful builds, so I either removed or commented out the inclusions from all the AEL related code. New outputs from bison/flex are included, etc. 2006-12-27 22:30 +0000 [r49010] Joshua Colp * /, main/ast_expr2f.c, pbx/ael/ael_lex.c: Merged revisions 49009 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49009 | file | 2006-12-27 17:28:46 -0500 (Wed, 27 Dec 2006) | 2 lines ast_copy_string is not available when LOW_MEMORY is used and things are being built in the utils directory, so we need to resort to the old method of strncpy. (issue #8579 reported by mottano) ........ 2006-12-27 22:14 +0000 [r49007-49008] Kevin P. Fleming * main/cdr.c, main/channel.c, main/udptl.c, main/pbx.c, main/dnsmgr.c, main/frame.c, main/manager.c, /, main/http.c, main/logger.c, main/enum.c, main/asterisk.c, main/rtp.c, main/term.c: Merged revisions 49006 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49006 | kpfleming | 2006-12-27 16:06:56 -0600 (Wed, 27 Dec 2006) | 2 lines since these variables all have static duration, none of them need initializers (they default to zero anyway) ........ * codecs/g722: add file to ignore list 2006-12-27 21:27 +0000 [r49004] Olle Johansson * /, channels/chan_sip.c: Only include include files once (imported from 1.4) 2006-12-27 21:21 +0000 [r48999-49001] Kevin P. Fleming * main/asterisk.c: apparently we need an explicit message to warn people * main/file.c, UPGRADE.txt, main/asterisk.c, doc/asterisk-conf.txt: make the 'languageprefix' option default to on, and deprecate turning it off * /, main/file.c, include/asterisk/options.h, main/asterisk.c: Merged revisions 48998 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48998 | kpfleming | 2006-12-27 15:08:30 -0600 (Wed, 27 Dec 2006) | 3 lines move extern declaration for this option to a header file where it belongs provide an initial value for 'languageprefix' option, instead of relying on randomness to provide a useful value ........ 2006-12-27 20:30 +0000 [r48992-48996] Olle Johansson * /, channels/chan_sip.c: Only set "rfc2833compensate" option once * /, channels/chan_sip.c: Only handle T38 options once * channels/chan_sip.c: -Remove "localmask" setting (deprecated in earlier version) - Remove "musiconhold" and "musicclass" settings (also deprecated earlier) 2006-12-27 18:34 +0000 [r48989-48990] Kevin P. Fleming * /, channels/chan_sip.c: Merged revisions 48988 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48988 | kpfleming | 2006-12-27 12:33:22 -0600 (Wed, 27 Dec 2006) | 2 lines make the option actually match the documentation ........ * include/asterisk/utils.h, include/asterisk/astmm.h, main/frame.c, /, main/astmm.c, channels/iax2-parser.c: Merged revisions 48987 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48987 | kpfleming | 2006-12-27 12:29:13 -0600 (Wed, 27 Dec 2006) | 2 lines allow 'show memory' and 'show memory summary' to distinguish memory allocations that were done for caching purposes, so they don't look like memory leaks ........ 2006-12-27 18:02 +0000 [r48976-48986] Olle Johansson * /, channels/chan_sip.c, configs/sip.conf.sample: Be politically correct * apps/app_sms.c: From coding guidelines: Comments should explain what the code does, not when something was changed or who changed it. If you have done a larger contribution, make sure that you are added to the CREDITS file. * /, channels/chan_sip.c, configs/sip.conf.sample: Add support for buggy Cisco MWI firmware > 8.0.3 (issue 8575 - flewid) * /, channels/chan_sip.c: Cleanup of handle_common_options * /, channels/chan_sip.c: Reset invitestate when sending new invite * /, channels/chan_sip.c: Issue #8600 - bogus SDP Content Length in T.38 re-invite 2006-12-26 05:23 +0000 [r48961-48967] Joshua Colp * /, apps/app_meetme.c: Merged revisions 48966 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48966 | file | 2006-12-26 00:20:08 -0500 (Tue, 26 Dec 2006) | 2 lines Get rid of a needless memory allocation and only create a conference structure in find_conf_realtime if data was read from realtime. (issue #8669 reported by robl) ........ * /, channels/chan_sip.c, include/asterisk/rtp.h, main/rtp.c: Merged revisions 48964 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48964 | file | 2006-12-25 23:31:58 -0500 (Mon, 25 Dec 2006) | 2 lines Add an API call that initializes an RTP structure. We need this because chan_sip is cheeky and uses a temporary RTP structure for codec purposes, and the API calls that are used rely on the lock. (Pointed out on asterisk-dev by Andy Wang) ........ * /, configure, configure.ac: Merged revisions 48960 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48960 | file | 2006-12-25 12:04:48 -0500 (Mon, 25 Dec 2006) | 2 lines Clean up autoconf file (gets rid of warnings seen when rebuilding configure) and rebuild configure. ........ 2006-12-25 06:42 +0000 [r48958-48959] Luigi Rizzo * codecs/g722/g722.h: provide INT16_MIN and INT16_MAX for platforms where they are not defined. * main/channel.c, apps/app_read.c, channels/chan_misdn.c, funcs/func_channel.c, include/asterisk/indications.h, apps/app_disa.c, main/app.c, res/snmp/agent.c, contrib/utils/zones2indications.c, include/asterisk/channel.h, res/res_indications.c, main/indications.c: rename the structs struct tone_zone_sound and struct tone_zone defined in indications.h to ind_tone_zone_sound and ind_tone_zone, to avoid conflicts with the structs with the same names defined in tonezone.h Hope i haven't missed any instance. 2006-12-25 05:22 +0000 [r48929-48957] Russell Bryant * /, funcs/func_math.c: Merged revisions 48956 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r48956 | russell | 2006-12-25 00:21:20 -0500 (Mon, 25 Dec 2006) | 14 lines Merged revisions 48955 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48955 | russell | 2006-12-25 00:19:48 -0500 (Mon, 25 Dec 2006) | 6 lines Fix an error introduced by copying and pasting the handling of the >= operator for the MATH function. If a single equal sign was used as an operator, the function would treat it is as if it were the >= operator. Now, it properly handles it as an invalid operator. (issue #8665, patch by tempest1) ........ ................ * funcs/func_callerid.c: Simplify the if statements used to check to see if the argument was "num" or "number". It is not possible to ever reach the second part of this conditional statement. Thanks to my brother, Brett, for pointing this out. :) * main/frame.c: Resolve some compiler warnings * /, channels/chan_oss.c: Merged revisions 48948 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48948 | russell | 2006-12-24 16:19:37 -0500 (Sun, 24 Dec 2006) | 3 lines Fix a typo in an error message that indicated that the MGCP channel type could not be registered, instead of the correct type, OSS. ........ * main/http.c, configs/http.conf.sample: Use spaces as a separator for the redirect option to improve readability * /, channels/chan_iax2.c: Merged revisions 48944 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r48944 | russell | 2006-12-24 02:25:38 -0500 (Sun, 24 Dec 2006) | 11 lines Merged revisions 48943 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48943 | russell | 2006-12-24 02:23:07 -0500 (Sun, 24 Dec 2006) | 3 lines Check for the proper return value on an error in a call to mmap(). This was reported by Andy Wang on the asterisk-dev list. Thanks! ........ ................ * channels/chan_sip.c: Merged revisions 48940 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r48940 | russell | 2006-12-24 01:49:31 -0500 (Sun, 24 Dec 2006) | 11 lines Merged revisions 48939 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48939 | russell | 2006-12-24 01:47:29 -0500 (Sun, 24 Dec 2006) | 3 lines Remove a couple of misplaced dots in log messages. This was reported by Andrea Spadaccini on the asterisk-dev mailing list. ........ ................ * main/http.c: Simplify the definition of http_uri_redirect such that only one allocation is done for exactly how much memory is needed. This was suggested by Luigi on the asterisk-dev mailing list. Thanks! * include/asterisk/http.h, main/http.c, CHANGES, configs/http.conf.sample: - Convert the list of URI handlers to use the linked list macros. While doing this, implementing locking of this list to make it thread-safe. - Add a "redirect" option to http.conf that allows redirecting one URI to another. I was inspired to do this while playing with the Asterisk GUI. I got tired of typing this URL to get to the GUI: http://localhost:8088/asterisk/static/config/cfgadvanced.html So, now I have the following line in http.conf: redirect=/=/asterisk/static/config/cfgadvanced.html Now, I can type the following into my browser and go to the GUI: http://localhost:8088 * main/manager.c: Remove a debug message. If this is still needed for debugging something, it should be made a LOG_DEBUG message. 2006-12-23 19:55 +0000 [r48928] Joshua Colp * include/asterisk/lock.h: We should probably declare the lock... and not just the constructor/deconstructor. 2006-12-23 19:51 +0000 [r48927] Russell Bryant * include/asterisk/lock.h: Use the correct function to destroy an rwlock in the destructor for an ast_rwlock_t 2006-12-22 22:34 +0000 [r48871-48907] Jason Parker * Makefile, /, main/stdtime/localtime.c: Merged revisions 48906 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48906 | qwell | 2006-12-22 16:33:46 -0600 (Fri, 22 Dec 2006) | 2 lines Minor fixes for Solaris. ........ * /, channels/chan_skinny.c: Merged revisions 48888 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48888 | qwell | 2006-12-22 15:40:20 -0600 (Fri, 22 Dec 2006) | 2 lines Note to self: Run make before committing... ........ * /, channels/chan_skinny.c: Merged revisions 48870 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48870 | qwell | 2006-12-22 14:43:05 -0600 (Fri, 22 Dec 2006) | 2 lines Fix for issue 7774 - patch by alamantia ........ 2006-12-22 10:35 +0000 [r48825-48857] Luigi Rizzo * apps/app_sms.c: improve readability of a few macros. * apps/app_sms.c: make sms_hexdump() thread safe; restructure and reduce indentation on some blocks. * apps/app_sms.c: make isodate thread-safe * apps/app_sms.c: - use the standard option parsing routines; - document existing but undocumented parameters to send a message (untested but unchanged; - ad a new option p(N) to set the initial message delay to N ms so this can be adapted from the dialplan to various countries; 2006-12-21 21:57 +0000 [r48785-48817] Joshua Colp * main/logger.c: Merge non-blocking logger from my branch. This should improve things under heavy load with lots of CLI/logging output. * /, redhat/asterisk.spec: Merged revisions 48783 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r48783 | file | 2006-12-21 15:26:29 -0500 (Thu, 21 Dec 2006) | 10 lines Merged revisions 48782 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48782 | file | 2006-12-21 15:25:01 -0500 (Thu, 21 Dec 2006) | 2 lines Add new silence sound files to the spec for Redhat. (issue #8652 reported by alvaro_palma_aste) ........ ................ 2006-12-21 20:15 +0000 [r48781] Steve Murphy * codecs/codec_g722.c: This little mod gets rid of that g722 compiler warning that breaks builds configured with --enable-dev-mode; the previous commit of 48767 was to merge in changes for bug 6334, unifying the open mode arguments for saner operation. 2006-12-21 19:52 +0000 [r48768] Luigi Rizzo * apps/app_sms.c: put generator functions next to each other. 2006-12-21 19:44 +0000 [r48767] Steve Murphy * include/asterisk.h, channels/chan_zap.c, apps/app_meetme.c, apps/app_festival.c, apps/app_dictate.c, apps/app_record.c, res/res_convert.c, channels/chan_iax2.c, res/res_monitor.c, cdr/cdr_sqlite.c, res/res_agi.c, main/file.c, main/app.c, apps/app_sms.c, apps/app_directory.c, apps/app_chanspy.c, apps/app_mixmonitor.c, main/db.c, apps/app_voicemail.c: a quick fix to app_sms.c to get rid of cursed compiler warnings so I can compile under --enable-dev-mode 2006-12-21 19:36 +0000 [r48736-48766] Luigi Rizzo * main/channel.c: same as in other places, check that generator->release is not NULL before calling it. This allows generators to set it to NULL when they have nothing to do there. Later, the three copies of the code that releases a generator should be moved to a function. * apps/app_sms.c: reduce indentation * apps/app_sms.c: restructure a block to reduce nesting * apps/app_sms.c: Add a bit of documentation on this code, including pointers to relevant documents and comment on timing issues. Initial merge of the code in http://bugs.digium.com/view.php?id=8586 by Filippo Grassilli (Hyppo) to support the SMS Protocol 2. In this commit i have tried to minimize the diffs, so further code cleanup will come in subsequent commits. 2006-12-21 15:52 +0000 [r48723] Steve Murphy * pbx/pbx_config.c: This small update will generate WARNINGS if there is garbage in your extensions.conf file (liken extem => instead of exten => !) 2006-12-21 04:05 +0000 [r48680-48709] Joshua Colp * include/asterisk/indications.h, main/indications.c: Really clean up indications to use the linkedlists API * main/pbx.c: Switch list of global variables to read/write locks. * main/pbx.c: Convert alternate dialplan switch list to use read/write locks. 2006-12-21 00:24 +0000 [r48663] Steve Murphy * configs/iax.conf.sample, main/jitterbuf.c, include/jitterbuf.h, CHANGES, channels/chan_iax2.c: As per bug 7978, this version introduces the jittertargetextra option in config files 2006-12-21 00:11 +0000 [r48661-48662] Matthew Fredrickson * codecs/codec_g722.c: Minor addition giving props to Steve Underwood for his hard work. Thanks again Steve! * codecs/Makefile, codecs/g722/Makefile (added), codecs/codec_g722.c (added), codecs/g722/g722_encode.c (added), codecs/g722 (added), build_tools/embed_modules.xml, codecs/g722/g722_decode.c (added), codecs/g722/g722.h (added), codecs/g722_slin_ex.h (added), codecs/slin_g722_ex.h (added): Add codec G.722 support. 2006-12-20 04:32 +0000 [r48638-48639] Joshua Colp * apps/app_page.c: Clean up app_page * /, apps/app_voicemail.c: Merged revisions 48637 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48637 | file | 2006-12-19 21:56:09 -0500 (Tue, 19 Dec 2006) | 2 lines vms doesn't exist on non-IMAP storage builds. ........ 2006-12-20 00:13 +0000 [r48598-48599] Luigi Rizzo * apps/app_sms.c: more formatting cleanup. Move some code into a function sms_compose1() in preparation for supporting protocol 2 as well. * apps/app_sms.c: formatting and code cleanup. Still a lot of copy&pasted code here... 2006-12-19 23:05 +0000 [r48591-48597] Joshua Colp * /, apps/app_voicemail.c: Merged revisions 48596 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48596 | file | 2006-12-19 18:04:30 -0500 (Tue, 19 Dec 2006) | 2 lines Pass 'vms' pointer to record_and_review so it is then passed to the IMAP store file function. (issue #8614 reported by punknow) ........ * res/snmp/agent.c: Update res_snmp to use new API declaration of pbx_builtin_serialize_variables (issue #8627 reported by johann8384) * /, doc/snmp.txt: Merged revisions 48592 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48592 | file | 2006-12-19 17:00:57 -0500 (Tue, 19 Dec 2006) | 2 lines find is not the same as bind when it comes to documentation. (issue #8626 reported by johann8384) ........ * res/res_limit.c: OpenBSD does not have RLIMIT_AS or RLIMIT_VMEM so until someone finds the right rlimit to use then let's not support the -v option on OpenBSD. (issue #8543 reported by jtodd) 2006-12-19 21:32 +0000 [r48588-48589] Luigi Rizzo * /: block 48583 * apps/app_sms.c: start documenting this code. On passing, fix the bogus datalen on outgoing frames just fixed in 1.4 rev.48583 2006-12-19 21:28 +0000 [r48587] Kevin P. Fleming * /, channels/Makefile: Merged revisions 48586 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48586 | kpfleming | 2006-12-19 15:28:04 -0600 (Tue, 19 Dec 2006) | 2 lines suppress compiler warnings in this module until it can be improved ........ 2006-12-19 16:36 +0000 [r48580-48581] Luigi Rizzo * apps/app_dial.c: better name for struct dial_localuser. * main/cli.c: remove now useless extern declarations. 2006-12-19 14:57 +0000 [r48578] Kevin P. Fleming * res/res_config_odbc.c, /, funcs/func_odbc.c: Merged revisions 48577 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48577 | kpfleming | 2006-12-19 08:57:09 -0600 (Tue, 19 Dec 2006) | 2 lines use the proper variable type for these unixODBC API calls, eliminating warnings on 64-bit platforms that use the 'new' 64-bit types for ODBC API calls ........ 2006-12-19 09:58 +0000 [r48573-48575] Luigi Rizzo * apps/app_dial.c: introduce a temporary variable for tmp->chan to shorten expressions. * apps/app_dial.c: stop what i think is a memory leak in case Dial fails to connect to a channel. Before committing to 1.4 i would like some other people to review and test this fix - thanks. * apps/app_dial.c: move a large block related to privacy handling to a separate function. 2006-12-19 03:47 +0000 [r48572] Joshua Colp * Makefile, /: Merged revisions 48571 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48571 | file | 2006-12-18 22:46:12 -0500 (Mon, 18 Dec 2006) | 2 lines Use env -i to start a fresh environment when going to build menuselect. This is more portable then using unset. (issue #8543 reported by jtodd) ........ 2006-12-18 17:44 +0000 [r48568] Luigi Rizzo * include/asterisk/channel.h: unbreak the macro used for incrementing the frame counters. I don't know when the bug was introduced, but with the typical usage c->fin = FRAMECOUNT_INC(c->fin) the frame counters stay to 0. 2006-12-18 17:30 +0000 [r48565-48567] Joshua Colp * channels/chan_iax2.c: Clean up find_idle_thread function and use atomic operations for dynamic thread count. * /, channels/chan_iax2.c: Merged revisions 48564 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48564 | file | 2006-12-18 12:15:49 -0500 (Mon, 18 Dec 2006) | 2 lines Put thread into proper list if we abort handling due to an error, and also hold the lock while putting it back into the proper idle list so we don't prematurely get a signal. (issue #8604 reported by arkadia) ........ 2006-12-18 16:57 +0000 [r48562-48563] Jason Parker * configure.ac: ctrl-w != w (nano search) (surprisingly, the fix was ever so slightly different in 1.4) 2006-12-18 16:24 +0000 [r48558-48560] Luigi Rizzo * include/asterisk/strings.h: apply the proposed fix for bug 8602 http://bugs.digium.com/view.php?id=8602 (i am not sure if there is still missing cast in front of the alloca() call - being a macro this is probably triggered only when actually used). Add function ast_str_reset() to reinitialize the string to an empty string instead of playing with the internal fields. * main/cdr.c, main/pbx.c, apps/app_dumpchan.c, include/asterisk/cdr.h, include/asterisk/pbx.h, apps/app_queue.c, main/cli.c: convert the final clients of ast_build_string to use ast_str_*() Now the only module left using it is chan_sip.c * main/logger.c: debugging shows that we always need more than 128 bytes for the verbose and logging messages so start with a larger buffer from the beginning. 2006-12-18 11:59 +0000 [r48555] Kevin P. Fleming * /, main/Makefile, codecs/gsm/Makefile, utils/astman.c, utils/smsq.c, codecs/ilbc/Makefile, utils/ael_main.c, codecs/lpc10/Makefile: Merged revisions 48554 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48554 | kpfleming | 2006-12-18 05:59:24 -0600 (Mon, 18 Dec 2006) | 3 lines remove some now-unnecessary explicit includes of autoconfig.h clean up per-file dependencies during 'make clean' ........ 2006-12-18 11:28 +0000 [r48550-48553] Luigi Rizzo * main/manager.c: Replace ast_build_string with ast_str_*(). On passing remove presumably duplicate code to generate the message for the manager_hooks: in the previous version, the message was almost the same as the one sent to regular sessions, with the exception of the empty line at the end, and a few (presumably unintentional) differences e.g. timestamps, debugging, and lowercase headers for "event" and "privilege". now we reuse the same message as before. * funcs/func_realtime.c: replace ast_build_string() with ast_str_*(). Unless i am very mistaken, function_realtime_read() was broken in that it would always return an empty string (because ast_build_string() advanced the pointer to the end of the string, and there was no reference to the initial value. This commit should fix this problem. * apps/app_queue.c: replace ast_build_string() with ast_str_*(); simplify __queues_show() 2006-12-17 18:33 +0000 [r48549] Kevin P. Fleming * /, build_tools/prep_tarball: Merged revisions 48548 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48548 | kpfleming | 2006-12-17 12:33:24 -0600 (Sun, 17 Dec 2006) | 2 lines need an additional argument here to make the downloads actually occur ........ 2006-12-17 12:47 +0000 [r48543] Luigi Rizzo * channels/chan_sip.c: define a mask SIP_INSECURE sam as for other sets of flags. 2006-12-16 21:38 +0000 [r48522-48529] Kevin P. Fleming * /, configure, include/asterisk/autoconfig.h.in, configure.ac, acinclude.m4: Merged revisions 48528 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48528 | kpfleming | 2006-12-16 15:34:41 -0600 (Sat, 16 Dec 2006) | 2 lines use m4 quoting for AC_MSG_NOTICE calls, to keep these calls from thinking they have multiple arguments ........ * /, agi, codecs, utils, main/Makefile, apps, Makefile.moddir_rules, Makefile.rules, cdr, codecs/ilbc, formats, utils/Makefile, agi/Makefile, Makefile, funcs, main/db1-ast, codecs/lpc10, build_tools/mkdep (removed), main, codecs/gsm, res, pbx, channels: Merged revisions 48525 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48525 | kpfleming | 2006-12-16 15:14:34 -0600 (Sat, 16 Dec 2006) | 2 lines simplify dependency tracking system, using the compiler's built-in method for generating them, and only doing dependency tracking if developer mode is enabled via the configure script ........ * funcs/func_curl.c: update to use trunk's version of the threadstorage API * Makefile, include/asterisk.h, /, main/stdtime/localtime.c: Merged revisions 48521 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48521 | kpfleming | 2006-12-16 14:12:41 -0600 (Sat, 16 Dec 2006) | 2 lines since we really, really have to have autoconfig.h included before all other headers (especially system headers), the Makefile will now force it to happen (this will fix build problems with files like ast_expr2f.c, where we can't control the inclusion order in the file itself) ........ 2006-12-16 11:23 +0000 [r48515-48520] Luigi Rizzo * main/utils.c: forgot this part... * main/cli.c: another conversion from ast_build_str to ast_str * main/translate.c: convert ast_build_str to ast_str_* * include/asterisk/http.h, main/manager.c, main/http.c, include/asterisk/strings.h: replace ast_build_string() with ast_str_*() functions. This makes the code easier to follow and saves some copies to intermediate buffers. 2006-12-16 04:25 +0000 [r48514] Kevin P. Fleming * funcs/func_curl.c, /: Merged revisions 48513 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48513 | kpfleming | 2006-12-15 22:25:09 -0600 (Fri, 15 Dec 2006) | 2 lines instead of initializing the curl library every time the CURL() function is invoked, do it only once per thread (this allows multiple calls to CURL() in the dialplan for a channel to run much more quickly, and also to re-use connections to the server) (thanks to JerJer for frequently complaining about this performance problem) ........ 2006-12-16 02:42 +0000 [r48508-48512] Luigi Rizzo * res/res_limit.c: prevent a compiler warning * main/manager.c, main/logger.c, main/utils.c, include/asterisk/strings.h, main/cli.c: simplify the ast_dynamic_str_*.... routines by renaming them to ast_str ... and putting the struct ast_threadstorage pointer into the struct ast_str. This makes the code a lot more readable. At this point we can use these routines also to replace ast_build_string(). * include/asterisk/utils.h, main/utils.c, include/asterisk/strings.h, include/asterisk/threadstorage.h: move the dynamic string support in a better place i.e. string.h While doing this, add a bit of documentation, and slightly extend the functionality as follows: + a max_len of -1 means that we take whatever the current size is, and never try to extend the buffer; + add support for alloca()-ted dynamic strings, which is very useful for all cases where we do an ast_build_string() now. Next step is to simplify the interface by using shorter names (e.g. ast_str as a prefix) and removing the _thread variant of the functions by saving the threadstorage reference into the struct ast_str. This can be done by overloading the 'type' field. Finally, I will do my best to remove the convoluted interface that results from trying to support platforms without va_copy(). * res/res_smdi.c: remove a duplicate include 2006-12-15 19:57 +0000 [r48503-48507] Joshua Colp * /, main/rtp.c: Merged revisions 48506 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48506 | file | 2006-12-15 14:55:28 -0500 (Fri, 15 Dec 2006) | 2 lines Turn payload_lock into bridge_lock and make it encompass all RTP structure contents that may relate to bridge information, including who we are bridged to. ........ * /, channels/chan_iax2.c: Merged revisions 48504 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48504 | file | 2006-12-15 14:38:51 -0500 (Fri, 15 Dec 2006) | 2 lines Hold call structure lock in places where a qualify or peer action can destroy it. ........ * /, channels/chan_iax2.c: Merged revisions 48502 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48502 | file | 2006-12-15 14:24:15 -0500 (Fri, 15 Dec 2006) | 2 lines Lock network retransmission queue in all places that it is used. ........ 2006-12-15 18:37 +0000 [r48495-48501] Luigi Rizzo * main/manager.c: unbreak the output for http session. Not long ago i replaced lseek() with fseek() but forgot that filr FILE's you need ftell to give you the current position. * main/channel.c, include/asterisk/channel.h: remove ast_safe_string_alloc() - it is completely equivalent to asprintf(). * channels/chan_zap.c: replace ast_safe_string_alloc() with asprintf() * channels/chan_features.c: replace ast_safe_string_alloc() with asprintf() * include/asterisk/threadstorage.h: small documentation improvements. 2006-12-15 13:36 +0000 [r48485-48491] Olle Johansson * main/tdd.c, include/asterisk/tdd.h: Doxygen changes * /, channels/chan_sip.c: Issue #8592 - treat 504 as congestion (imported from 1.2/1.4) * /, channels/chan_sip.c: Update to latest IANA specs 2006-12-15 06:34 +0000 [r48479-48480] Joshua Colp * include/asterisk/lock.h: Add support to see what holds the lock when doing trylock. * /, channels/chan_iax2.c: Merged revisions 48478 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48478 | file | 2006-12-15 01:28:05 -0500 (Fri, 15 Dec 2006) | 2 lines Use a wakeup variable so that we don't wait on IO indefinitely if packets need to be retransmitted. ........ 2006-12-15 04:03 +0000 [r48476-48477] Luigi Rizzo * main/channel.c, include/asterisk/channel.h: constify ast_state2str() and note it is not reentrant. * main/pbx.c, include/asterisk/channel.h: remove the macro LOAD_OH and expand it inline in the only place where it was used. 2006-12-14 17:39 +0000 [r48462-48473] Joshua Colp * /, include/asterisk/rtp.h, main/rtp.c: Merged revisions 48472 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48472 | file | 2006-12-14 12:36:12 -0500 (Thu, 14 Dec 2006) | 2 lines Payload values on the RTP structure can change AFTER a bridge has started. This comes from the packet handling of the SIP response when indication that it was answered has been sent. Therefore we need to protect this data with a lock when we read/write. (issue #8232 reported by tgrman) ........ * /, main/rtp.c: Merged revisions 48461 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48461 | file | 2006-12-13 22:33:30 -0500 (Wed, 13 Dec 2006) | 2 lines Remove direct RTCP bridging. I've come to the conclusion that we should handle this through the core and not just forward it on. Should solve a few bugs. ........ 2006-12-13 23:08 +0000 [r48458-48459] Luigi Rizzo * main/pbx.c: make sure that showdialplan sends only one 'Response: Success ' message even in case of a recursive call. * main/pbx.c: clean up function manager_show_dialplan_helper() reducing indentation and normalizing loops. While doing this, remove some unused variables, fix an uninitialized string (idaction), and mark some places where the behaviour is not what we would expect (e.g. an empty context is reported as an error same as a non-existent one). Given that this function is not in 1.4, the above can be changed without too many backward compatibility concerns. Not applicable to 1.4 or below. 2006-12-13 21:23 +0000 [r48455] Matt O'Gorman * codecs/codec_zap.c: support for deactivating translation paths that are no longer available and more descriptive show transcoder cli command. 2006-12-13 00:56 +0000 [r48433] Russell Bryant * channels/chan_zap.c: revert check for a zaptel transcoder related definition that was done in the wrong module. 2006-12-12 23:28 +0000 [r48432] Kevin P. Fleming * /, build_tools/prep_tarball: Merged revisions 48427 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48427 | kpfleming | 2006-12-12 17:18:14 -0600 (Tue, 12 Dec 2006) | 2 lines when making a release, we can always use wget and we can't run the configure script to find that out... ........ 2006-12-12 22:32 +0000 [r48416-48417] Russell Bryant * include/asterisk/app.h, channels/chan_sip.c, include/asterisk/channel.h, include/asterisk/pbx.h: Fix various spelling mistakes in comments. * channels/chan_zap.c: Make chan_zap inform you that your version of zaptel is too old instead of just failing to compile. It seems like the proper way to do this would be in the configure script. However, that wouldn't help existing checkouts unless we forced the configure script to be executed after any code was changed. 2006-12-12 19:55 +0000 [r48415] Matt O'Gorman * codecs/codec_zap.c: fixed nubb error on my part, transcoder now unlocks and locks correctly, as well as counts in the correct direction. 2006-12-12 10:36 +0000 [r48408-48410] Luigi Rizzo * main/manager.c: properly initialize a malloc'ed buffer * main/manager.c: normalize the scanning of "general" options in the config file. * main/cli.c: Make sure tab-completion works even when we have typed a fully matching word (e.g. "sip"); this is implemented by this one-line change - for (;; dst++, src += n) { + for (;src < argindex; dst++, src += n) { However this code is not exactly trivial to understand, so i am also adding some comments to help figuring out what it does. 2006-12-12 04:14 +0000 [r48402] Joshua Colp * /, apps/app_voicemail.c: Merged revisions 48401 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48401 | file | 2006-12-11 23:13:48 -0500 (Mon, 11 Dec 2006) | 2 lines Use S_OR in my previous app_voicemail. This is the way it should have been done. ........ 2006-12-11 23:02 +0000 [r48397-48400] Matt O'Gorman * /, sounds/Makefile: Merged revisions 48399 via svnmerge from https://svn.digium.com/svn/asterisk/branches/1.4 ........ r48399 | mogorman | 2006-12-11 17:02:10 -0600 (Mon, 11 Dec 2006) | 2 lines new sounds package with 100% more silence ........ * /, apps/app_externalivr.c: Merged revisions 48396 via svnmerge from https://svn.digium.com/svn/asterisk/branches/1.4 ................ r48396 | mogorman | 2006-12-11 16:11:35 -0600 (Mon, 11 Dec 2006) | 12 lines Merged revisions 48394 via svnmerge from https://svn.digium.com/svn/asterisk/branches/1.2 ........ r48394 | mogorman | 2006-12-11 15:55:43 -0600 (Mon, 11 Dec 2006) | 4 lines app_externalivr needs a real silence file, and additional changes to add silence files into core instead of extra patch provided by bug 8177 with minor additions. ........ ................ 2006-12-11 21:35 +0000 [r48392] Joshua Colp * /, apps/app_voicemail.c: Merged revisions 48391 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48391 | file | 2006-12-11 16:31:23 -0500 (Mon, 11 Dec 2006) | 2 lines Return non-existant callerid handling to that which it was before. In 1.4 and trunk callerid can be allocated but not have any contents so we have to use ast_strlen_zero before passing it to the relevant functions. (issue #8567 reported by pabelanger) ........ 2006-12-11 21:04 +0000 [r48390] Matt O'Gorman * codecs/codec_zap.c: add support for dynamic channel creation and destruction, and show transcoder to show number of channels in use. 2006-12-11 18:11 +0000 [r48389] Luigi Rizzo * main/manager.c: make sure the argument to ast_malloc() is > 0. Long explaination: The behaviour of the underlying malloc(0) differs depending on the operating system. Some return NULL (SysV behaviour); some still allocate a small chunk of memory and return a valid pointer (e.g. traditional BSD); some (e.g. FreeBSD 6.x) return a non-null pointer that causes a memory fault if used, even just for reading. Given the above variety, better never call malloc(0). 2006-12-11 17:00 +0000 [r48388] Steve Murphy * main/app.c: This update fixes the problem reported in bug 8551; that ast_app_getdata() behaves differently in trunk for the case of a null prompt. 2006-12-11 05:40 +0000 [r48384] Tilghman Lesher * /, funcs/func_strings.c: Merged revisions 48382 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48382 | tilghman | 2006-12-10 23:37:09 -0600 (Sun, 10 Dec 2006) | 2 lines STRFTIME() does not actually require an argument (issue 8540) ........ 2006-12-11 05:38 +0000 [r48378-48383] Joshua Colp * /, main/rtp.c: Merged revisions 48381 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48381 | file | 2006-12-11 00:36:45 -0500 (Mon, 11 Dec 2006) | 2 lines Merge in my latest RTP changes. Break out RTP and RTCP callback functions so they no longer share a common one. ........ * /, apps/app_meetme.c: Merged revisions 48379 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48379 | file | 2006-12-11 00:30:01 -0500 (Mon, 11 Dec 2006) | 2 lines Use the correct API call to say a device state changed. (Yes, I'm a nub.) ........ * /, apps/app_meetme.c: Merged revisions 48377 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48377 | file | 2006-12-10 23:57:38 -0500 (Sun, 10 Dec 2006) | 2 lines Don't access the conference structure after it has been freed. ........ 2006-12-11 00:52 +0000 [r48376] Tilghman Lesher * apps/app_nbscat.c, /, apps/app_festival.c, apps/app_mp3.c, res/res_agi.c, apps/app_zapras.c, apps/app_externalivr.c, apps/app_ices.c, res/res_musiconhold.c: Merged revisions 48375 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r48375 | tilghman | 2006-12-10 18:47:21 -0600 (Sun, 10 Dec 2006) | 13 lines Merged revisions 48374 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48374 | tilghman | 2006-12-10 18:33:59 -0600 (Sun, 10 Dec 2006) | 5 lines When doing a fork() and exec(), two problems existed (Issue 8086): 1) Ignored signals stayed ignored after the exec(). 2) Signals could possibly fire between the fork() and exec(), causing Asterisk signal handlers within the child to execute, which caused nasty race conditions. ........ ................ 2006-12-10 03:14 +0000 [r48373] Steve Murphy * channels/chan_zap.c, /: Merged revisions 48372 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r48372 | murf | 2006-12-09 20:04:18 -0700 (Sat, 09 Dec 2006) | 9 lines Merged revisions 48371 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48371 | murf | 2006-12-09 19:14:13 -0700 (Sat, 09 Dec 2006) | 1 line This version applies the patch suggested by stevens in bug 7836 (make inbound channel RINGING state consistent with other channels). ........ ................ 2006-12-09 16:44 +0000 [r48359-48365] Russell Bryant * channels/chan_iax2.c: convert the thread IO state and type to use enums. * /, channels/chan_iax2.c: Merged revisions 48363 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48363 | russell | 2006-12-09 10:59:42 -0500 (Sat, 09 Dec 2006) | 8 lines Use locking when accessing the registrations list. This list is not actually used very often, so the likelihood of there being a problem is pretty small, but still possible. For example, if the CLI command to list the registrations was called at the same time that a reload was occurring and the registrations list was getting destroyed and rebuilt, a crash could occur. In passing, go ahead and convert this list to use the linked list macros. ........ * channels/chan_iax2.c: chan_iax2 has an extremely large function, socket_process(), to handle incoming frames. The function, before this commit, was roughly 1400 lines long. So, I am working on breaking this up into functions so that the code is easier to follow and debug. Also, I will be committing these changes in chunks as I do them to ease tracking down any potentially introduced problems. Break out roughly 150 lines from socket_process() and introduce a new function, socket_process_meta() which handles the parsing of an incoming meta frame. Also, restructure some of this code a bit to reduce the deep nesting that was in this code. * channels/chan_iax2.c: - Fix a few spelling mistakes - Use sizeof() to pass an array size to a function - Use a single bit for a variable in the chan_iax2_pvt stuct since that is all it needs. - Add some comments about the iaxs, iaxl, and lastused arrays. 2006-12-07 18:21 +0000 [r48358] Russell Bryant * /, res/res_musiconhold.c: Merged revisions 48357 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r48357 | russell | 2006-12-07 13:17:28 -0500 (Thu, 07 Dec 2006) | 11 lines Merged revisions 48356 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48356 | russell | 2006-12-07 13:14:13 -0500 (Thu, 07 Dec 2006) | 3 lines Ensure that the file position is not incremented beyond the total number of files available for playback. (issue #8539, ulogic) ........ ................ 2006-12-07 16:42 +0000 [r48351] Luigi Rizzo * include/asterisk/http.h, main/manager.c, main/http.c, configs/manager.conf.sample: - Generalize the function ssl_setup() so that the certificate info are passed as an argument. - Update the code in main/http.c to use the new interface (the diff is large but mostly mechanical, due to the name change of several variables); - And since now it is trivial, implement "AMI over TLS", and document the possible options in manager.conf - And since the test client (openssl s_client -connect host:port ) does not generate \r\n as a line terminator, make get_input() also accept just a \n as a line terminator (Mac users: do you also need the \r-only version ?) The option parsing in manager.conf is not very efficient, and needs to be cleaned up and made similar to what we have in http.conf 2006-12-07 16:03 +0000 [r48350] Steve Murphy * main/manager.c, /: Merged revisions 47986,47995,47997,48001,48003-48004,48008-48014,48016,48018-48019 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r47986 | oej | 2006-11-24 07:00:19 -0700 (Fri, 24 Nov 2006) | 6 lines Doxygen update - Document cause codes - Document a bit more on channel variables - global, predefined and local - Fix some doxygen in channel.h. Adding one comment for two definitions does not work. They won't be copied to each. ................ r47995 | murf | 2006-11-24 10:40:49 -0700 (Fri, 24 Nov 2006) | 1 line This fix inspired by a patch supplied in bug 8189, which points out problems with the PLC code ................ r47997 | murf | 2006-11-24 11:17:25 -0700 (Fri, 24 Nov 2006) | 1 line removed the svnmerge-integrated property from trunk; it's confusing svnmerge in newly created branches ................ r48001 | rizzo | 2006-11-25 02:02:42 -0700 (Sat, 25 Nov 2006) | 5 lines set pointers to NULL after freeing memory to avoid multiple free() probably 1.4/1.2 issue as well if someone can look into that. ................ r48003 | oej | 2006-11-25 02:45:57 -0700 (Sat, 25 Nov 2006) | 9 lines - Adding comment on suspicious memory allocation. Seems like it's never freed, but I don't have a clear understanding of the frame allocation/deallocation, so I just mark this for investigation. (Reported by Ed Guy). We're trying to see if a free() hurts... - Doxygen comments on p2p rtp bridge stuff. I am a bit worried about shortcutting rtcp this way, but will need feedback from rtcp gurus. This should work for video calls too, and possibly UDPTL. ................ r48004 | oej | 2006-11-25 02:48:30 -0700 (Sat, 25 Nov 2006) | 2 lines Changing ERROR to lesser level. Imported from 1.2/1.4 ................ r48008 | rizzo | 2006-11-25 10:37:04 -0700 (Sat, 25 Nov 2006) | 7 lines generalize a bit the functions used to create an tcp socket and then run a service on it. The code in manager.c does essentially the same things, so we will be able to reuse the code in here (probably moving it to netsock.c or another appropriate library file). ................ r48009 | mattf | 2006-11-25 13:30:04 -0700 (Sat, 25 Nov 2006) | 1 line Updates to show linkset command ................ r48010 | mattf | 2006-11-25 13:54:38 -0700 (Sat, 25 Nov 2006) | 2 lines Add ss7 show linkset command ................ r48011 | mattf | 2006-11-25 14:32:33 -0700 (Sat, 25 Nov 2006) | 1 line Make sure we don't send a group reset on a group larger than 32 CICs ................ r48012 | mattf | 2006-11-25 14:35:23 -0700 (Sat, 25 Nov 2006) | 1 line bug fix ................ r48013 | mattf | 2006-11-25 14:46:58 -0700 (Sat, 25 Nov 2006) | 1 line Make compiler happier ................ r48014 | mattf | 2006-11-25 14:50:42 -0700 (Sat, 25 Nov 2006) | 1 line Little fix so we use the right message ................ r48016 | murf | 2006-11-25 17:15:42 -0700 (Sat, 25 Nov 2006) | 9 lines Merged revisions 48015 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48015 | murf | 2006-11-25 17:01:34 -0700 (Sat, 25 Nov 2006) | 1 line A little bit of func_cdr documentation upgrade-- no bug# involved, although 8221 may have inspired it. ........ ................ r48018 | murf | 2006-11-25 17:31:13 -0700 (Sat, 25 Nov 2006) | 9 lines Merged revisions 48017 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48017 | murf | 2006-11-25 17:26:16 -0700 (Sat, 25 Nov 2006) | 1 line might as well also document the raw values of the flag vars ........ ................ r48019 | russell | 2006-11-25 23:55:33 -0700 (Sat, 25 Nov 2006) | 6 lines - Add some comments on thread storage with a brief explanation of what it is as well as what the motivation is for using it. - Add a comment by the declaration of ast_inet_ntoa() noting that this function is not reentrant, and the result of a previous call to the function is no longer valid after calling it again. ................ 2006-12-06 20:46 +0000 [r48332-48338] Luigi Rizzo * main/manager.c: remove duplicated code to start the server threads, use the infrastructure exposed in http.c earlier today. As a bonus, now we can restart the session on a different port just reloading the module. On passing, fix a bug in the handling of 'enabled' in the configuration file - previously, a missing "enabled=" line in manager.conf meant "whatever the state was before" instead of a specific value. * main/manager.c: Part of the transformations necessary to add TLS support, as described in http://lists.digium.com/pipermail/asterisk-dev/2006-December/025213.html In detail, this commit does the following: b) change the function get_input() to use fread() instead of read() to collect the data. One can still do the ast_wait_for_input() on the original descriptor returned by accept(). c) change the function send_string() to work on the FILE *. As a side effect, this change now really guarantees that we don't spend more than "writetimeout" milliseconds on each line sent. d) modify the function action_command() so that it creates a temporary file descriptor to be passed to ast_cli_command(), and then read back the data from the temp file and write it to the output with send_string(). The code is similar to what is done in generic_http_callback() to support AMI-over-HTTP. 2006-12-06 16:54 +0000 [r48327] Olle Johansson * /, channels/chan_sip.c: Handle multiple 487's correctly 2006-12-06 16:19 +0000 [r48325] Russell Bryant * configs/iax.conf.sample, /: Merged revisions 48323 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r48323 | russell | 2006-12-06 11:15:45 -0500 (Wed, 06 Dec 2006) | 11 lines Merged revisions 48322 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48322 | russell | 2006-12-06 11:05:54 -0500 (Wed, 06 Dec 2006) | 3 lines Fix the name of the rtignoreregexpire option in the sample configuration file. (issue #8526, arkadia) ........ ................ 2006-12-06 16:17 +0000 [r48324] Luigi Rizzo * include/asterisk/http.h, main/http.c: Make externally visible some generic code useful to create and implement services over tcp and/or tcp-tls. This commit is nothing more than moving structure definitions (and documentation) from main/http.c to include/asterisk/http.h (temporary location until we find a better place), and removing the 'static' qualifier from server_root() and server_start(). The name change (adding the ast_ prefix as a minimum, and then possibly a more meaningful name) is postponed to future commits. Does not apply to other versions of asterisk. 2006-12-06 12:34 +0000 [r48318] Olle Johansson * /, channels/chan_sip.c: Don't send Contact in SIP Messages (imported from 1.2/1.4). Reported by Gunnar at Omnitor. 2006-12-06 07:39 +0000 [r48299-48307] Russell Bryant * apps/app_osplookup.c, apps/app_meetme.c, apps/app_queue.c, apps/app_voicemail.c: Resolve some pointer signedness compiler warnings in app_osplookup, and constify a bunch of usage strings for CLI commands. * channels/chan_local.c, channels/chan_skinny.c, channels/chan_agent.c, channels/chan_features.c, channels/chan_alsa.c, channels/iax2-provision.c, channels/chan_gtalk.c, channels/chan_oss.c, channels/chan_mgcp.c: Constify a bunch of usage strings for CLI commands. * res/res_config_pgsql.c, res/res_limit.c, res/res_agi.c, res/res_crypto.c, res/res_realtime.c, res/res_jabber.c, res/res_odbc.c: Constify a bunch of usage strings for CLI commands. * main/channel.c, main/udptl.c, main/frame.c, main/translate.c, main/file.c, pbx/pbx_dundi.c, main/db.c, main/rtp.c: Staticize one, and Constify a bunch of usage strings for CLI commands. * channels/chan_zap.c, channels/chan_sip.c, channels/chan_iax2.c, main/asterisk.c, main/cli.c: Constify a bunch of the usage strings for CLI commands. * channels/chan_iax2.c: Instead of creating an unused instance of an unnamed enum, give it a name. * include/asterisk/cli.h: Make the "usage" member of the ast_cli_entry struct const to resolve a compiler warning. 2006-12-05 20:46 +0000 [r48282] Joshua Colp * configure: Regenerate configure for Qwell's last commit. 2006-12-05 20:44 +0000 [r48280] Jason Parker * /, configure.ac: Merged revisions 48279 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48279 | qwell | 2006-12-05 14:42:52 -0600 (Tue, 05 Dec 2006) | 4 lines Fix curl version number testing to be much more friendly to non-bash shells. Issue 8508, patch by me. This *SHOULD* be POSIX compliant now.. ........ 2006-12-05 20:39 +0000 [r48277] Olle Johansson * include/asterisk/rtp.h, include/asterisk/channel.h, main/rtp.c: Doxygen updates 2006-12-05 20:15 +0000 [r48276] Jason Parker * main/tdd.c, include/asterisk/fskmodem.h, main/callerid.c, main/fskmodem.c: Expand on r48273 (from issue 8506), to translate more of the fskmodem stuff to English. r48273 dealt with the comments and such, this deals with the code itself. (This couldn't have been easily done if it weren't for 48273 - thanks again for that merbanan) 2006-12-05 19:41 +0000 [r48269-48273] Olle Johansson * include/asterisk/fskmodem.h, main/fskmodem.c: Issue #8506 - translate spanish comments in fskmodem to english (according to bug guidelines) Thanks merbanan! * /: Blocking invitestate patch that is already merged to svn trunk. * /, configs/sip.conf.sample: Adding docs on t.38 2006-12-05 14:33 +0000 [r48266] TransNexus OSP Development * apps/app_osplookup.c: 1. Change to remove the compiling warning: "app_osplookup.c:2169: warning: initialization discards qualifiers from pointer target type" 2006-12-05 11:09 +0000 [r48258-48259] Olle Johansson * main/frame.c, include/asterisk/frame.h, main/rtp.c: Well, yes... * main/frame.c, include/asterisk/frame.h, main/rtp.c: Reserving flags for coming code (currently in the "videocaps" branch) implementing T.140 support in RTP. T.140/RFC 4351 is TDD over IP - text telephony for hearing impaired. It defines a realtime text chat, much like the old "talk" application in Unix. T.140 is character by character in real time. It's not the same as our current MESSAGE format - that is more like IM, but can be gatewayed to MESSAGE with a text "codec" if needed. More patches will follow, as soon as we've separated this code from the video capabilities functions in the videocaps branch. Code by John Martin, Aupix (disclaimer on file) 2006-12-05 01:46 +0000 [r48253-48255] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 48254 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48254 | tilghman | 2006-12-04 19:41:02 -0600 (Mon, 04 Dec 2006) | 2 lines Oops, forgot to release the odbc handle ........ * /, apps/app_voicemail.c: Merged revisions 48252 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r48252 | tilghman | 2006-12-04 19:34:34 -0600 (Mon, 04 Dec 2006) | 14 lines Merged revisions 48251 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48251 | tilghman | 2006-12-04 19:26:08 -0600 (Mon, 04 Dec 2006) | 6 lines If the recording in the database is too large, it will fail to retrieve with an mmap error. Not too sure why this doesn't happen when we put it in the database, also, but since that doesn't seem to be broken, I'm not going to fix it (at least until someone reports it). Solution is to ask for the file in smaller chunks. (Bug 8385) ........ ................ 2006-12-04 21:49 +0000 [r48249] Jason Parker * /, apps/app_voicemail.c: Merged revisions 48248 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48248 | qwell | 2006-12-04 15:48:41 -0600 (Mon, 04 Dec 2006) | 2 lines Fix an issue which didn't allow unavail/greet/busy/etc messages from being saved into ODBC (and probably IMAP). ........ 2006-12-04 17:55 +0000 [r48229-48231] Jason Parker * /, configs/voicemail.conf.sample: Merged revisions 48230 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48230 | qwell | 2006-12-04 11:54:46 -0600 (Mon, 04 Dec 2006) | 4 lines Add documentation to voicemail.conf.sample for ODBC storage. Issue 8499 - patch by blitzrage. ........ * /, doc/snmp.txt: Merged revisions 48228 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48228 | qwell | 2006-12-04 11:43:24 -0600 (Mon, 04 Dec 2006) | 4 lines Attempt to document some of the dependencies that are needed for net-snmp Issue 8499 - initial patch by blitzrage. ........ 2006-12-03 06:35 +0000 [r48224] Russell Bryant * /, sounds/Makefile: Merged revisions 48223 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48223 | russell | 2006-12-03 01:34:14 -0500 (Sun, 03 Dec 2006) | 3 lines When "fetch" is in use, instead of "wget", --continue is not a valid option. (issue #8451) ........ 2006-12-02 22:03 +0000 [r48200-48220] Olle Johansson * /, channels/chan_sip.c: Cleaning up handle_response a bit. (Imported from 1.4) * .cleancount: Removing two .h files means we need to update cleancount to force make depend again (or ?) * channels/chan_sip.c: Send CANCEL to call with early media (PROGRESS INBAND). This is imported from branch "invitestate" and "invitestate-1.4" *** *** *** IF YOU HAVE ISSUES WITH BYEs/CANCELs - PLEASE UPDATE AND TEST AGAIN! *** Thank you! *** *** /Olle * channels/chan_sip.c: Invitestate updates * agi/Makefile: Oops. Something is wrong in the agi directory. Asking for autoconfig.h. I have it disabled locally, but no reason to commit that change. * apps/app_sms.c: Doxygenification * main/coef_out.h (removed), main/tdd.c, main/callerid.c, main/fskmodem.c, main/coef_in.h (removed): - Code formatting - remove coef_in.h and coef_out.h that was only included as data definitions in fskmodem.c If you understand spanish, please help us translate the comments in fskmodem.c. Thanks! * /, channels/chan_sip.c, include/asterisk/rtp.h, configs/sip.conf.sample, main/rtp.c: - Disable RTP timeouts during T.38 transmission - Encapsulate RTP timers to the RTP structure, so we have one set for video and one for audio - Document RTP keepalive configuration option - Cleanup and document the monitor support function to hangup on RTP timeouts - Add RTP keepalive to SIP show settings Imported from 1.4 with modifications for trunk. 2006-12-01 23:39 +0000 [r48194] Kevin P. Fleming * apps/app_dial.c, /: Merged revisions 48193 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r48193 | kpfleming | 2006-12-01 17:37:28 -0600 (Fri, 01 Dec 2006) | 10 lines Merged revisions 48192 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48192 | kpfleming | 2006-12-01 17:30:59 -0600 (Fri, 01 Dec 2006) | 2 lines if Dial() is going to send music-on-hold to the calling party, it has to send PROGRESS first to ensure that the reverse audio path has been setup first (BE-106) ........ ................ 2006-12-01 23:20 +0000 [r48191] Russell Bryant * Makefile, /, configure, configure.ac, makeopts.in, sounds/Makefile: Merged revisions 48190 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48190 | russell | 2006-12-01 18:16:28 -0500 (Fri, 01 Dec 2006) | 12 lines FreeBSD 6.1 does not include wget by default. However, it has fetch which will work just fine for our purposes of downloading the sounds packages. So, check for both wget and fetch and the configure script and use what was found to download them. If neither one was found, and sound packages are selected that must be downloaded, the install process will print out an informative error message indicating the situation. Also, fix a couple places where "make" was hard coded into some output messages by replacing them with the $(MAKE) variable. (issue #8451, initial patch by pabelanger, with additional modifications by me) ........ 2006-12-01 20:49 +0000 [r48188] Olle Johansson * main/channel.c: Formatting fix 2006-12-01 20:26 +0000 [r48187] Jason Parker * /, configs/extensions.conf.sample: Merged revisions 48186 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r48186 | qwell | 2006-12-01 14:25:51 -0600 (Fri, 01 Dec 2006) | 10 lines Merged revisions 48183 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48183 | qwell | 2006-12-01 14:19:10 -0600 (Fri, 01 Dec 2006) | 2 lines Fix a small typo - issue 8848, reported by pabelanger ........ ................ 2006-12-01 19:41 +0000 [r48180] Tilghman Lesher * /, main/cli.c: Merged revisions 48179 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48179 | tilghman | 2006-12-01 13:38:59 -0600 (Fri, 01 Dec 2006) | 2 lines Double-unlock error (reported by blitzrage on IRC) ........ 2006-12-01 18:16 +0000 [r48175-48178] Olle Johansson * /, channels/chan_sip.c, configs/sip.conf.sample: - Remove T.38 early media, since T.38 requires two way communication (imported from 1.4) - Small fixes to limitonpeer * include/asterisk/threadstorage.h: Tiny doxygen improvement 2006-11-30 21:22 +0000 [r48169] Joshua Colp * /, include/asterisk/rtp.h, channels/chan_gtalk.c, main/rtp.c: Merged revisions 48168 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48168 | file | 2006-11-30 16:18:24 -0500 (Thu, 30 Nov 2006) | 2 lines Do not do a partial bridge for Google Talk since we need to handle STUN. (issue #8448 reported by phsultan) ........ 2006-11-30 20:55 +0000 [r48164-48167] Olle Johansson * /, channels/chan_sip.c: Issue #8319 (imported from 1.2, 1.4) - Increment nonce-count properly (noriyuki) * include/asterisk/devicestate.h, main/pbx.c, main/devicestate.c, include/asterisk/channel.h, include/asterisk/pbx.h: Documentation updates 2006-11-30 20:29 +0000 [r48153-48163] Joshua Colp * /, channels/chan_iax2.c: Merged revisions 48158 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r48158 | file | 2006-11-30 15:07:55 -0500 (Thu, 30 Nov 2006) | 10 lines Merged revisions 48157 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48157 | file | 2006-11-30 15:06:43 -0500 (Thu, 30 Nov 2006) | 2 lines Only print out debug message if bridged channel is not NULL. (issue #8412 reported by jubilex) ........ ................ * /, res/res_features.c: Merged revisions 48155 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r48155 | file | 2006-11-30 14:05:14 -0500 (Thu, 30 Nov 2006) | 10 lines Merged revisions 48154 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48154 | file | 2006-11-30 14:04:11 -0500 (Thu, 30 Nov 2006) | 2 lines Do not listen for DTMF on the bridge that comes into existence when ParkedCall is executed. This means native bridging can now occur for this. (issue #8406 reported by kebl0155) ........ ................ * main/cdr.c, /: Merged revisions 48152 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r48152 | file | 2006-11-30 13:47:40 -0500 (Thu, 30 Nov 2006) | 10 lines Merged revisions 48151 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48151 | file | 2006-11-30 13:42:45 -0500 (Thu, 30 Nov 2006) | 2 lines Print certain CDR messages out at the NOTICE level versus WARNING since they can occur when used with the CDR applications and are perfectly fine. (issue #8367 reported by dartvader) ........ ................ 2006-11-30 18:25 +0000 [r48149-48150] Olle Johansson * main/devicestate.c: Small update * agi/Makefile, contrib/asterisk-ng-doxygen, agi/eagi-test.c, main/devicestate.c, agi/eagi-sphinx-test.c: Doxygen updates 2006-11-30 18:20 +0000 [r48144-48148] Joshua Colp * /, configs/sip.conf.sample: Merged revisions 48143 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r48143 | file | 2006-11-30 12:57:35 -0500 (Thu, 30 Nov 2006) | 10 lines Merged revisions 48142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48142 | file | 2006-11-30 12:55:23 -0500 (Thu, 30 Nov 2006) | 2 lines Document 'port' for SIP peers, came up because of the current mailing list thread. (issue #8450 reported by blitzrage) ........ ................ 2006-11-30 17:15 +0000 [r48130-48139] Olle Johansson * include/asterisk/doxyref.h, main/devicestate.c: Adding some generic docs on extension and device states - watchers and providers * doc/manager.txt, /: Add information on status events * /, channels/chan_sip.c: Merging patch from 1.2/1.4. I think this was originally spotted by Luigi, but hit me in the back today. 2006-11-30 03:29 +0000 [r48116-48123] Joshua Colp * channels/chan_sip.c: I am pretty sure that oej only meant to change the variable name in the source, not the configuration option name so let's turn it back to srvlookup instead of global_srvlookup. (issue #8442 reported by jtodd) * /, apps/app_voicemail.c: Merged revisions 48115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48115 | file | 2006-11-29 16:05:17 -0500 (Wed, 29 Nov 2006) | 2 lines Use MAILTMPLEN instead of sizeof in mm_login. (issue #8420 reported by slimey) ........ 2006-11-29 20:57 +0000 [r48111-48114] Olle Johansson * /, configs/sip.conf.sample: Clarify some settings for status reports in subscriptions, queues and manager * /, configs/sip.conf.sample: Explain RTP timeouts * main/rtp.c: Change logging for p2p rtp bridge mode 2006-11-29 17:59 +0000 [r48109-48110] Russell Bryant * include/asterisk/threadstorage.h: - Fix a few spelling mistakes. - Add some more documentation for the ast_dynamic_str_............() function to document the behavior of the function in the case of a partial write. Also, document the return value and note that the function should never need to be called directly. * main/utils.c: Go ahead and make this write unconditional. Making it conditional is more work in both the append and non-append modes. Also, always truncating the partial write makes the behavior of the function more consistent, where in any type of write, no partial result is left in the buffer. Thanks for the feedback, luigi 2006-11-29 16:53 +0000 [r48108] Joshua Colp * /, main/rtp.c: Merged revisions 48107 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r48107 | file | 2006-11-29 11:50:33 -0500 (Wed, 29 Nov 2006) | 10 lines Merged revisions 48106 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48106 | file | 2006-11-29 11:47:10 -0500 (Wed, 29 Nov 2006) | 2 lines If the frame was duplicated before writing out then we need to free it. (issue #8429 reported by edguy3) ........ ................ 2006-11-29 05:08 +0000 [r48103] Russell Bryant * main/utils.c: Remove an XXX command suggesting that this truncation should not be conditional, and also add a more verbose comment explaining why it is only needed in the case of appending to the string for any curious readers that come along in the future. 2006-11-29 04:28 +0000 [r48100-48102] Joshua Colp * /, apps/app_voicemail.c: Merged revisions 48101 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48101 | file | 2006-11-28 23:26:53 -0500 (Tue, 28 Nov 2006) | 2 lines Don't crash if the mailstream was not created. ........ * sounds/Makefile: Use the proper version of extra sounds. (issue #8441 reported by jtodd) 2006-11-28 23:13 +0000 [r48098-48099] Russell Bryant * channels/chan_iax2.c: Add a comment to note near some code that performs a very expensive operation that occurs for every incoming media frame. * codecs/codec_zap.c: resolve a couple of compiler warnings 2006-11-28 18:28 +0000 [r48096] Jason Parker * Makefile, /: Merged revisions 48095 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48095 | qwell | 2006-11-28 12:26:53 -0600 (Tue, 28 Nov 2006) | 2 lines Export several more variables in top level Makefile. Inspired by issue 8438. ........ 2006-11-28 17:08 +0000 [r48090] Luigi Rizzo * main/manager.c: don't use outputstr in the struct mansession, it's just an extra allocation on a path where we have way too many already. Unfortunately the AMI-over-HTTP requires multiple copies, because we need to generate a header, then the raw output to an intermediate buffer, then convert it to html/xml, and finally copy everything into a malloc'ed buffer because that's what the generic_http_callback interface expects. 2006-11-28 16:59 +0000 [r48089] Joshua Colp * channels/chan_phone.c, /: Merged revisions 48088 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r48088 | file | 2006-11-28 11:57:16 -0500 (Tue, 28 Nov 2006) | 10 lines Merged revisions 48087 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48087 | file | 2006-11-28 11:56:01 -0500 (Tue, 28 Nov 2006) | 2 lines According to the research I have done we never needed to include compiler.h in the first place so let's not! (issue #8430 reported by edguy3) ........ ................ 2006-11-28 15:53 +0000 [r48062-48086] Luigi Rizzo * main/manager.c: initialize the dynamic string in a sane way. * main/utils.c: some simplifications to ast_dynamic_str_thread_build_va_couldnt_we_choose_a_shorter_name() I am unsure whether the truncation of the string in case of a failed attempt should be done unconditionally. See the XXX mark. Russel, ideas ? * main/manager.c: do not return 500 Internal error if the AMI command provides no output. * main/manager.c: mosty comment and documentation cleanup on waitevent. * main/manager.c: Move the code to purge stale sessions to a function, to simplify the body of the main loop of the accepting thread. Rename purge_unused() to purge_events() so one knows what the function does. * main/manager.c: Various simplifications of the code: + use a wrapper around ast_carefulwrite(), used in two places, to make life easier when we decide to use a different interface to the socket. + put an ast_verbose() message on astman_append on a case that should never happen now that we use a temporary file for AMI-over-HTTP sessions + document and slightly simplify process_events() by removing unnecessary parentheses. + in get_input(), use ast_wait_for_input() instead of poll(). We may want to move to a completely non-blocking * main/manager.c: More informative message on invalid commands. * main/manager.c: another normalization of AMI vs HTTP identification. Should really define a macro IS_AMI(s) so it is clear what we want to do. * main/manager.c: always use managerid to determine whether this is an AMI or HTTP session, and document it. * main/http.c: In the previous commit i forgot to set the poll_timeout to -1, causing the http threads to do busy waiting around the socket... Fix the mistake, sorry for the inconvenience! * main/http.c: document the support for running a server on TCP/TLS and opening an SSL socket. We are almost ready to make this code available to other modules. * main/http.c, configs/http.conf.sample: add a new http.conf option, sslbindaddr. Because https is more secure than http, it usually makes sense to keep this service more open than the one on the unencrypted port. * main/http.c: in the helper thread, separate the FILE * creation from the actual function doing work on the socket. This is another generalization to provide a generic mechanism to open TCP/TLS socket with a thread managing the accpet and children threads managing the individual sessions. * main/http.c: staticize a global variable and remove an unused field structure. 2006-11-27 18:10 +0000 [r48056] Joshua Colp * /, apps/app_voicemail.c: Merged revisions 48054 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r48054 | file | 2006-11-27 13:06:50 -0500 (Mon, 27 Nov 2006) | 10 lines Merged revisions 48053 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48053 | file | 2006-11-27 13:03:57 -0500 (Mon, 27 Nov 2006) | 2 lines Use the proper function to get the new message count instead of always using the filesystem. (issue #8421 reported by slimey) ........ ................ 2006-11-27 17:31 +0000 [r48050] Tilghman Lesher * /, res/res_musiconhold.c: Merged revisions 48049 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r48049 | tilghman | 2006-11-27 11:20:37 -0600 (Mon, 27 Nov 2006) | 10 lines Merged revisions 48045 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48045 | tilghman | 2006-11-27 11:15:54 -0600 (Mon, 27 Nov 2006) | 2 lines Random MOH wasn't really random (bug 8381) ........ ................ 2006-11-27 15:48 +0000 [r48039-48040] Joshua Colp * pbx/pbx_spool.c: More fixes for referencing a structure after it has been freed. (issue #8425 reported by arkadia) * pbx/pbx_spool.c, /: Merged revisions 48038 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r48038 | file | 2006-11-27 10:32:19 -0500 (Mon, 27 Nov 2006) | 10 lines Merged revisions 48037 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48037 | file | 2006-11-27 10:30:37 -0500 (Mon, 27 Nov 2006) | 2 lines Do not reference the freed outgoing structure in the debug message. (issue #8425 reported by arkadia) ........ ................ 2006-11-27 14:47 +0000 [r48034] Luigi Rizzo * funcs/func_cdr.c: remove an extra comma in an initializer Detected by: AST_DEVMODE=yes 2006-11-27 06:59 +0000 [r48032-48033] Olle Johansson * include/asterisk/doxyref.h, include/asterisk/threadstorage.h: Doxygen updates * /, channels/chan_sip.c: Change error message (imported from 1.4) 2006-11-26 06:55 +0000 [r48019] Russell Bryant * include/asterisk/utils.h, include/asterisk/threadstorage.h: - Add some comments on thread storage with a brief explanation of what it is as well as what the motivation is for using it. - Add a comment by the declaration of ast_inet_ntoa() noting that this function is not reentrant, and the result of a previous call to the function is no longer valid after calling it again. 2006-11-26 00:31 +0000 [r48016-48018] Steve Murphy * /, funcs/func_cdr.c: Merged revisions 48017 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48017 | murf | 2006-11-25 17:26:16 -0700 (Sat, 25 Nov 2006) | 1 line might as well also document the raw values of the flag vars ........ * /, funcs/func_cdr.c: Merged revisions 48015 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48015 | murf | 2006-11-25 17:01:34 -0700 (Sat, 25 Nov 2006) | 1 line A little bit of func_cdr documentation upgrade-- no bug# involved, although 8221 may have inspired it. ........ 2006-11-25 21:50 +0000 [r48009-48014] Matthew Fredrickson * channels/chan_zap.c: Little fix so we use the right message * channels/chan_zap.c: Make compiler happier * channels/chan_zap.c: bug fix * channels/chan_zap.c: Make sure we don't send a group reset on a group larger than 32 CICs * channels/chan_zap.c: Add ss7 show linkset command * channels/chan_zap.c: Updates to show linkset command 2006-11-25 17:37 +0000 [r48008] Luigi Rizzo * main/http.c: generalize a bit the functions used to create an tcp socket and then run a service on it. The code in manager.c does essentially the same things, so we will be able to reuse the code in here (probably moving it to netsock.c or another appropriate library file). 2006-11-25 09:48 +0000 [r48003-48004] Olle Johansson * /, channels/chan_sip.c: Changing ERROR to lesser level. Imported from 1.2/1.4 * main/rtp.c: - Adding comment on suspicious memory allocation. Seems like it's never freed, but I don't have a clear understanding of the frame allocation/deallocation, so I just mark this for investigation. (Reported by Ed Guy). We're trying to see if a free() hurts... - Doxygen comments on p2p rtp bridge stuff. I am a bit worried about shortcutting rtcp this way, but will need feedback from rtcp gurus. This should work for video calls too, and possibly UDPTL. 2006-11-25 09:02 +0000 [r48001] Luigi Rizzo * main/channel.c: set pointers to NULL after freeing memory to avoid multiple free() probably 1.4/1.2 issue as well if someone can look into that. 2006-11-24 18:17 +0000 [r47995-47997] Steve Murphy * /: removed the svnmerge-integrated property from trunk; it's confusing svnmerge in newly created branches * /, main/translate.c: This fix inspired by a patch supplied in bug 8189, which points out problems with the PLC code 2006-11-24 14:00 +0000 [r47986] Olle Johansson * include/asterisk/doxyref.h, main/pbx.c, include/asterisk/causes.h, include/asterisk/channel.h: Doxygen update - Document cause codes - Document a bit more on channel variables - global, predefined and local - Fix some doxygen in channel.h. Adding one comment for two definitions does not work. They won't be copied to each. 2006-11-23 11:04 +0000 [r47957-47960] Olle Johansson * /, channels/chan_sip.c: Remove unused memory allocation * doc/asterisk-conf.txt: Document new configuration option. 2006-11-22 21:49 +0000 [r47933-47945] Joshua Colp * /, main/rtp.c: Merged revisions 47944 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47944 | file | 2006-11-22 16:47:43 -0500 (Wed, 22 Nov 2006) | 2 lines Video will never reach Packet2Packet bridging and can do more harm then good. ........ * CHANGES: Clarify a bit more. * CHANGES: Need to update the CHANGES file as well for the maxfiles option. * main/asterisk.c: Add support to set the maximum number of files open when Asterisk loads using the 'maxfiles' configuration option. (issue #7499 reported by rkarlsba) 2006-11-22 11:28 +0000 [r47923] Olle Johansson * channels/chan_h323.c: Don't over-deprecate... :-) 2006-11-22 05:49 +0000 [r47912] Mark Spencer * main/manager.c: Restore some sense of security to manager 2006-11-21 17:34 +0000 [r47898] Joshua Colp * /, main/rtp.c: Merged revisions 47897 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47897 | file | 2006-11-21 12:32:27 -0500 (Tue, 21 Nov 2006) | 2 lines If we have the non standard G726-32 setting turned on we want to return G726-32 to the SDP, not our AAL2 string. (issue #8330 reported by voipgate) ........ 2006-11-21 15:25 +0000 [r47893] Olle Johansson * /, channels/chan_sip.c: Treat 101 as 100, not 183 session progress 2006-11-21 11:53 +0000 [r47880-47881] Luigi Rizzo * apps/app_dial.c: better fix for the previous bug. In general this code needs a deep revision, because the body of do_forward() deletes/overwrites the output channel without freeing the resouce in some cases, and without notifying the caller. Also, on FreeBSD with MALLOC_OPTIONS set i am seeing various panics (duplicate freee etc.) * apps/app_dial.c: do not ast_hangup() on a NULL channel. In the original code this would happen in the case of o->forwards >= AST_MAX_FORWARDS Likely an 1.2/1.4 isse as well - please someone have a look, while I am hunting a few more similar panics now. 2006-11-20 20:04 +0000 [r47866] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 47864-47865 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47864 | tilghman | 2006-11-20 14:01:58 -0600 (Mon, 20 Nov 2006) | 2 lines Oops, merge missed release of odbc object ........ ........ 2006-11-20 19:52 +0000 [r47851-47861] Joshua Colp * main/frame.c, /: Merged revisions 47860 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r47860 | file | 2006-11-20 14:51:36 -0500 (Mon, 20 Nov 2006) | 10 lines Merged revisions 47859 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47859 | file | 2006-11-20 14:50:21 -0500 (Mon, 20 Nov 2006) | 2 lines Don't forget to byte swap if we are exiting the smoother feed early. (issue #8287 reported by arturs) ........ ................ * main/rtp.c: Use RTP/RTCP fds on the RTP structure, don't bother storing them. * /, main/rtp.c: Merged revisions 47852 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47852 | file | 2006-11-20 10:58:50 -0500 (Mon, 20 Nov 2006) | 2 lines Only remove/destroy the RTCP I/O item if it exists. ........ * apps/app_dial.c, /, apps/app_directed_pickup.c, include/asterisk/channel.h, .cleancount: Merged revisions 47850 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47850 | file | 2006-11-20 10:51:37 -0500 (Mon, 20 Nov 2006) | 2 lines Use a separate variable in the channel structure to store the context that the channel was dialed from. (issue #8382 reported by jiddings) ........ 2006-11-20 14:08 +0000 [r47847] Steve Murphy * /: Erased the svnmerge-integrated prop from trunk. Please, in your svnmerge-ings, don't let these props leak into the trunk or branches. 2006-11-20 11:46 +0000 [r47844-47846] Olle Johansson * /, configs/sip.conf.sample: Update docs for videosupport * /, channels/chan_sip.c: Properly reset schedule items (rizzo) 2006-11-19 04:22 +0000 [r47835-47836] Steve Murphy * UPGRADE.txt: Added a few words to explain the change to AEL concerning Gosub() * doc/ael.txt: Added a few words of explanation about macros 2006-11-18 22:14 +0000 [r47822-47834] Luigi Rizzo * main/manager.c: comments-only change: document a bit more when manager events are delivered to the clients. * main/cdr.c, res/res_features.c, res/res_realtime.c: ESS-ification. no need to bring this in 1.4, it is just code cleanup * include/asterisk/cli.h, main/cli.c: Move this macro from cli.c to cli.h so apps can use it without duplicating the macro or the code: /*! * In many cases we need to print singular or plural * words depending on a count. This macro helps us e.g. * printf("we have %d object%s", n, ESS(n)); */ #define ESS(x) ((x) == 1 ? "" : "s") * /, channels/chan_sip.c: Merged revisions 47823 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47823 | rizzo | 2006-11-18 18:59:35 +0100 (Sat, 18 Nov 2006) | 5 lines fix bug 7450 - Parsing fails if From header contains angle brackets (the bug was only in a corner case where the < was right after the opening quote, and the fix is trivial). ........ * channels/chan_oss.c: prevent the sound thread from consuming all the available CPU doing busy-wait on the output audio device. As it is set now, it tries to push a frame every 10ms, which is still too frequent but avoids deep restructuring of the code (which i should do, though). Note, this is only for ring tones, regular audio coming from the network is still delivered as soon as it is available. Eventually this could well end up in the 1.4 branch, but since i am probably the only user of chan_oss there isn't much urgency to do that. 2006-11-17 23:18 +0000 [r47821] Steve Murphy * include/asterisk/file.h, main/channel.c, res/res_features.c, main/file.c, main/app.c, apps/app_directory.c, apps/app_followme.c, apps/app_voicemail.c: This update fulfils the request of bug 7109, which claimed the language arg to ast_stream_and_wait() was redundant. Almost all calls just used chan->language, and seeing how chan is the first argument, this certainly seems redundant. A change of language could just as easily be done by simply changing the channel language before calling. 2006-11-17 22:56 +0000 [r47815-47818] Luigi Rizzo * main/cli.c: remove a debugging message * main/cli.c: convert "help" to new style, fix completion of arguments past the first one that i broke earlier today. * main/cli.c: standardize "module show [like]" 2006-11-17 21:51 +0000 [r47814] Jason Parker * configs/voicemail.conf.sample, apps/app_voicemail.c: Add ability to notify an external application/script that the voicemail password was, while also still changing the password "internally". Issue 7371, initial patch by pdunkel, with rewrite/config comments by me. Additional modifications (yay bitmask) by pdunkel. 2006-11-17 21:50 +0000 [r47813] Luigi Rizzo * main/cli.c: describe a bit the patterns that you can have in the commands, and add support for wildcard (spelled as '%'). On passing fix a bug in the expansion code which was hidden and appeared when implementing the wildcard The fix is just the line 'src != argindex', in case someone wants to test this on 1.4 - but i would just keep this in trunk. 2006-11-17 20:46 +0000 [r47806] Jason Parker * apps/app_queue.c: Add ability to add custom queue log via manager interface. Issue 7806, patch by alexrch, with slight modifications by me. 2006-11-17 18:26 +0000 [r47801] Matthew Fredrickson * channels/chan_zap.c: Add some sense of link state. Don't make calls if link is down. Only reset if we're bringing the link up for the first time. 2006-11-17 12:26 +0000 [r47787-47790] Luigi Rizzo * main/cli.c: merge the implemenmtation of "core set debug" and "core set verbose". No externally visible changes. * channels/chan_oss.c: remove an unused function * channels/chan_oss.c: use the regexp cli support on some of the command * include/asterisk/cli.h, main/cli.c: introduce a bit of regexp support in the internal CLI api. Now you can specify a cli command as "console autoanswer [on|off]" which means the on|off argument is optional, or "console {mute|unmute}" which means the mute|unmute argument is mandatory. The blocks in [] or {} do not necessarily need to be at the end of the string. Completions for the variant parts are generated automatically. This should significantly simplify the implementation of the various handlers. 2006-11-17 01:05 +0000 [r47784] Matthew Fredrickson * channels/chan_zap.c: Make sure we choose the last channel as the dchannel if it's not E1 (for BRI). (#8077) Thanks Tzafrir. 2006-11-16 23:20 +0000 [r47783] Jason Parker * apps/app_dial.c, /, apps/app_db.c: Merged revisions 47782 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47782 | qwell | 2006-11-16 17:19:46 -0600 (Thu, 16 Nov 2006) | 2 lines Fix a couple of typos. Initially pointed out by mrobinson. ........ 2006-11-16 23:05 +0000 [r47779] Luigi Rizzo * channels/chan_oss.c: convert two entries to new style 2006-11-16 23:00 +0000 [r47778] Kevin P. Fleming * /, doc/billing.txt: Merged revisions 47777 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r47777 | kpfleming | 2006-11-16 17:00:10 -0600 (Thu, 16 Nov 2006) | 12 lines update documentation regarding IAX2 transfers and CDRs Merged revisions 47776 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47776 | kpfleming | 2006-11-16 16:57:31 -0600 (Thu, 16 Nov 2006) | 2 lines update clearly wrong documentation regarding cdr_custom ........ ................ 2006-11-16 22:51 +0000 [r47775] Jason Parker * channels/chan_zap.c: Remove the interim variable for range modifications, and set it on the structure directly. Also move the default checking to where it gets set initially. Fixes suggested by file. 2006-11-16 22:44 +0000 [r47772] Luigi Rizzo * channels/chan_oss.c: convert some handlers to new style. 2006-11-16 22:32 +0000 [r47771] Jason Parker * channels/chan_zap.c, configs/zapata.conf.sample: Add ability to modify range for dring matching. Issue #8369, patch by ssuehring, modified slightly by me. 2006-11-16 22:03 +0000 [r47769-47770] Luigi Rizzo * channels/chan_oss.c: fix indentation * main/cli.c: remove an unused function 2006-11-16 21:13 +0000 [r47763-47765] Joshua Colp * /, channels/chan_sip.c: Merged revisions 47764 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47764 | file | 2006-11-16 16:11:06 -0500 (Thu, 16 Nov 2006) | 2 lines Compare technology using the pointers instead of a straight comparison based on name. (issue #8228 reported by dean bath) ........ 2006-11-16 20:10 +0000 [r47759] Kevin P. Fleming * /, configure, configure.ac: Merged revisions 47758 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47758 | kpfleming | 2006-11-16 14:09:10 -0600 (Thu, 16 Nov 2006) | 2 lines check for pre-1.4 versions of Zaptel and abort the configure script if found with an appropriate error message ........ 2006-11-16 19:29 +0000 [r47756] Olle Johansson * /, channels/chan_sip.c, configs/sip.conf.sample: Make it possible to enable/disable onhold tracking, in order to make life easier for realtime users. 2006-11-16 18:32 +0000 [r47747-47752] Joshua Colp * channels/chan_local.c, /: Merged revisions 47751 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r47751 | file | 2006-11-16 13:29:12 -0500 (Thu, 16 Nov 2006) | 10 lines Merged revisions 47750 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47750 | file | 2006-11-16 13:26:50 -0500 (Thu, 16 Nov 2006) | 2 lines Because of the way chan_local is written we should be extra careful and make sure our callback functions have a tech_pvt. (issue #8275 reported by mflorell) ........ ................ * /, apps/app_meetme.c: Merged revisions 47748 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47748 | file | 2006-11-16 12:52:48 -0500 (Thu, 16 Nov 2006) | 2 lines Don't unreference the SLA object if there is no SLA object in the devicestate callback. (issue #8354 reported by loloski) ........ * /: Be gone 1.2 props! 2006-11-16 17:15 +0000 [r47734-47746] Olle Johansson * /: Merging a fix that was already fixed. * channels/chan_sip.c: Merging implementation of invite states from my "invitestate" branch for 1.2. This is a bit more clean platform for the handling of BYE/CANCEL than what we had. It might also need to changes in other parts of the code, since we know the state of the INVITE transaction. Please observe that this is is not dialog states at all, this is INVITE transaction states. Hello Michael Proctor, and thank you! :-) * /: Block upgrade to UPGRADE * /, channels/chan_sip.c, configs/sip.conf.sample: - CANCEL never uses authentication - Add docs on canreinvite 2006-11-16 14:58 +0000 [r47727-47732] Luigi Rizzo * main/cli.c: reduce indentation on a large function. * main/cli.c: use atomic instructions to update the inuse counters for CLI entriesC. The lock is not protecting this field. I wonder if the field should be declared 'volatile' as well. * main/cli.c: make kevin (and 64 bit machines) happy and remove a cast from char* to int in handling the return values from new-style handlers. On passing, note that main/loader.c::ast_load_resource() always return 0 so errors are not propagated up. I am not sure this is the intended behaviour. 2006-11-16 08:18 +0000 [r47718] Paul Cadach * main/channel.c, /, funcs/func_channel.c, include/asterisk/channel.h: Merged revisions 44809 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44809 | pcadach | 2006-10-10 23:44:54 +0700 (Втр, 10 Окт 2006) | 1 line CHANNEL() function sometime mix parameter and value ........ 2006-11-15 22:32 +0000 [r47713] Joshua Colp * channels/chan_local.c, /: Merged revisions 47712 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r47712 | file | 2006-11-15 17:31:17 -0500 (Wed, 15 Nov 2006) | 10 lines Merged revisions 47711 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47711 | file | 2006-11-15 17:29:30 -0500 (Wed, 15 Nov 2006) | 2 lines Make sure that the pvt structure exists before trying to do fixup on Local channels. (issue #7937 reported by mada123, fix by alamantia with mods by me) ........ ................ 2006-11-15 21:57 +0000 [r47710] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 47709 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47709 | tilghman | 2006-11-15 15:56:55 -0600 (Wed, 15 Nov 2006) | 2 lines Fix ODBC_STORAGE for when context is NULL ........ 2006-11-15 21:36 +0000 [r47708] Joshua Colp * main/channel.c, /: Merged revisions 47707 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47707 | file | 2006-11-15 16:33:41 -0500 (Wed, 15 Nov 2006) | 2 lines We need to ensure timelimit stuff is included as well so warnings get played. (issue #8050 reported by KNK) ........ 2006-11-15 21:21 +0000 [r47706] Olle Johansson * channels/chan_sip.c: Hunting the initreq change for an ACK 2006-11-15 20:59 +0000 [r47703-47704] TransNexus OSP Development * apps/app_osplookup.c: 1. Fix the bug that Asterisk hangs up the calls if the OSP AuthRsp messages without destination protocol infomation. 2. Fix the bug that Asterisk generats wrong dial string (no in IAX2/[username[:password]@]peer[:port][/exten[@context]][/options] format) for IAX. 3. Add support for oh323 channel driver. 4. Re-formate the code. * include/asterisk/astosp.h: 1. Re-format the code. 2006-11-15 20:51 +0000 [r47702] Kevin P. Fleming * /, main/file.c: Merged revisions 47701 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47701 | kpfleming | 2006-11-15 14:50:06 -0600 (Wed, 15 Nov 2006) | 2 lines don't try to call fclose() if fopen() failed ........ 2006-11-15 20:40 +0000 [r47700] Olle Johansson * /, channels/chan_sip.c: - Don't reply to ACK - Improve SIP history for debugging (Imported from 1.4) 2006-11-15 20:28 +0000 [r47685-47694] Kevin P. Fleming * /, apps/app_voicemail.c: Merged revisions 47693 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r47693 | kpfleming | 2006-11-15 14:27:38 -0600 (Wed, 15 Nov 2006) | 12 lines Merged revisions 47677 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47677 | kpfleming | 2006-11-15 11:56:42 -0600 (Wed, 15 Nov 2006) | 4 lines ensure that message duration is included in email notifications for forwarded messages (BE-96, fix by me after corydon used his clue-bat on me) ensure that duration in the message metadata is updated if prepending is done during forwarding (related to BE-96) remove prototype for API call that does not exist ........ ................ * /, main/config.c: Merged revisions 47690 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r47690 | kpfleming | 2006-11-15 14:01:22 -0600 (Wed, 15 Nov 2006) | 20 lines Merged revisions 47686,47688-47689 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47686 | kpfleming | 2006-11-15 13:42:05 -0600 (Wed, 15 Nov 2006) | 2 lines clear the category's variable tail pointer as well when variables are detached from it ........ r47688 | kpfleming | 2006-11-15 13:47:43 -0600 (Wed, 15 Nov 2006) | 2 lines when appending a list of variable to a category, ensure the tail pointer points to the last variable in the list ........ r47689 | kpfleming | 2006-11-15 13:58:46 -0600 (Wed, 15 Nov 2006) | 2 lines when re-writing the config file, don't repeat the path if it hasn't changed ........ ................ * /, main/config.c: Merged revisions 47684 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r47684 | kpfleming | 2006-11-15 12:43:30 -0600 (Wed, 15 Nov 2006) | 10 lines Merged revisions 47682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47682 | kpfleming | 2006-11-15 12:39:47 -0600 (Wed, 15 Nov 2006) | 2 lines ouch... don't use printf, use ast_log/ast_verbose ........ ................ 2006-11-15 17:40 +0000 [r47662-47669] Luigi Rizzo * channels/chan_oss.c: fix indentation * main/cli.c: small simplifications and localization of variables. * main/cli.c: new-style "core show channels" * main/cli.c: more changes to new style of "module load" and "load". Under FreeBSD, the filename_completion used in the above commands does not work. Not sure why, but on passing i note that the function is part of readline and is not reentrant, so it needs to be fixed one way or another. * main/cli.c: move another deprecated command to the new style * main/cli.c: move "core set debug" to the new style, and remove duplicated code for the deprecated handler. On passing fix a long standing bug in find_cli() which would not find the longest match - this part (trivial, basically just update matchlen on a match) must go in 1.4 and possibly 1.2 as well. 2006-11-15 16:09 +0000 [r47657-47661] Olle Johansson * /: Messed up earlier, cleaning up... * /, channels/chan_sip.c: Send proper SIP error message improperly when we can't allocate dialog (out of file handles is one cause) * channels/chan_sip.c: Update doxygen docs to reflect the code... 2006-11-15 15:02 +0000 [r47652-47654] Luigi Rizzo * include/asterisk/cli.h, main/cli.c: one more step cleaning the internal CLI interface: the NEW_CLI macro now supports extra arguments (to deprecate other commands). use this to implement unload and reload, and remove some unused functions. usual completion fixes (as these function accept multiple arguments). The summary is still a bit inconsistent. * include/asterisk/cli.h, main/cli.c: update the internal cli api following comments from kevin. This change basically simplifies the interface of the new-style handler removing almost all the tricks used in the previous implementation to achieve backward compatibility (which is still present and guaranteed.) 2006-11-15 04:47 +0000 [r47646] Joshua Colp * /, main/rtp.c: Merged revisions 47645 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47645 | file | 2006-11-14 23:45:24 -0500 (Tue, 14 Nov 2006) | 2 lines If NAT detection is turned on or already detected then say NAT is active when setting the remote RTP peer when doing early bridging. (issue #8365 reported by marcelbarbulescu) ........ 2006-11-15 00:19 +0000 [r47642] Kevin P. Fleming * /, main/term.c: Merged revisions 47641 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47641 | kpfleming | 2006-11-14 18:19:05 -0600 (Tue, 14 Nov 2006) | 2 lines more formatting cleanup, and avoid running off the end of the string ........ 2006-11-15 00:15 +0000 [r47640] Joshua Colp * /, main/rtp.c: Merged revisions 47639 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47639 | file | 2006-11-14 19:14:07 -0500 (Tue, 14 Nov 2006) | 2 lines Turn notice about unknown RTCP packet type into a debug message instead. ........ 2006-11-15 00:06 +0000 [r47636] Kevin P. Fleming * /, channels/misdn/isdn_lib.c: Merged revisions 47635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47635 | kpfleming | 2006-11-14 18:05:44 -0600 (Tue, 14 Nov 2006) | 2 lines silence compiler warning on 64-bit platforms (this variable is an 'int' anyway, comparing it to 'signed long' is not useful) ........ 2006-11-14 22:19 +0000 [r47633] Joshua Colp * /, apps/app_voicemail.c: Merged revisions 47632 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r47632 | file | 2006-11-14 17:17:16 -0500 (Tue, 14 Nov 2006) | 10 lines Merged revisions 47631 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47631 | file | 2006-11-14 17:15:10 -0500 (Tue, 14 Nov 2006) | 2 lines Update copyright information in the ADSI logo blob. ........ ................ 2006-11-14 22:08 +0000 [r47630] Luigi Rizzo * main/cli.c: add missing casts and remove an unused function. 2006-11-14 22:07 +0000 [r47623-47629] Joshua Colp * /, channels/chan_sip.c: Merged revisions 47628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47628 | file | 2006-11-14 17:05:03 -0500 (Tue, 14 Nov 2006) | 2 lines Only keep the video RTP structure around if 1. Video support is enabled and 2. A video codec is enabled on the dialog ........ * /, funcs/func_uri.c: Merged revisions 47625 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47625 | file | 2006-11-14 16:30:44 -0500 (Tue, 14 Nov 2006) | 2 lines Small documentation clarification for URIENCODE. (issue #8294 reported by salaud) ........ * apps/app_dial.c: Make local copy of arguments to parse. (issue #8362 reported by homesick) 2006-11-14 18:58 +0000 [r47622] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 47621 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47621 | tilghman | 2006-11-14 12:54:40 -0600 (Tue, 14 Nov 2006) | 3 lines Conversion of res_odbc API to include ast_ prefix did not completely transition app_voicemail when ODBC_STORAGE is used (reported on IRC by caio1982, not in bugtracker) ........ 2006-11-14 17:00 +0000 [r47619-47620] Luigi Rizzo * main/cli.c: fix completion for "module reload mod_1 mod_2 ... " (should do the same for the other similar commands, "reload", "module unload" and so on. * main/cli.c: partly convert to new style set-verbose, with completion fixes 2006-11-14 16:48 +0000 [r47618] Joshua Colp * /, apps/app_amd.c: Merged revisions 47617 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47617 | file | 2006-11-14 11:45:57 -0500 (Tue, 14 Nov 2006) | 2 lines Use LOG_DEBUG to print out the indication that app_amd is using default settings instead of using LOG_NOTICE. This stops needless logging of this information under normal circumstances. (issue #8361 reported by Seb7) ........ 2006-11-14 16:38 +0000 [r47614-47616] Luigi Rizzo * main/cli.c: replace two deprecated functions with calls to the standard ones, with fixes to argc/argv * main/cli.c: remove duplicated implementation for a deprecated function, use the original one instead with appropriate changes in argc/argv. This is not always applicable, but in some simple cases it is. 2006-11-14 16:15 +0000 [r47610-47611] Olle Johansson * include/asterisk/cli.h: need to check quoting in the doxygen docs... * include/asterisk/cli.h: Some improvements to doxygen docs 2006-11-14 16:09 +0000 [r47606-47609] Luigi Rizzo * main/cli.c: new-style for 'core show uptime', include 'complete' support and better error checking * main/cli.c: apply previous fix to old-style cli entries as well. * main/cli.c: fix "core set debug atleast ", and fix the simple case where a command can have multiple completions, the first ones coming from keywords in previous CLI entries. There is no solution yet for the complex case of N1 completions from a CLI entry, followed by N2 from the next one, and so on, because the _complete() handlers do not tell us how many matches it generates, so we don't know how many to skip when interrogating the other handlers. The only solution is to avoid, as much as possible, multiple CLI entries with the same prefix. * include/asterisk/cli.h, main/cli.c: Bring in the improved internal API for the CLI. WATCH OUT: this changes the binary interface (ABI) for modules, so e.g. users of g729 codecs need a rebuilt module (but read below). The new way to write CLI handlers is described in detail in cli.h, and there are a few converted handlers in cli.c, look for NEW_CLI. After converting a couple of commands i am convinced that it is reasonably convenient to use, and it makes it easier to fix the pending CLI issues. On passing, note a bug with the current 'complete' architecture: if a command is a prefix of multiple CLI entries, we miss some of the possible options. As an example, "core set debug" can continue with "channel" from one CLI entry, and "off" or "atleast" from another one. We address this problem in a separate commit (when i have figured out a fix, that is). ABI issues: I asked Kevin if it was ok to make this change and he said yes. While it would have been possible to make the change without breaking the module ABI, the code would have been more convoluted. I am happy to restore the old ABI (while still being able to use the "new style" handlers) if there is demand. 2006-11-14 13:17 +0000 [r47595-47600] Olle Johansson * channels/chan_sip.c: Adding some debug output to trace bug in realtime * channels/chan_sip.c: Adding a new debug line for issue #7351 - trying to find where an ACK can overwrite the initreq... * /, channels/chan_sip.c: Issue #8272 imported from 1.2/1.4 - Let the peerpoke system destroy it's own packets, please. * channels/chan_sip.c: Remove flags not used any more (thanks Luigi) 2006-11-13 22:40 +0000 [r47586-47587] Matt O'Gorman * codecs/codec_zap.c: oops no parens * main/frame.c, codecs/codec_zap.c: fix bytesize to 5.3kb for g723 codec and add support for multimode of tc400p 2006-11-13 21:32 +0000 [r47585] Joshua Colp * /, cdr/cdr_pgsql.c: Merged revisions 47584 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r47584 | file | 2006-11-13 16:28:57 -0500 (Mon, 13 Nov 2006) | 10 lines Merged revisions 47583 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47583 | file | 2006-11-13 16:26:36 -0500 (Mon, 13 Nov 2006) | 2 lines Initialize global pointers for connection and result to NULL. (issue #8356 reported by james) ........ ................ 2006-11-13 20:21 +0000 [r47582] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 47581 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r47581 | tilghman | 2006-11-13 14:20:01 -0600 (Mon, 13 Nov 2006) | 10 lines Merged revisions 47580 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47580 | tilghman | 2006-11-13 14:18:30 -0600 (Mon, 13 Nov 2006) | 2 lines Having more than 255 old messages caused corruption in the new/old count ........ ................ 2006-11-13 19:20 +0000 [r47579] Olle Johansson * /, channels/chan_sip.c: Small fix for uncommon scenario. 2006-11-13 19:19 +0000 [r47577-47578] Steve Murphy * /: Blocking 47576 from merging into trunk. Already done in 47577 * main/config.c: This solves bug 8342, whereby a crash occurs under certain circumstances while reading a config file with comments-- a call to CB_ADD shouldn't happen if withcomments is zero 2006-11-13 19:14 +0000 [r47575] Joshua Colp * channels/chan_h323.c: Make chan_h323 build again and make the CLI commands work. (reported on asterisk-dev mailing list by Di-Shi Sun) 2006-11-13 18:24 +0000 [r47568] Steve Murphy * /: blocked 47564 from 1.4 to be merged onto trunk; 47566 already did it 2006-11-13 18:23 +0000 [r47567] Joshua Colp * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Add 'loose' option to joinempty and leavewhenempty which is almost exactly like 'strict' except it does not count paused queue members as unavailable. (issue #8263 reported by gnarf) 2006-11-13 18:20 +0000 [r47566] Steve Murphy * pbx/ael/ael-test/ref.ael-test3: Eager people beat me to fixing the messed if, but we all forgot to update the regressions. Until now. 2006-11-13 17:55 +0000 [r47556-47560] Joshua Colp * apps/app_meetme.c: Don't play the "entering conference number " prompts if the 'q' option is used. If others believe this should be in 1.2/1.4 then we can put it in, but I'm uncomfortable doing so right now as it is a change of behavior. (issue #8138 reported by tmancill) * pbx/pbx_ael.c: Clean up last commit to better conform to standards. 2006-11-13 17:36 +0000 [r47554-47555] Steve Murphy * /: Blocking 47553 from 1.4 to trunk... 47554 is done for it. * pbx/pbx_ael.c: AEL need not complain about parkedcalls not being found... just confuses users 2006-11-13 17:10 +0000 [r47543-47552] Joshua Colp * /, apps/app_sms.c: Merged revisions 47551 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r47551 | file | 2006-11-13 12:08:07 -0500 (Mon, 13 Nov 2006) | 10 lines Merged revisions 47549 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47549 | file | 2006-11-13 12:05:32 -0500 (Mon, 13 Nov 2006) | 2 lines When sending an SMS with a user data header properly set the UDH flag in the first byte. (issue #8347 reported by hoffmeis) ........ ................ * main/cli.c: Return module show to a working state. (issue #8353 reported by jserve) 2006-11-13 16:08 +0000 [r47541] Olle Johansson * /, channels/chan_sip.c: Only produce error message once, don't fill the screen with them... (Testing SIPP thanks to JerJer and Greg) 2006-11-13 14:29 +0000 [r47536-47539] Luigi Rizzo * channels/chan_sip.c: merge from astobj2-r47450: use UNLINK to remove a packet from its queue, and related code rearrangement. Approved by: oej This could be made better if we declared struct sip_pvt *dialpg = pkt->owner; at the beginning of the function, and use it throughout the function. I'll let the boss decide :) * channels/chan_sip.c: merge from codename-pineapple and astobj2 47499: simplify __sip_ack() removing a strcmp for looking up packets. no functional change, only performance, so don't need to merging to earlier branches now. Approved By: oej * main/cli.c: stop looking for new entries when we know we are done. there is no functional change, so it is not necessary to bother merging this to 1.4 now. * main/cli.c: free memory when unregistering an entry. needs to be merged to 1.4 2006-11-13 05:58 +0000 [r47530] Tilghman Lesher * res/res_odbc.c, configs/res_odbc.conf.sample: Feature: allow the sanity SQL to be customized per connection class (Issue 6453) 2006-11-13 05:51 +0000 [r47529] Russell Bryant * /, configure, acinclude.m4: Merged revisions 47527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47527 | russell | 2006-11-13 00:48:18 -0500 (Mon, 13 Nov 2006) | 5 lines AC_PROG_SED is included in autoconf 2.60, but apparently it is not included in 2.59. So, to maintain compatability with 2.59 since it is a small change, copy this macro into acinclude.m4 and rename it to AST_PROG_SED. (issue #8345) ........ 2006-11-13 05:48 +0000 [r47524-47528] Tilghman Lesher * /, res/res_odbc.c: Merged revisions 47526 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r47526 | tilghman | 2006-11-12 23:46:18 -0600 (Sun, 12 Nov 2006) | 10 lines Merged revisions 47525 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47525 | tilghman | 2006-11-12 23:45:11 -0600 (Sun, 12 Nov 2006) | 2 lines If the execute fails a second time, make sure that we don't pass back a stale handle ........ ................ * channels/chan_zap.c, /: Merged revisions 47523 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r47523 | tilghman | 2006-11-12 19:12:01 -0600 (Sun, 12 Nov 2006) | 10 lines Merged revisions 47522 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47522 | tilghman | 2006-11-12 18:34:44 -0600 (Sun, 12 Nov 2006) | 2 lines Don't play dialtone if the seizing the channel fails (Bug 7754) ........ ................ 2006-11-12 20:47 +0000 [r47521] Olle Johansson * channels/chan_sip.c: Part of patch in #7403 to improve tag checking in pedantic mode (stephen_dredge) 2006-11-12 19:22 +0000 [r47520] Russell Bryant * channels/chan_iax2.c: The use of an ifdef to check for FreeBSD is useless here because the two versions of the format string are identical. Also, since each format is only used once, get rid of the use of defines all together. (issue #8344, julieng) In passing, also clean up the formatting a but to get rid of the nesting without the use of braces, as defined in the coding guidelines. 2006-11-12 16:15 +0000 [r47508-47514] Olle Johansson * /, channels/chan_sip.c: Restore auto-framing (DEA). Imported from 1.4 * /, channels/chan_sip.c: - Support UDPTL as well as udptl in T38 sdp. * /, channels/chan_sip.c: - Don't hangup because of failed re-invite. Go back to previous state. - Keep RTP running during T.38 session We might improve the code to issue ast_rtp_stop if T.38 re-invite not fails later on in the code, but I don't see many reasons to. * /, channels/chan_sip.c: - Add some comments to t.38 code - Remove improper blocking of ptime: in SDP 2006-11-12 06:31 +0000 [r47493-47498] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 47497 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r47497 | russell | 2006-11-12 01:23:23 -0500 (Sun, 12 Nov 2006) | 12 lines Merged revisions 47496 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47496 | russell | 2006-11-12 01:09:03 -0500 (Sun, 12 Nov 2006) | 4 lines Only do the check to determine whether the channel calling this function is an IAX2 channel when getting the IP address using the special argument, CURRENTCHANNEL. (issue #8341, jcovert) ........ ................ * Makefile, /: Merged revisions 47494 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47494 | russell | 2006-11-11 10:31:08 -0500 (Sat, 11 Nov 2006) | 6 lines Add the target "menuconfig" as an alias for the "menuselect" target. This is just a favor to users so that if you accidentally type "make menuconfig" instead of "make menuselect", it still works. (inspired by a comment on IRC from wangster calling me an "especially devious asterisk developer" for having it be menuselect instead of menuconfig. :) ) ........ * /, main/term.c: Merged revisions 47492 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47492 | russell | 2006-11-11 10:18:02 -0500 (Sat, 11 Nov 2006) | 2 lines Tweak the formatting of this new function to better conform to coding guidelines. ........ 2006-11-11 02:12 +0000 [r47491] Matt O'Gorman * main/logger.c, include/asterisk/term.h, main/term.c: safe terminal output is sweet. 2006-11-10 22:06 +0000 [r47478] Matthew Fredrickson * channels/chan_zap.c: Make sure we don't use 32bits for a value that only requires 1 bit. Also, fix a compiler warning for one of the SS7 functions. 2006-11-10 21:55 +0000 [r47467-47477] Olle Johansson * /, channels/chan_sip.c: Add some history and fix some debug output for autodestruct. * /, channels/chan_sip.c: Proper fix for adding debug... * /, channels/chan_sip.c: Make sure we destroy dialog in case of loop * /, channels/chan_sip.c: Cleanup imported from 1.4 2006-11-10 20:05 +0000 [r47459-47465] Joshua Colp * pbx/pbx_dundi.c: Fine, take this. * main/cli.c: A trunk that builds is a productive trunk. * pbx/pbx_dundi.c: Hello compiler working, goodbye compiler warning. (fix compiler warning introduced from pbx_dundi optimizations) * /, channels/chan_h323.c: Merged revisions 47457 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47457 | file | 2006-11-10 14:36:25 -0500 (Fri, 10 Nov 2006) | 2 lines Let's give this a go... ........ 2006-11-10 19:35 +0000 [r47456] Matthew Fredrickson * channels/chan_zap.c: Add fix for 7321. Ability to hide calleridname from zapata.conf 2006-11-10 19:01 +0000 [r47455] Olle Johansson * /, channels/chan_sip.c: Issue 8336- fix support for multipart SDP (imported from 1.2/1.4). (Alphaque) 2006-11-10 17:22 +0000 [r47445] Luigi Rizzo * build_tools/prep_moduledeps: manual merge from 1.4: grep -m not available on bsd, use head -1 which works for all 2006-11-10 17:01 +0000 [r47439] Tilghman Lesher * /, channels/chan_sip.c, channels/chan_skinny.c, channels/chan_h323.c, channels/chan_iax2.c, channels/chan_mgcp.c, main/cli.c: Merged revisions 47436 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47436 | tilghman | 2006-11-10 10:51:55 -0600 (Fri, 10 Nov 2006) | 2 lines Discussion of these CLI changes resulted in more consistency (Bug 8236) ........ 2006-11-10 16:55 +0000 [r47438] Joshua Colp * /, apps/app_chanspy.c: Merged revisions 47437 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47437 | file | 2006-11-10 11:53:16 -0500 (Fri, 10 Nov 2006) | 2 lines Only split up extension and context if a value exists. (issue #8332 reported by loloski) ........ 2006-11-10 16:38 +0000 [r47434-47435] Kevin P. Fleming * /, apps/app_queue.c: Merged revisions 47433 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47433 | kpfleming | 2006-11-10 10:36:49 -0600 (Fri, 10 Nov 2006) | 2 lines if adding a queue member is LOG_NOTICE, then removing them should be LOG_NOTICE, not LOG_DEBUG ........ * /, apps/app_queue.c: Merged revisions 47432 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47432 | kpfleming | 2006-11-10 10:34:04 -0600 (Fri, 10 Nov 2006) | 2 lines reflect addition/removal of dynamic queue members in queue_log, so that people using dialplan replacement for AgentCallbackLogin can still track login/logout (issue #7736, reported/patched by whoiswes but this commit was written by me and covers all three paths for AQM/RQM) ........ 2006-11-10 13:14 +0000 [r47415-47419] Olle Johansson * /, channels/chan_sip.c: Ripping out bad support for 491 replies to INVITE's. Let's try again properly later. * /, channels/chan_sip.c: Fix badly defined flag. Thanks fenlander! * channels/chan_sip.c: Small simplification and documentation correction. 2006-11-10 04:30 +0000 [r47408-47410] Russell Bryant * pbx/pbx_dundi.c: Various little bits of code cleanup to reduce nesting, remove useless casts, and to remove a duplicated error message after a memory allocation error * include/asterisk/app.h, apps/app_read.c, main/app.c: Add the ability to specify multiple prompts to the Read() dialplan application, similar to Background() and Playback(). (issue #7897, jsmith, with some modifications) 2006-11-10 03:45 +0000 [r47399-47406] Joshua Colp * /, channels/chan_h323.c: Merged revisions 47405 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47405 | file | 2006-11-09 22:44:36 -0500 (Thu, 09 Nov 2006) | 2 lines Fix building of chan_h323 by completeing some structure definitions. (issue #8327 reported by Mithraen) ........ * main/pbx.c: This should already be called while locked. * /, apps/app_voicemail.c: Merged revisions 47398 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47398 | file | 2006-11-09 17:32:30 -0500 (Thu, 09 Nov 2006) | 2 lines Do conversion in a more easier to read and working way for \r, \n, and \t. (issue #8324 reported by johnlange) ........ 2006-11-09 21:32 +0000 [r47392] Russell Bryant * channels/chan_zap.c, /, build_tools/prep_moduledeps, apps/app_voicemail.c: Merged revisions 47391 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47391 | russell | 2006-11-09 16:26:27 -0500 (Thu, 09 Nov 2006) | 7 lines Work around an issue that caused menuselect to display a bogus description for app_voicemail and chan_zap. These modules use some preprocessor directives to determine what it will report to Asterisk as its description. However, the way we extract this information from the source files for menuselect is not smart enough to figure this out. (issue #8326, #8328) ........ 2006-11-09 17:08 +0000 [r47382] Joshua Colp * channels/chan_phone.c, /: Merged revisions 47380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r47380 | file | 2006-11-09 11:53:25 -0500 (Thu, 09 Nov 2006) | 10 lines Merged revisions 47379 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47379 | file | 2006-11-09 11:48:05 -0500 (Thu, 09 Nov 2006) | 2 lines Don't include compiler.h on kernels 2.6.18 and higher as, well, it's apparently going to be removed. This should make all you FC6 fans happy as your Asterisk will now build without any mods. ........ ................ 2006-11-09 16:30 +0000 [r47353-47378] Russell Bryant * /, main/cli.c: Merged revisions 47377 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47377 | russell | 2006-11-09 11:28:15 -0500 (Thu, 09 Nov 2006) | 2 lines fix tab completion for "core debug channel" and "core no debug channel" ........ * /, main/cli.c: Merged revisions 47375 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47375 | russell | 2006-11-09 11:24:02 -0500 (Thu, 09 Nov 2006) | 3 lines Fix "core show channel". Also, fix tab completion for both "core show channel" and "core show channels". ........ * /, main/cli.c: Merged revisions 47372 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47372 | russell | 2006-11-09 11:18:33 -0500 (Thu, 09 Nov 2006) | 3 lines Fix "core debug channel ". I guess someone needs to go through and audit every CLI command that changed number of arguments ... ........ * /, main/cli.c: Merged revisions 47366 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47366 | russell | 2006-11-09 10:49:39 -0500 (Thu, 09 Nov 2006) | 3 lines Fix another CLI command, "core show uptime" ... (issue #8323, reported by johnlange, fixed by myself) ........ * /, main/asterisk.c: Merged revisions 47352 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47352 | russell | 2006-11-09 01:31:37 -0500 (Thu, 09 Nov 2006) | 3 lines fix "core show version" to reflect the new number of arguments for this CLI command (issue #8316, kshumard) ........ 2006-11-09 00:46 +0000 [r47343-47351] Steve Murphy * /: Blocking 47344 from automerging into trunk * /: Blocking 47348 from automerging into trunk * main/channel.c: This mod via bug 7531 * channels/chan_skinny.c: committed in behalf of bug 8190 2006-11-08 22:35 +0000 [r47341] Olle Johansson * channels/chan_sip.c: - Add Max-Forwards higher in the packet, following recommendations - Fix documentation for sip_pvt_lock/unlock - doxygen does not inherit like zapata.conf !!! - Change doc for a sip_pvt setting 2006-11-08 21:59 +0000 [r47337-47339] Kevin P. Fleming * main/frame.c: restore display of G.722 codec * /, channels/chan_sip.c: Merged revisions 47333 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47333 | kpfleming | 2006-11-08 12:07:16 -0600 (Wed, 08 Nov 2006) | 2 lines add simple fix for SDP to report proper sample rate for G.722 media sessions ........ 2006-11-08 18:26 +0000 [r47335] Joshua Colp * main/pbx.c, CHANGES: Display CID matching information when using dialplan show. (issue #8279 reported by caio1982) 2006-11-08 17:06 +0000 [r47325-47332] Russell Bryant * /, utils/streamplayer.c: Merged revisions 47331 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47331 | russell | 2006-11-08 12:03:09 -0500 (Wed, 08 Nov 2006) | 5 lines I occasionally get email from users that are trying to figure out what this does, or due to some misunderstanding as to what it is supposed to do, can't get it to work. So, I have added some text here to hopefully explain what this application does and does not do. ........ * /, configure, configure.ac, acinclude.m4: Merged revisions 47327 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47327 | russell | 2006-11-08 11:31:59 -0500 (Wed, 08 Nov 2006) | 4 lines Copy the macros from libtool.m4 to our own acinclude.m4 such that libtool is no longer required to be installed to be able to generate the configure script. ........ 2006-11-08 15:28 +0000 [r47321] Kevin P. Fleming * channels/chan_sip.c: coding guidelines, coding guidelines, coding guidelines 2006-11-08 13:59 +0000 [r47314-47318] Luigi Rizzo * channels/chan_sip.c: merge from team/rizzo/astobj2 rev.47271 avoid doing p > 0 when p is a pointer; move a lock closer to the place where it is needed Approved By: oej * channels/chan_sip.c: merge from team/rizzo/astobj2 rev.47246 Same as for peers and users, replace ASTOBJ_UNREF(r, sip_registry_destroy) with unref_registry(r); Approved By: oej * channels/chan_sip.c: merge from team/rizzo/astobj2, rev 47243, 47244, 47245: Replace ASTOBJ_UNREF(peer, sip_destroy_peer) with unref_peer(peer); This places the name of the destructor in one place only (where it should be), eliminates the chance of errors in case you specify the wrong destructor, and also lets the compiler do type checking on the argument, again helping with keeping the code clean. Same for users. remove two duplicate definitions. Approved By: oej * channels/chan_sip.c: merge rev.47224 from team/rizzo/astobj2: hide dialoglist lock/unlocking in wrapper functions. Approved By: oej * channels/chan_sip.c: silence compiler about uninitialized variables. The compiler is wrong, but it has the last word. 2006-11-08 08:01 +0000 [r47313] Olle Johansson * /, channels/chan_sip.c: Destroy dialog properly at unload (rizzo) 2006-11-08 07:21 +0000 [r47306] Luigi Rizzo * channels/chan_jingle.c, channels/chan_gtalk.c: fix compilation. Overall i think the previous change to ast_channel_alloc() to close bug 7506 should have been done by defining an ast_set_callerid_noevent() function that does the setting without generating the event. Lot less code duplication, and easier to handle. 2006-11-08 03:13 +0000 [r47304-47305] Russell Bryant * configure.ac: add a comment about where AC_PROG_LD comes from * aclocal.m4 (removed), /: remove aclocal.m4 from the tree since it is just an intermediate file created while generating the configure script. 2006-11-07 23:14 +0000 [r47295-47300] Luigi Rizzo * main/asterisk.c: fix "core show profile" parsing. Needs to go in 1.4 too, but ENOTIME now * apps/app_queue.c: %ld and time_t don't match, so cast the argument to long to ease portability problems 2006-11-07 21:47 +0000 [r47290] Steve Murphy * main/pbx.c, channels/chan_local.c, channels/chan_vpb.cc, channels/chan_zap.c, channels/chan_sip.c, res/res_features.c, channels/chan_agent.c, main/utils.c, include/asterisk/channel.h, channels/chan_gtalk.c, channels/chan_iax2.c, channels/chan_oss.c, main/channel.c, channels/chan_jingle.c, channels/chan_phone.c, channels/chan_misdn.c, channels/chan_skinny.c, channels/chan_features.c, channels/chan_h323.c, channels/chan_alsa.c, channels/chan_nbs.c, include/asterisk/stringfields.h, channels/chan_mgcp.c, apps/app_voicemail.c: A fair number of changes for the sake of bug 7506 2006-11-07 20:16 +0000 [r47285-47288] Joshua Colp * channels/chan_local.c, /: Merged revisions 47287 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47287 | file | 2006-11-07 15:14:58 -0500 (Tue, 07 Nov 2006) | 2 lines This is not the commit you are looking for... ........ * channels/chan_local.c, /: Merged revisions 47284 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47284 | file | 2006-11-07 15:08:52 -0500 (Tue, 07 Nov 2006) | 2 lines Make MOH work as it did before in chan_local, without this then it can go funky when transfers and MOH are involved. (issue #7671 reported by jmls) ........ 2006-11-07 18:56 +0000 [r47280] Kevin P. Fleming * /, configs/musiconhold.conf.sample: Merged revisions 47279 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47279 | kpfleming | 2006-11-07 12:56:21 -0600 (Tue, 07 Nov 2006) | 2 lines clean up sample config, and make native file playback the more obvious default choice ........ 2006-11-07 18:50 +0000 [r47278] Matt O'Gorman * apps/app_voicemail.c: rge overhaul to voicemail imap support. Allows support for more imap servers, also a better implementation of several parts of the original work. patch provided by 8033 with major upgrades. minor differences from 1.4 patch do to changes in app_voicemail 2006-11-07 17:33 +0000 [r47269] Olle Johansson * /, channels/chan_sip.c: Break -> continue to make stuff work... Thanks, Luigi! 2006-11-07 14:25 +0000 [r47257-47259] Kevin P. Fleming * /: remove another broken property merge * /: remove properties that shouldn't be merged to this branch * /: use editable URL for menuselect, and switch to trunk 2006-11-07 13:26 +0000 [r47251-47252] Olle Johansson * /, channels/chan_sip.c: issue #8265 - don't reply to ACK. Imported from 1.2, 1.4 * include/asterisk/frame.h: Stealing Tilghman's explanation from the -dev list and producing documentation... 2006-11-07 08:34 +0000 [r47242] Luigi Rizzo * main/utils.c: explain why ast_carefulwrite is written the way it is, and also that it doesn't really work as claimed. 2006-11-07 01:28 +0000 [r47232-47240] Russell Bryant * /, res/res_musiconhold.c: Merged revisions 47239 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r47239 | russell | 2006-11-06 20:25:10 -0500 (Mon, 06 Nov 2006) | 13 lines Merged revisions 47238 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47238 | russell | 2006-11-06 20:22:58 -0500 (Mon, 06 Nov 2006) | 5 lines If random order is enabled for files mode music on hold, set a random initial position, instead of always starting at the first file, and doing the random operation only when switching to the next file. (bug reported by John Lange on the asterisk-dev mailing list) ........ ................ * utils/check_expr.c: check for failure after call to calloc() (issue #8295) 2006-11-06 17:27 +0000 [r47230] Kevin P. Fleming * UPGRADE.txt: minor change to test live syncing 2006-11-06 17:05 +0000 [r47229] Joshua Colp * main/manager.c, utils/astman.c, include/asterisk/manager.h: Add support for manager hooks, so you could fire off manager events over IRC if you were crazy enough. (issue #5161 reported by anthm with mods by moi) 2006-11-05 01:04 +0000 [r47210-47213] Russell Bryant * pbx/pbx_dundi.c: Make pbx_dundi compile again. Sorry. :( * configs/zapata.conf.sample: List ss7 with the rest of the valid signalling types. Group SS7 options together and comment them out by default. 2006-11-04 22:16 +0000 [r47209] Olle Johansson * channels/chan_sip.c: Don't lock dialoglist if monitor runs __sip_destroy. Hmmm. I did not change pbx_dundi and yet it doesn't compile ;-) 2006-11-04 22:08 +0000 [r47206-47207] Russell Bryant * pbx/pbx_dundi.c: use the AST_MODULE_LOAD_* return codes from load_module() * pbx/pbx_dundi.c: simplify a couple of loops 2006-11-04 21:48 +0000 [r47205] Olle Johansson * channels/chan_sip.c: Move IP address selection for media out of add_sdp 2006-11-04 21:44 +0000 [r47204] Russell Bryant * pbx/pbx_dundi.c: Do some minor cleanup to the section of code that sets the EID by getting the mac address for an ethernet interface 2006-11-04 21:17 +0000 [r47200-47203] Olle Johansson * channels/chan_sip.c: Make srvlookup global_srvlookup to follow the rest of the code * channels/chan_sip.c: Simplify previous patch * channels/chan_sip.c, configs/sip.conf.sample: Adding new config option "limitpeersonly" to only apply call limits to the peer side of a type=friend. This is for trying to support BJ in his quest to solve some issues with the queue system and type=friend objects. BJ: Please test! * /, channels/chan_sip.c: Importing patch for Invite/replaces from 1.4 2006-11-04 18:12 +0000 [r47197-47198] Russell Bryant * /, main/cli.c: Merged revisions 47196 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47196 | russell | 2006-11-04 13:10:22 -0500 (Sat, 04 Nov 2006) | 2 lines Fix another bug in "core set debug" ... ........ * /, main/asterisk.c, main/cli.c: Merged revisions 47195 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47195 | russell | 2006-11-04 12:59:39 -0500 (Sat, 04 Nov 2006) | 2 lines Really fix the "core set debug" and "core set verbose" CLI commands. ........ 2006-11-04 17:45 +0000 [r47194] Olle Johansson * channels/chan_sip.c: Reverting rev 47093 until we have an agreement on how to implement this, if at all. 2006-11-04 17:40 +0000 [r47193] Russell Bryant * /, main/cli.c: Merged revisions 47192 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47192 | russell | 2006-11-04 12:38:24 -0500 (Sat, 04 Nov 2006) | 3 lines fix the "atleast" option to the "core set verbose" and "core set debug" CLI commands ........ 2006-11-04 11:00 +0000 [r47179-47189] Luigi Rizzo * apps/app_dial.c: move out another large block to a large function, and document some possibly missing parts in the privacy screening code. Now that it is more streamlined it is easier to see differences in handling the various cases. Have not tested the code in depth. * res/res_agi.c: useless cast removal... * main/logger.c: remove many unnecessary casts * main/app.c: remove a useless cast * configs/manager.conf.sample: document the "debug" parameter, and the change manager list -> manager show * apps/app_dial.c: fix indentation of a block, and do minor simplifications at the end of another one. * apps/app_dial.c: complete previous commit. * apps/app_dial.c: move another block into a function. On passing, avoid two null-pointer string dereference while printing messages (which are sometimes not fatal in some platforms, but still wrong). These two lines at least should be merged to 1.4 once i am done with all the changes here. * apps/app_dial.c: move a large block into a separate function. Mark with XXX a possible bug in previous code which used the wrong source in case of a forwarded call. the function do_forward() needs to be split further, as the initial part is replicated in another places (with some minor differences, most likely forgotten when updating after the copy). 2006-11-03 23:27 +0000 [r47178] Steve Murphy * channels/chan_sip.c: This fix introduced via bug 8233 2006-11-03 23:24 +0000 [r47160-47177] Luigi Rizzo * apps/app_dial.c: another small set of simplifications * apps/app_dial.c: change HANDLE_CAUSE into a function. * apps/app_dial.c: remove redundant checks * apps/app_dial.c: start integrating the simplifications proposed in bug 0005860, as usual a bit at a time to ease locating new bugs or fixes worth merging into other branches. In this commit, introduce a macro, S_REPLACE, that replaces a string possibly freeing the previous value. In one of these places (see the comment marked XXX) the previous code might leak memory - if so, this ought to be merged in 1.4 The macro might be worth putting in one of the global headers (e.g. include/asterisk/strings.h) as the construct is used in a million places in the asterisk code. 2006-11-03 19:15 +0000 [r47146] Joshua Colp * apps/app_voicemail.c: One has to create the path and filename in order to copy a file there. (issue #8278 reported by davebath) 2006-11-03 18:53 +0000 [r47072-47132] Luigi Rizzo * main/manager.c, include/asterisk/manager.h: add a new cli/manager.conf option "debug" to enable/disable debugging code in the manager. At the moment the debugging code is very lightweight, if the option is enabled manager messages also carry a sequence number and the info where they have been generated e.g. SequenceNumber: 10 File: chan_sip.c Line: 11927 Func: handle_response_register It is not worthwhile having this as a compile time option right now, because the extra work involved at runtime is just checking one variable. * channels/chan_zap.c: remove old/useless usecnt stuff * channels/chan_vpb.cc: remove old/useless usecnt stuff. I think this module doesn't compile, anyways, because it has not been updated to the new module interface. * main/cli.c: Fix "core show channels" and "core show modules". Not sure it applies like this to 1.4 because of deprecate versions of the same command(s). * res/res_jabber.c: move variable declarations to the beginning of a block. * /: block other changes of mine already applied to trunk. * /: block more changes of mine already applied to trunk * /: blocking 47107 * /: blocking 47108 * channels/chan_sip.c: Save the 'From' header received in a REGISTER message so we can show it e.g. in the Manager interface. This information is available as a callerid (or something like that) during a call, but not when a device is registered but silent. It may be useful to have it available e.g. when developing a user interface/operator panel, to map numbers to names. experimental, so not committed to 1.4 * channels/chan_jingle.c, channels/chan_gtalk.c: remove useless usecnt stuff * channels/chan_phone.c: remove useless usecnt stuff * channels/chan_alsa.c: remove useless usecnt stuff * channels/chan_agent.c: remove useless usecnt stuff * channels/chan_features.c: remove useless usecnt handling * channels/chan_skinny.c: remove useless usecnt handling code 2006-11-02 23:55 +0000 [r47052-47054] Tilghman Lesher * main/udptl.c, /, channels/chan_skinny.c, res/res_agi.c, channels/chan_h323.c, res/res_jabber.c, main/rtp.c: Merged revisions 47053 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47053 | tilghman | 2006-11-02 17:49:13 -0600 (Thu, 02 Nov 2006) | 2 lines More changes making the CLI more consistent with "category verb arguments" (continuation of issue 8236) ........ * main/pbx.c, channels/chan_local.c, main/frame.c, channels/chan_sip.c, /, res/res_features.c, res/res_crypto.c, channels/chan_agent.c, res/res_musiconhold.c, apps/app_queue.c, channels/chan_iax2.c, main/config.c, main/cli.c, main/channel.c, main/manager.c, channels/chan_skinny.c, res/res_agi.c, channels/chan_features.c, main/logger.c, main/file.c, main/http.c, res/res_indications.c, main/image.c, res/res_odbc.c, main/asterisk.c, channels/chan_mgcp.c, apps/app_voicemail.c: Merged revisions 47051 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47051 | tilghman | 2006-11-02 17:00:20 -0600 (Thu, 02 Nov 2006) | 2 lines Reverse change of "show" to "list" and make several other commands more consistent with "category verb arguments" ........ 2006-11-02 21:40 +0000 [r47037] Joshua Colp * main/pbx.c, include/asterisk/pbx.h: Let's make application/function/hint lists read/write lists... just for kicks 2006-11-02 21:34 +0000 [r47035] Matthew Fredrickson * channels/chan_zap.c: Updates to do unblock correctly 2006-11-02 20:24 +0000 [r46999-47021] Olle Johansson * /, channels/chan_sip.c: Move check for codec translators to an earlier place in the call, so we can fail gracefully (imported from 1.4) * /, channels/chan_sip.c: Disable code for not implemented functionality (T38 over RTP/TCP) 2006-11-02 18:34 +0000 [r46991-46994] Russell Bryant * include/asterisk/astobj.h: Sure enough, some of the uses of astobj are doing recursive locking. This doesn't work with rwlocks, so, this is reverted for now. * include/asterisk/astobj.h: astobj was already set up to use read and write locks. Now that we have wrappers for them, use them here. 2006-11-02 18:01 +0000 [r46967-46972] Joshua Colp * main/translate.c: Convert translation core linked list over to read/write based one, since it spends most of it's time only reading. * include/asterisk/linkedlists.h: Add AST_RWLIST_* set of macros which implement linked lists using read/write locks, the actual list manipulation is still done via the old macros. 2006-11-02 17:51 +0000 [r46966] Russell Bryant * /, res/res_musiconhold.c: Merged revisions 46965 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r46965 | russell | 2006-11-02 12:49:54 -0500 (Thu, 02 Nov 2006) | 11 lines Merged revisions 46964 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r46964 | russell | 2006-11-02 12:47:56 -0500 (Thu, 02 Nov 2006) | 3 lines ignore files in a music on hold directory that begin with '.' (issue #8249, cboie) ........ ................ 2006-11-02 16:51 +0000 [r46940] Joshua Colp * include/asterisk/lock.h: Set the AST_RWLOCK_INIT_VALUE to the PTHREAD_RWLOCK_INIT_VALUE if it is available, that way outside stuff can determine whether to use a constructor or deconstructor for initialization instead of using the init value. 2006-11-02 16:50 +0000 [r46939] Matthew Fredrickson * channels/chan_zap.c: Changes to show blocked/unblocked states, as well as in service, out of service state 2006-11-02 16:45 +0000 [r46938] Kevin P. Fleming * /, channels/chan_sip.c: Merged revisions 46937 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46937 | kpfleming | 2006-11-02 10:45:32 -0600 (Thu, 02 Nov 2006) | 2 lines don't send INVITE when we have determined that we can't offer any audio formats due to lack of trancoding support (or incorrect configuration) ........ 2006-11-02 16:28 +0000 [r46931-46935] Joshua Colp * configure, include/asterisk/autoconfig.h.in, configure.ac, include/asterisk/lock.h: I'm crazy so I will add this... pthread rwlock wrappers, along with autoconf stuff that detects the presence of the initializer and the ability to set the kind of lock (in our case we rather like writer preferred locks so writer starvation doesn't occur... but on something like Darwin we don't get that) * /, channels/chan_sip.c: Merged revisions 46930 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r46930 | file | 2006-11-02 11:06:39 -0500 (Thu, 02 Nov 2006) | 10 lines Merged revisions 46920 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r46920 | file | 2006-11-02 11:02:27 -0500 (Thu, 02 Nov 2006) | 2 lines Repeat after me oej: I will at least make sure my code compiles before I commit it. ........ ................ 2006-11-02 16:03 +0000 [r46926] Matthew Fredrickson * channels/chan_zap.c: Add simple down event support 2006-11-02 15:47 +0000 [r46906] Nadi Sarrar * channels/misdn/isdn_lib.c, channels/misdn_config.c: find_free_chan_in_stack: cleanup buggy usage 2006-11-02 15:31 +0000 [r46902] Olle Johansson * /, channels/chan_sip.c: Don't overwrite pkt->flags (imported from 1.2/1.4) 2006-11-02 14:15 +0000 [r46846-46886] Russell Bryant * main/callerid.c: various whitespace changes to reduce indentation and to better conform to formatting guidelines * main/callerid.c: Change the buffer used in callerid_feed() and callerid_feed_jp() to be allocated on the stack using alloca() instead of using malloc() since they are only used locally to these functions. * /, main/say.c: Merged revisions 46857 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46857 | russell | 2006-11-01 18:01:48 -0500 (Wed, 01 Nov 2006) | 2 lines fix saying one hundred and two hundred in hebrew (issue #7810, eldadran) ........ * CHANGES: Add a couple of things to the CHANGES file * Makefile, /, configure, codecs/gsm/Makefile, configure.ac, build_tools/strip_nonapi, makeopts.in: Merged revisions 46847 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46847 | russell | 2006-11-01 17:51:21 -0500 (Wed, 01 Nov 2006) | 3 lines Fixes for cross-compilation on mips (issue #8058, ywalther, with some modifications) ........ * aclocal.m4, /, build_tools/menuselect-deps.in, configure, build_tools/embed_modules.xml, configure.ac: Merged revisions 46845 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46845 | russell | 2006-11-01 17:32:12 -0500 (Wed, 01 Nov 2006) | 5 lines Add a check in the configure script to determine whether ld is GNU ld or not. This is needed because module embedding only works for gnu ld. GNU ld is now listed as a dependency for all of the module embedding options in menuselect. (issue #8143) ........ 2006-11-01 20:38 +0000 [r46823] Matt O'Gorman * /, channels/chan_gtalk.c: Merged revisions 46822 via svnmerge from https://svn.digium.com/svn/asterisk/branches/1.4 ........ r46822 | mogorman | 2006-11-01 14:35:41 -0600 (Wed, 01 Nov 2006) | 2 lines bind address support from bug 8164 ........ 2006-11-01 19:48 +0000 [r46801] Steve Murphy * res/res_config_odbc.c: a fix for bug 8251; the var_val needs to accept longer strings or mass confusion and a lot of lost time is the result 2006-11-01 18:41 +0000 [r46782] Joshua Colp * /, main/Makefile: Merged revisions 46780 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46780 | file | 2006-11-01 13:39:47 -0500 (Wed, 01 Nov 2006) | 2 lines Force poll() emulation for Darwin to always be on. It's too broken to consider being used. This resolves the console issue OSX users have been seeing. I would have liked to autoconf this but I haven't been able to come up with a test case that works. Que sera. ........ 2006-11-01 18:40 +0000 [r46779-46781] Russell Bryant * doc/channelvariables.txt, pbx/pbx_dundi.c: Add the ability to pass options to the Dial application when using the DUNDi switch in the dialplan by setting the DUNDIDIALARGS channel variable. (issue #8084, patch by bluecrow76, with small modifications and documentation updates) * /, res/res_monitor.c: Merged revisions 46778 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r46778 | russell | 2006-11-01 13:26:35 -0500 (Wed, 01 Nov 2006) | 17 lines Merged revisions 46776 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r46776 | russell | 2006-11-01 13:24:17 -0500 (Wed, 01 Nov 2006) | 9 lines soxmix and Asterisk expect different file extensions for certain formats. This was already handled for the wav49 format. However, it was not handled for ulaw and alaw. I fixed this in such a way that using the alternate extensions for ulaw and alaw will only happen if we know we're calling soxmix, and not a custom script defined using the MONITOR_EXEC variable. The wav49 processing was left alone so that external scripts will see no behavior change. (issue #7550, reported by mnicholson, proposed patch by junky, committed fix is a bit different) ........ ................ 2006-11-01 18:26 +0000 [r46777] Joshua Colp * /, channels/chan_iax2.c: Merged revisions 46775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46775 | file | 2006-11-01 13:21:34 -0500 (Wed, 01 Nov 2006) | 2 lines It's another round of chan_iax2 fixes! Should hopefully fix the deadlock issues people have been reporting. IAXtel now has qualify turned on for 800 peers and it is handling it fine. ........ 2006-11-01 18:16 +0000 [r46759-46774] Steve Murphy * CHANGES: OOps. forgot to add this to CHANGES * main/say.c, apps/app_voicemail.c: This introduces Brazilian Portuguese via 7663 * main/config.c: Cleanups suggested by Russell. 2006-11-01 17:09 +0000 [r46758] Luigi Rizzo * res/res_features.c: move variable declaration in the middle of a block 2006-11-01 16:51 +0000 [r46745] Russell Bryant * channels/chan_zap.c, /: Merged revisions 46744 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46744 | russell | 2006-11-01 11:39:09 -0500 (Wed, 01 Nov 2006) | 2 lines Prevent an infinite loop when config processing gets to a jitterbuffer option ........ 2006-11-01 00:07 +0000 [r46732] Matt O'Gorman * res/res_features.c: change default return extension after parking timeout. 6953 with minor changes. 2006-10-31 22:19 +0000 [r46719] Kevin P. Fleming * /, main/translate.c, include/asterisk/translate.h: Merged revisions 46714 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46714 | kpfleming | 2006-10-31 15:47:48 -0600 (Tue, 31 Oct 2006) | 2 lines add an API so that translators can activate/deactivate themselves when needed ........ 2006-10-31 22:07 +0000 [r46717-46718] Jason Parker * main/translate.c: Fix "core show translation" output. Issue #8243, patch by Damin. 2006-10-31 18:10 +0000 [r46683-46696] Luigi Rizzo * channels/chan_iax2.c: remove old/useless usecount handling * channels/chan_sip.c: remove old/useless usecount stuff. * channels/chan_oss.c: remove old/useless usecount management code. 2006-10-31 15:22 +0000 [r46661] Russell Bryant * main/manager.c: Fix the new send text manager command. There is no way this could have worked. - Check the channel name string length to be zero, not non-zero - Check the message string length to be zero, not non-zero - unlock the channel *after* calling sendtext 2006-10-31 13:56 +0000 [r46582-46650] Olle Johansson * channels/chan_sip.c: Set #define for TIMER T1 value * channels/chan_sip.c: Cleaning up code * funcs/func_enum.c, /, include/asterisk/enum.h, main/enum.c: Issue #80898 - Restoring func_enum (otmar) * main/manager.c: Add manager sendtext action. (Issue 6131, ZX81 - thanks!) * /, channels/chan_sip.c, configs/sip.conf.sample: Fix rport handling. ...where did the 1.2 properties come from, really? they're back. * /, channels/chan_sip.c: - If peer that register fails ACL, fail him - Remove the 1.2 props I've set by mistake earlier * /: Block patch that only applies to 1.4 * main/loader.c: Take two, using find_resource on Kevin's suggestion. Might need better locking support, giving up if we can't get the lock. Right now, using existing locking in find_resource 2006-10-31 06:37 +0000 [r46556-46565] Russell Bryant * apps/app_cdr.c: add author doxygen tag (issue #8241, kshumard) * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 46563 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46563 | russell | 2006-10-31 01:30:53 -0500 (Tue, 31 Oct 2006) | 3 lines Start Asterisk later in the boot process to ensure it starts after stuff like MySQL (issue #8253, Alric) ........ * /, main/utils.c: Merged revisions 46561 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r46561 | russell | 2006-10-31 01:19:56 -0500 (Tue, 31 Oct 2006) | 11 lines Merged revisions 46560 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r46560 | russell | 2006-10-31 01:18:36 -0500 (Tue, 31 Oct 2006) | 3 lines When handling the case where the hostname is just an IPV4 numeric address, be sure to set the address type. (issue #8247, alexr) ........ ................ * /, res/res_agi.c: Merged revisions 46558 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r46558 | russell | 2006-10-31 01:14:13 -0500 (Tue, 31 Oct 2006) | 11 lines Merged revisions 46557 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r46557 | russell | 2006-10-31 01:13:09 -0500 (Tue, 31 Oct 2006) | 3 lines fix some copy/paste bugs in the checking of arguments for the "control stream file" AGI command (issue #8255, mnicholson) ........ ................ * /, main/translate.c: Merged revisions 46554 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46554 | russell | 2006-10-31 00:55:07 -0500 (Tue, 31 Oct 2006) | 5 lines Add a small tweak to the code that checks to see whether destination formats are translatable based on the source format. If we have already determined that there is no translation path in one direction, don't bother checking the other direction. ........ 2006-10-30 23:11 +0000 [r46541] Steve Murphy * apps/app_dial.c, utils/astman.c: These changes submitted by moy via bug 6992, to add a Dial 'End' event to asterisk. I include some changes to astman to cover other events that have been added. 2006-10-30 22:27 +0000 [r46529] Kevin P. Fleming * /, main/translate.c: Merged revisions 46526 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46526 | kpfleming | 2006-10-30 16:19:55 -0600 (Mon, 30 Oct 2006) | 3 lines when unregistering a translator, don't rebuild the translation matrix unless needed when filtering formats out of an offer, ensure we check for translation ability in both directions ........ 2006-10-30 21:56 +0000 [r46513-46514] Olle Johansson * funcs/func_module.c: show, list, view, display... whatever. * funcs/func_module.c (added), include/asterisk/module.h, main/loader.c: Adding dialplan function IFMODULE, so you can create dialplans that handle various PBX installations and checks if a module is loaded before using it. example IFMODULE(chan_sip3.so) issue #6671 in the bug tracker, finally gone. Thanks to mithraen for keeping it updated. 2006-10-30 21:46 +0000 [r46512] Kevin P. Fleming * /, include/asterisk/linkedlists.h: Merged revisions 46511 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46511 | kpfleming | 2006-10-30 15:46:07 -0600 (Mon, 30 Oct 2006) | 2 lines ensure that items removed from a list are always unlinked from the list (next pointer set to NULL) ........ 2006-10-30 21:22 +0000 [r46508-46509] Olle Johansson * channels/chan_sip.c: Update sip list to eventlist format. * main/pbx.c, main/manager.c, include/asterisk/manager.h: Issue #3930 - Add manager command for listing dialplan (coded april 2005, in bugtracker since) 2006-10-30 21:11 +0000 [r46507] Joshua Colp * /, configure, configure.ac: Merged revisions 46506 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46506 | file | 2006-10-30 16:09:13 -0500 (Mon, 30 Oct 2006) | 2 lines Don't explicitly link in crypt as it is not used on some platforms. ........ 2006-10-30 19:56 +0000 [r46476-46489] Olle Johansson * channels/chan_sip.c, configs/sip.conf.sample: Change name of "contact" setting to "callback" which better reflects what it is to the person that configures asterisk. That we use it internally in the contact header is a totally different story. Still not convinced this is a good option. * channels/chan_sip.c: Globals need the "global_" prefix in chan_sip, and need to be reset to default value at reload. 2006-10-30 18:17 +0000 [r46475] Joshua Colp * /, channels/chan_iax2.c: Merged revisions 46474 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46474 | file | 2006-10-30 13:13:07 -0500 (Mon, 30 Oct 2006) | 2 lines We need to lock the pvt structure during retransmission as another worker thread may be doing something as well. ........ 2006-10-30 18:04 +0000 [r46466] Matthew Fredrickson * channels/chan_zap.c: Make sure we give the linkset number, not the offset in the linksets array 2006-10-30 18:02 +0000 [r46461] Olle Johansson * channels/chan_sip.c: Small conversion to ast_channel_unlock 2006-10-30 17:32 +0000 [r46459] Matthew Fredrickson * channels/chan_zap.c: Specify which linkset we're getting the messages from in the message 2006-10-30 16:59 +0000 [r46439] Olle Johansson * main/rtp.c: In debug mode, recognize that someone is sending zrtp, even though we can't do anything with it yet. Ideally a first step would be a passthrough mode. 2006-10-30 16:50 +0000 [r46436] Matthew Fredrickson * channels/chan_zap.c: Don't make errors when we don't need them 2006-10-30 16:33 +0000 [r46379-46434] Olle Johansson * include/asterisk/file.h, include/asterisk/doxyref.h, /, channels/chan_sip.c, main/ast_expr2f.c, include/asterisk/module.h, formats/format_ogg_vorbis.c, main/app.c, include/asterisk/channel.h, include/asterisk/lock.h, include/asterisk/frame.h, main/asterisk.c, apps/app_voicemail.c: Issue 8246 Doxygen updates (kshumard) THANK YOU! * /: The RTCP patch started in trunk, so don't start all over again :-) * main/asterisk.c: Small formatting changes * main/rtp.c: Bind RTCP to the same IP as RTP. I currently don't see this as a bug that needs to be fixed in 1.4/1.2 too, but feel free to backport if you see it that way. RTCP now binds to ALL IP addresses on the host, RTP to a specific address. * /, channels/chan_sip.c: Issue #7869 - Stop retransmission of 302 redirects. * /, channels/chan_sip.c: Issue #7608 - Notifications sent with wrong content-type (imported from 1.2, 1.4) * /: Block patch from other branch * channels/chan_sip.c: Issues related to issue #7828 - segfault with MWI subscriptions and realtime. * /, channels/chan_sip.c: - Fix the OUTGOING stuff (merge from 1.4) - Make sure we UNREF authpeer when not needed * apps/app_voicemail.c: Spelling fix. * channels/chan_sip.c: Documentation update again * channels/chan_sip.c: Documentation update (I guess) * channels/chan_sip.c: Documentation correction * channels/chan_sip.c: maxtime is not needed any more now that we actually set the T1 timer based on the qualify result. * /, channels/chan_sip.c: Only accept message once * channels/chan_sip.c: Adding documentation inspired by a virtual drink with an anonymous man in New Jersey * channels/chan_sip.c: Don't duplicate function if not needed... - removing transmit_reinvite_with_t38_sdp in favour of adding an argument to transmit_reinvite_with_sdp * /, channels/chan_sip.c: Merge from 1.4 : Don't send 183 reliably... * channels/chan_sip.c: - Don't lock the dialoglist during the whole destruction of a single SIP dialog. Only lock when needed - when we remove the dialog from the dialog list If this doesn't lead to severe problems, it might help with some locking issues in 1.4/1.2. - Remove the term "interface" as a synonym for a SIP dialog. Sorry, Mark, but no one understands it... ;-) 2006-10-28 16:39 +0000 [r46378] Joshua Colp * utils/Makefile, /: Merged revisions 46377 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46377 | file | 2006-10-28 12:37:44 -0400 (Sat, 28 Oct 2006) | 2 lines Don't build muted on OpenBSD, it is not supported. ........ 2006-10-27 19:28 +0000 [r46372] BJ Weschke * apps/app_queue.c: Let's make sure we hold the mutex lock before we go looking at values in the queue structure that could potentially be changing while we're running. 2006-10-27 19:04 +0000 [r46371] Russell Bryant * channels/chan_zap.c, /: Merged revisions 46370 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46370 | russell | 2006-10-27 14:03:32 -0500 (Fri, 27 Oct 2006) | 4 lines move the copy of the default settings to the global settings back out of process_zap, so that they aren't overwritten when process_zap is called multiple times ........ 2006-10-27 18:59 +0000 [r46369] BJ Weschke * configs/queues.conf.sample, CHANGES, apps/app_queue.c: * Added option to run macro when a queue member is connected to a caller, see queues.conf.sample for details. * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and setqueueentryvar options for each queue, see queues.conf.sample for details. (#8216, jmls reported and submitted) 2006-10-27 18:31 +0000 [r46368] Olle Johansson * /, contrib/asterisk-ng-doxygen: raise the pressure on Christian :-) 2006-10-27 17:46 +0000 [r46366] Matthew Fredrickson * channels/chan_zap.c: First pass at implementation to be able to block and unblock zap channels for use. 2006-10-27 17:45 +0000 [r46365] Olle Johansson * channels/chan_sip.c: Put this patch on hold pending further testing... 2006-10-27 17:42 +0000 [r46359-46364] Russell Bryant * /, res/res_agi.c, apps/app_externalivr.c, res/res_musiconhold.c, main/asterisk.c: Merged revisions 46363 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46363 | russell | 2006-10-27 12:39:31 -0500 (Fri, 27 Oct 2006) | 5 lines We should always be using _exit() after a fork() or vfork() instead of exit(). This is because exit() does some extra cleanup which in some implementations of vfork(), for example, can actually modify the state of the parent process, causing very weird bugs or crashes. (issue #7971, Nick Gavrikov) ........ * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: Add the ability to customize some of the prompts used within the voicemail application by configuring them in voicemail.conf (issue #7415, patch by fkasumovic, with some fixes and documentation updates by myself) * channels/chan_zap.c, /: Merged revisions 46358 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46358 | russell | 2006-10-27 10:32:40 -0500 (Fri, 27 Oct 2006) | 5 lines Instead of iterating all of the options once to look for jitterbuffer options, and then again for everything else, move the processing of jitterbuffer options into the main loop so that there are no erroneous messages about ignoring unknown options. (issue #8226) ........ 2006-10-27 11:18 +0000 [r46354] Christian Richter * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h, channels/misdn/isdn_msg_parser.c, configs/misdn.conf.sample, channels/misdn/isdn_lib.c, channels/misdn_config.c: Merged revisions 46351-46353 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r46351 | crichter | 2006-10-27 11:49:20 +0200 (Fr, 27 Okt 2006) | 9 lines Merged revisions 46176 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r46176 | crichter | 2006-10-25 10:41:59 +0200 (Mi, 25 Okt 2006) | 1 line added nttimeout option to configure wether we disconnect calls on NT timeouts or not during an overlapdial session ........ ................ r46352 | crichter | 2006-10-27 11:58:44 +0200 (Fr, 27 Okt 2006) | 1 line fixed not compile issue, which was just introduced ................ r46353 | crichter | 2006-10-27 12:03:23 +0200 (Fr, 27 Okt 2006) | 9 lines Merged revisions 46350 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r46350 | crichter | 2006-10-27 11:24:01 +0200 (Fr, 27 Okt 2006) | 1 line fixed a bug which caused chan_misdn to try to allocate 2 times the same channel on high load, which then caused instability of mISDN. removed a useless function from isdn_lib.c ........ ................ 2006-10-26 20:27 +0000 [r46348] Jason Parker * /, apps/app_page.c: Merged revisions 46347 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46347 | qwell | 2006-10-26 15:25:44 -0500 (Thu, 26 Oct 2006) | 2 lines Fix small formatting issue, that causes misaligned line ........ 2006-10-26 20:22 +0000 [r46346] Olle Johansson * channels/chan_sip.c: Show if the channel is ready for video or T.38 udptl 2006-10-26 18:04 +0000 [r46341] Jason Parker * contrib/scripts/astgenkey.8: oops - somebody forgot to change this - long ago, probably. 2006-10-26 17:52 +0000 [r46330-46339] Russell Bryant * main/pbx.c, apps/app_osplookup.c, main/manager.c, apps/app_meetme.c, apps/app_festival.c, main/say.c, apps/app_alarmreceiver.c, apps/app_sms.c, apps/app_rpt.c, main/rtp.c, apps/app_voicemail.c: fix various spelling mistakes in comments (issue #8237, jmls) * /, main/translate.c: Merged revisions 46329 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46329 | russell | 2006-10-26 11:31:05 -0500 (Thu, 26 Oct 2006) | 11 lines - If the source has no audio or no video portion, do not call powerof() to get the format index. - Don't run through the audio and video loops if there is no audio or video portion of the source If 0 is passed to powerof, it will return -1. This value of -1 was then being used as an array index in these loops, which caused a crash on some systems. Other than this issue, this code works as we expected it to. If a format is not in the source, and we have to translation path to it, it is not offered in the list of acceptable destination formats. (fixes issue #8231) ........ 2006-10-26 12:47 +0000 [r46308-46319] Luigi Rizzo * main/manager.c: fix a problem that i recently introduced when the manager receives long commands. * configs/sip.conf.sample: document the match_auth_username option 2006-10-26 04:19 +0000 [r46299] Russell Bryant * /, doc/backtrace.txt: Merged revisions 46298 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46298 | russell | 2006-10-25 23:18:00 -0500 (Wed, 25 Oct 2006) | 2 lines update backtrace documentation to reflect changes in 1.4 (issue #8230, kshumard) ........ 2006-10-26 01:38 +0000 [r46288] Mark Spencer * main/manager.c, main/config.c: Fix comment preservation code (thanks murf!) 2006-10-25 20:21 +0000 [r46259-46277] Olle Johansson * /, channels/chan_sip.c: Old todo: Don't add Contact headers on BYE and CANCEL. * channels/chan_sip.c: First stab at transaction direction fix, this for trunk for testing * /, channels/chan_sip.c: Ugly code to try to remove issue discovered by Luigi as well as attack bug #7608 2006-10-25 19:24 +0000 [r46256] Matthew Fredrickson * channels/chan_zap.c: Send CPG when we get a CONTROL_PROGRESS frame and make sure that it sends ACM (not CPG) when we get CONTROL_PROCEEDING. 2006-10-25 19:14 +0000 [r46251] Matthew Fredrickson * channels/chan_zap.c, configs/zapata.conf.sample: Update changes to do US style point code parsing/formatting (xxx.xxx.xxx) 2006-10-25 19:10 +0000 [r46250] Russell Bryant * /, apps/app_queue.c: Merged revisions 46249 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46249 | russell | 2006-10-25 14:08:18 -0500 (Wed, 25 Oct 2006) | 2 lines update warning message to include "agi" option (issue #8225, jmls) ........ 2006-10-25 17:12 +0000 [r46238] Kevin P. Fleming * /, sounds/sounds.xml, sounds/Makefile: Merged revisions 46237 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46237 | kpfleming | 2006-10-25 12:08:58 -0500 (Wed, 25 Oct 2006) | 2 lines add support for prebuilt G.722 prompts and music on hold files ........ 2006-10-25 16:01 +0000 [r46215-46224] Olle Johansson * /, channels/chan_sip.c: Merge from 1.4 * /: Block change to 1.4 to block change to 1.2... This is confusing, but I think I got it right. 2006-10-25 14:55 +0000 [r46201-46203] Kevin P. Fleming * /, channels/chan_sip.c, main/translate.c, include/asterisk/translate.h: Merged revisions 46082-46083,46152-46153 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46082 | kpfleming | 2006-10-23 22:45:42 -0500 (Mon, 23 Oct 2006) | 2 lines add an API call to allow channel drivers to determine which media formats are compatible (passthrough or transcode) with the format an existing channel is already using ........ r46083 | kpfleming | 2006-10-23 22:53:32 -0500 (Mon, 23 Oct 2006) | 2 lines ensure that the translation matrix is properly lock-protected every place it is used ........ r46152 | kpfleming | 2006-10-24 18:45:19 -0500 (Tue, 24 Oct 2006) | 2 lines if multiple translators are registered for the same source/dest combination, ensure that the lowest-cost one is always inserted earlier in the list ........ r46153 | kpfleming | 2006-10-24 19:10:54 -0500 (Tue, 24 Oct 2006) | 2 lines code zone experiment: don't offer formats in the outbound INVITE that aren't either passthrough or translatable ........ * channels/chan_iax2.c: restore bugfix that was reverted by trunk_mtu patch * channels/chan_sip.c, /, apps/app_record.c, apps/app_softhangup.c, res/res_adsi.c, main/utils.c, pbx/dundi-parser.c, apps/app_ices.c, apps/app_getcpeid.c, apps/app_queue.c, channels/chan_iax2.c, main/cli.c, main/cdr.c, channels/chan_phone.c, pbx/pbx_spool.c, channels/chan_features.c, channels/chan_h323.c, pbx/pbx_ael.c, channels/chan_alsa.c, pbx/pbx_realtime.c, apps/app_sms.c, channels/chan_nbs.c, main/image.c, main/db.c, channels/chan_mgcp.c, cdr/cdr_custom.c, apps/app_parkandannounce.c, apps/app_voicemail.c: Merged revisions 46200 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46200 | kpfleming | 2006-10-25 09:32:08 -0500 (Wed, 25 Oct 2006) | 2 lines apparently developers are still not aware that they should be use ast_copy_string instead of strncpy... fix up many more users, and fix some bugs in the process ........ 2006-10-25 14:26 +0000 [r46199] Olle Johansson * CREDITS, configs/iax.conf.sample, channels/chan_iax2.c: Ok, second attempt... 2006-10-25 14:18 +0000 [r46198] Luigi Rizzo * CHANGES: document a couple of recently introduced feature also including the version number where the feature appeared. 2006-10-25 14:14 +0000 [r46183-46197] Olle Johansson * CREDITS, configs/iax.conf.sample, channels/chan_iax2.c: On the other hand, don't use 1.4 patches for trunk... Sorry. * CREDITS, configs/iax.conf.sample, channels/chan_iax2.c: Add ability to adapt the IAX trunk packets to the MTU size, to avoid bad audio when the number of channels fill the MTU on a given link. In the future, this needs to be configurable per peer with trunking enabled. * channels/chan_sip.c: Adding comments in the source is more persistent than just adding them to the commit message :-) * channels/chan_sip.c: Always add doxygen comments to new functions, more lines than one are appreciated really. (Read the coding guidelines). I've worked hard to make chan_sip a better place to code in, let's keep it that way and don't add more stuff without comments. Thank you. 2006-10-25 00:32 +0000 [r46155] Kevin P. Fleming * main/frame.c, /, main/translate.c, formats/format_pcm.c, channels/chan_h323.c, channels/chan_iax2.c, include/asterisk/frame.h, main/rtp.c: Merged revisions 46154 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46154 | kpfleming | 2006-10-24 19:26:17 -0500 (Tue, 24 Oct 2006) | 2 lines add passthrough and file format support for G.722 16KHz audio (issue #5084, original patch by andrew, updated by mithraen) ........ 2006-10-24 20:22 +0000 [r46141] Mark Spencer * res/res_agi.c: Fix FastAGI to not wait for the non-existant pid 2006-10-24 19:33 +0000 [r46131] Joshua Colp * /, channels/chan_iax2.c: Merged revisions 46130 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46130 | file | 2006-10-24 15:29:56 -0400 (Tue, 24 Oct 2006) | 2 lines We need to initialize our scheduler pthread condition... yes. ........ 2006-10-24 17:14 +0000 [r46104-46120] Luigi Rizzo * main/manager.c: i really think it is safe to commit this version, that simplifies the manager queue handling as described in the comment, and will make a lot easier to make further work on this code. * channels/chan_sip.c: correct fix for the bug i previously introduced - the strings are meant to be always initialized, independently from their content. 2006-10-24 05:24 +0000 [r46094] Russell Bryant * Makefile, /: Merged revisions 46093 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46093 | russell | 2006-10-24 01:23:33 -0400 (Tue, 24 Oct 2006) | 3 lines Restore the ability to remove the firmware directory without causing the installation to fail (issue #8111) ........ 2006-10-24 03:15 +0000 [r46081] Kevin P. Fleming * doc/imapstorage.txt, /: Merged revisions 46080 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46080 | kpfleming | 2006-10-23 22:13:08 -0500 (Mon, 23 Oct 2006) | 2 lines simplify and correct voicemail IMAP storage build instructions ........ 2006-10-24 03:09 +0000 [r46079] Tilghman Lesher * main/channel.c, /: Merged revisions 46078 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46078 | tilghman | 2006-10-23 22:01:00 -0500 (Mon, 23 Oct 2006) | 3 lines Pass through a frame if we don't know what it is, rather than trying to pass a NULL, which will segfault a channel driver (Bug 8149) ........ 2006-10-24 01:28 +0000 [r46055-46068] Russell Bryant * utils/muted.c, /, utils/ael_main.c: Merged revisions 46067 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46067 | russell | 2006-10-23 21:27:42 -0400 (Mon, 23 Oct 2006) | 7 lines In muted.c, check the return value of strdup. In ael_main.c, check the return value of calloc. (issue #8157) In passing fix a few minor bugs in ael_main.c. The last argument to strncpy() was a hard-coded 100, where it should have been 99. I changed this to use sizeof() - 1. ........ * /, apps/app_meetme.c: Merged revisions 46065 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46065 | russell | 2006-10-23 21:04:14 -0400 (Mon, 23 Oct 2006) | 2 lines Fix the descriptions of some of the MeetMeAdmin options (issue #8098, mflorell) ........ * channels/chan_sip.c: Fix a seg fault on a registration. Line 7706, in parse_register_contact, explicitly passes NULL as the "pass" argument to this function. 2006-10-23 21:46 +0000 [r46003-46045] Luigi Rizzo * channels/chan_sip.c: Unlike ast_strdup(), ast_strdupa() does not take a NULL pointer as argument, so fix the places where this might happen. This is also a fix that ought to go into 1.4 [The difference between the two functions is a bit confusing, and in asterisk i believe all string handling functions should be able to handl a NULL string as argument, but changing the API in trunk and not in 1.4 would make backporting harder.] * channels/chan_sip.c: remove a useless check for ocseq = 0. As discussed on the mailing lists, 0 is a legal value for Cseq, so there is no point to treat it specially. * channels/chan_sip.c: get_header() always returns a non-NULL value, so checking for NULL is certainly wrong and usually disables the checks that we want to make instead. This commit fixes a number of the above bugs where the result of get_header() is immediately checked for NULL. This is certainly a candidate for merging into 1.4 * channels/chan_sip.c: put another duplicated block of code in a function. * channels/chan_sip.c: reformat a statement and comment a potentially wrong assignement (altering state on an unvalidated message). * channels/chan_sip.c: Remove unnecessary casts from const char * to char *, if necessary by slightly rearranging the code. * channels/chan_sip.c: another use for parse_uri(). On passing, remove a wrong comment (that probably I wrote myself!) and introduce a temporary variable to avoid a misleading cast. 2006-10-23 17:08 +0000 [r46000] Russell Bryant * /, res/res_jabber.c: Merged revisions 45999 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r45999 | russell | 2006-10-23 13:07:45 -0400 (Mon, 23 Oct 2006) | 2 lines don't crash when an incoming message has no "from" (issue #8205, jmls) ........ 2006-10-23 16:54 +0000 [r45945-45989] Luigi Rizzo * main/utils.c: use autodetected support for gethostbyname_r * channels/chan_sip.c: + make sure parse_uri never returns NULL pointers - this simplifies its usage. + add another client for parse_uri, in handling Contact: strings (on passing, document the content of the "fullcontact" field); + in register_verify(), mark with XXX what i believe is another misinterpretation on the URI format when '@' is missing. No code changed here, so no fixes applied. * channels/chan_sip.c: After reading better the SIP RFC on sip URI (19.1.1) fix parse_uri() to interpret a missing userinfo section as a domain-only URI, and comment a wrong interpretation of the above in check_user_full(). The function has been patched to preserve the existing behaviour (in what admittedly is a corner case, but could be received under attacks). Hopefully the From: based matching will go away soon! * channels/chan_sip.c: in function get_also_info(), move argument stripping before splitting around the @, otherwise the refer_to_domain might contain arguments as well, causing failures. I think this is a true bug that ought to be fixed in 1.4 as well. * channels/chan_sip.c: start putting the URI parsing code in one place, introducing the function parse_uri() that splits a URI in its components. Right now use it only in one place, because the custom parsing that is done here and there sometimes has bugs that i want to figure out first. * channels/chan_sip.c: put common code in function terminate_uri() so we need to fix it only in one place. * channels/chan_sip.c: More cleanup of check_user_full with no functional change apart from a small (but disabled by default) new option. In detail: + introduce a new value for enum check_auth_result, AUTH_DONT_KNOW, used (read below) when a function does not have a conclusive response. Possibly this is the same as AUTH_NOT_FOUND, but need to check further. + move the large blocks (checking in the users list and in the peers list, respectively) from check_user_full() to separate functions. They return AUTH_DONT_KNOW in case they don't find a match, so the caller know that it has to try the next method. There is still some duplication of code here, but i have not tried yet to remove it. + [new option] a new option in sip.conf, match_auth_username, has been introduced, and disabled by default. If set, and the incoming request carries authentication info, the username to match in the users list is taken from there rather than from the From: field. This change is easy to identify, being made of - one line to declare the variable match_auth_username - a block of 15 lines in check_user_full() - one line in sip list settings - two lines for parsing the config file. check_user_full() is now a lot cleaner - basically a sequence of checks that are applied to the request. This will help future work with new matching schemes. 2006-10-23 00:33 +0000 [r45929] Joshua Colp * /, cdr/cdr_odbc.c: Merged revisions 45928 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r45928 | file | 2006-10-22 20:27:39 -0400 (Sun, 22 Oct 2006) | 10 lines Merged revisions 45927 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r45927 | file | 2006-10-22 20:25:28 -0400 (Sun, 22 Oct 2006) | 2 lines Don't leak memory mmmk? ........ ................ 2006-10-22 21:57 +0000 [r45917] Christian Richter * channels/chan_misdn.c, /: Merged revisions 45916 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r45916 | crichter | 2006-10-22 23:44:46 +0200 (Sun, 22 Oct 2006) | 9 lines Merged revisions 45808 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r45808 | crichter | 2006-10-21 14:35:13 +0200 (Sat, 21 Oct 2006) | 1 line fixed issue, that if chan_misdn is loaded and couldn't be initialized it would cause a segfault after 'reload'. Reported by Drew/Matt thx. ........ ................ 2006-10-22 21:08 +0000 [r45904-45915] Luigi Rizzo * channels/chan_sip.c: more streamlining of check_user_full * channels/chan_sip.c: simplify the flow of function check_user_full() A large block needs reindentation now, but we don't do that because it can be moved to a separate function. * channels/chan_sip.c: put duplicated code in functions. 2006-10-22 19:34 +0000 [r45893] Russell Bryant * configure, include/asterisk/autoconfig.h.in: regenerate the configure script and autoconfig.h.in to reflect recent changes for https support for the built in http server 2006-10-22 19:09 +0000 [r45858-45892] Luigi Rizzo * main/Makefile, configure.ac, main/http.c, configs/http.conf.sample: Fix a few issues in the previous (disabled) HTTPS code, and support linux as well (using fopencookie(), which should be available in glibc). Update configure.ac to check for funopen (BSD) and fopencookie(glibc), and while we are at it also for gethostbyname_r (the generated files need to be updated, or you need to run bootstrap.sh yourself). Document the new options in http.conf.sample (names are only tentative, better ones are welcome). At this point we can safely enable the option. Anyone willing to try this on Sun and Apple platforms ? * main/http.c: Implement https support. The changes are not large. Most of the diff comes from putting the global variables describing an accept session into a structure, so we can reuse the existing code for running multiple accept threads on different ports. Once this is done, and if your system has the funopen() library function (and ssl, of course), it is just a matter of calling the appropriate functions to set up the ssl connection on the existing socket, and everything works on the secure channel now. At the moment, the code is disabled because i have not implemented yet the autoconf code to detect the presence of funopen(), and add -lssl to main/Makefile if ssl libraries are present. And a bit of documentation on the http.conf arguments, too. If you want to manually enable https support, that is very simple (step 0 1 2 will be eventually detected by ./configure, the rest is something you will have to do anyways). 0. make sure your system has funopen(3). FreeBSD does, linux probably does too, not sure about other systems. 1. uncomment the following line in main/http.c // #define DO_SSL /* comment in/out if you want to support ssl */ 2. add -lssl to AST_LIBS in main/Makefile 3. add the following options to http.conf sslenable=yes sslbindport=4433 ; pick one you like sslcert=/tmp/foo.pem ; path to your certificate file. 4. generate a suitable certificate e.g. (example from mini_httpd's Makefile: openssl req -new -x509 -days 365 -nodes -out /tmp/foo.pem -keyout /tmp/foo.pem and here you go: https://localhost:4433/asterisk/manager now works. * main/http.c: it is useless and possibly wrong to use ast_cli() to send the reply back to http clients. Use fprintf/fwrite instead, since we are already using a FILE * to read the input. If you wonder why, this is because it makes it trivial to implement https support (as long as your system has funopen()). And this is what i am going to put in with the next few commits... 2006-10-22 04:44 +0000 [r45847] Joshua Colp * Makefile, main/Makefile: Let's have build.h created a bit earlier so that func_version can use it and not stop the build on a fresh machine that has never had Asterisk installed on it before... 2006-10-21 20:24 +0000 [r45836] Luigi Rizzo * main/http.c: the default port number was erroneously stored in host order, and reading from the config file used ntohs instead of htons. this ought to be merged to 1.4 as well. 2006-10-21 18:52 +0000 [r45820] Joshua Colp * /, main/loader.c: Merged revisions 45817 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r45817 | file | 2006-10-21 14:48:58 -0400 (Sat, 21 Oct 2006) | 2 lines Don't use promotion on Darwin because it doesn't seem to work quite right in all cases, this should solve the unresolved symbol issue people have been seeing. ........ 2006-10-21 18:50 +0000 [r45819] Russell Bryant * /, res/res_monitor.c: Merged revisions 45818 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r45818 | russell | 2006-10-21 14:49:46 -0400 (Sat, 21 Oct 2006) | 3 lines Add a couple missing unregistrations of manager actions and remove duplicate unregistrations of applications. (issue #8194, jmls) ........ 2006-10-20 20:59 +0000 [r45786] Luigi Rizzo * channels/chan_sip.c: introduce sip_pvt_lock() and sip_pvt_unlock() wrappers to lock these data structures. This improve readability, and also hides the underlying locking mechanism so it is a lot easier to add diagnostic code, or move the object locks somewhere else, etc. On passing, rename the lock field in sip_pvt to pvt_lock, also for ease of readability. 2006-10-20 19:04 +0000 [r45776] Joshua Colp * Makefile, /: Merged revisions 45775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r45775 | file | 2006-10-20 15:03:03 -0400 (Fri, 20 Oct 2006) | 2 lines Pass DESTDIR and ASTSBINDIR so that the utilities get installed in the proper location (reported on asterisk-dev mailing list) ........ 2006-10-20 15:54 +0000 [r45764] Russell Bryant * channels/chan_sip.c: put the constants for whether methods can create a dialog or not in an enum 2006-10-20 11:24 +0000 [r45753] Luigi Rizzo * main/manager.c: minor comment changes, code rearrangement and field renaming to minimize diffs with future modifications. The current implementation is problematic for the following reasons: + all insertions are O(N) because the event list does not have a tail pointer; + there is only a single lock protecting both session and users queues. + the implementation of the queue itself is not documented. I think i have figured it out, more or less, but am unclear on whether there is proper locking in place The rewrite (which i have working locally) uses a tailq so insertions are O(1), separate locks for the event and session queues, and has a documented implementation so hopefully we can figure out if/where bug exist. 2006-10-20 08:14 +0000 [r45742-45743] Olle Johansson * /, channels/chan_sip.c: Let's repair the SIP attack shield :-) * main/manager.c: Doxygen corrections 2006-10-19 22:06 +0000 [r45712-45724] Steve Murphy * funcs/func_version.c (added): This new function, VERSION(), created via bug report 8176, may help dialplan programmers in the future. In the meantime, they can use the algorithm I outline on the bug report notes; If anyone invents something better, I'd hope they post it * utils/astman.c: astman was slightly weirding out over the new Dial and Newcallerid events 2006-10-19 17:26 +0000 [r45696] Luigi Rizzo * main/manager.c: more fixes to comments and very minor code rearrangement. 2006-10-19 17:25 +0000 [r45693-45695] Joshua Colp * /, res/res_jabber.c: Merged revisions 45694 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r45694 | file | 2006-10-19 13:24:40 -0400 (Thu, 19 Oct 2006) | 2 lines Let's remember to unregister JabberStatus too (issue #8184 reported by jmls) ........ * /, apps/app_externalivr.c: Merged revisions 45692 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r45692 | file | 2006-10-19 13:19:47 -0400 (Thu, 19 Oct 2006) | 10 lines Merged revisions 45691 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r45691 | file | 2006-10-19 13:16:37 -0400 (Thu, 19 Oct 2006) | 2 lines Respect language selection when seeing if the file exists (issue #8178 reported by mnicholson) ........ ................ 2006-10-19 17:07 +0000 [r45690] Luigi Rizzo * main/manager.c: implement proper XML/HTML formatting of multiple messages (e.g. the result of waitevent). Also fix some comments. 2006-10-19 16:06 +0000 [r45679] Joshua Colp * /, channels/chan_sip.c: Merged revisions 45678 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r45678 | file | 2006-10-19 12:03:09 -0400 (Thu, 19 Oct 2006) | 2 lines If the jitterbuffer is forced on then we can't partially bridge (reported by wangster on #asterisk-dev) ........ 2006-10-19 10:05 +0000 [r45648-45668] Luigi Rizzo * channels/chan_sip.c: move a large block out of do_monitor() and into a function, to improve readability. * channels/chan_sip.c: + move the definition of netlock as it was not related to the comment just above; + decouple the struct definition and variable declaration (iflist); * main/manager.c: more documentation of data structure and functions. Of interest: + ast_get_manager_by_name_locked() is now without the ast_ prefix as it is a local function; + unuse_eventqent() renamed to unref_event(), and returns the pointer to the next entry. + marked with XXX a couple of usages of unref_event() because i suspect we are addressing the wrong entry. 2006-10-19 07:17 +0000 [r45647] Olle Johansson * /, channels/chan_sip.c: Cleaning up... Removing duplicate (again) 2006-10-19 02:16 +0000 [r45634] Kevin P. Fleming * channels/chan_sip.c, include/asterisk/threadstorage.h: restore freeing of threadstorage objects without custom cleanup functions allow custom threadstorage init functions to return failure use a custom init function for chan_sip's temp_pvt, to improve performance a bit 2006-10-19 01:04 +0000 [r45623-45624] Russell Bryant * /, channels/chan_sip.c: Merge fix to not leak the stringfields of a thread speicif sip_pvt. This also includes the fix not to leak the actual sip_pvt. Merged revisions 45622 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r45622 | russell | 2006-10-18 20:59:51 -0400 (Wed, 18 Oct 2006) | 2 lines Don't leak the actual thread-specific sip_pvt struct ........ * main/channel.c, main/frame.c, main/manager.c, channels/chan_sip.c, channels/chan_skinny.c, main/logger.c, main/utils.c, channels/iax2-parser.c, include/asterisk/threadstorage.h, main/cli.c: Extend the thread storage API such that a custom initialization function can be called for each thread specific object after they are allocated. Note that there was already the ability to define a custom cleanup function. Also, if the custom cleanup function is used, it *MUST* call free on the thread specific object at the end. There is no way to have this magically done that I can think of because the cleanup function registered with the pthread implementation will only call the function back with a pointer to the thread specific object, not the parent ast_threadstorage object. 2006-10-18 22:40 +0000 [r45611] Luigi Rizzo * main/manager.c: silent warning from a debugging message (which will go away soon, anyways) 2006-10-18 22:19 +0000 [r45610] Joshua Colp * apps/app_meetme.c, CHANGES: Just for Nicholson - here's an option, C, to Meetme that will allow it to continue in the dialplan if the person is kicked out. (issue #7994 reported by mnicholson with mods by myself) 2006-10-18 21:41 +0000 [r45597-45599] Luigi Rizzo * main/manager.c: remove trailing whitespace * main/manager.c: ouch! remember to unlink temporary files once done with them. * main/manager.c: + move output_format variables in the http section of the file; + more comments on struct mansession and global variables; + small improvements to the session matching code so it supports multiple sessions from the same IP 2006-10-18 21:05 +0000 [r45596] Joshua Colp * /, main/asterisk.c: Merged revisions 45595 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r45595 | file | 2006-10-18 17:03:34 -0400 (Wed, 18 Oct 2006) | 2 lines Don't modify things if we are using vfork as this is very bad and may cause unexpected behavior (issue #7970 reported by Nick Gavrikov) ........ 2006-10-18 17:53 +0000 [r45572-45583] Luigi Rizzo * main/manager.c: another bunch of comments on the data structures. * main/manager.c: despite the large changes, this commit only moves functions around so that functions belonging to the same group are close to each other. At the beginning of each group i have added a bit of documentation to explain what the group does and what is the typical flow - basically, all i have learned by code inspection over the past few days should be documented for you to read. I have not put many doxygen annotations just because i am not sure what are the proper ones. Hopefully some doxygen experts will jump in. Next on the plate: try to figure out how "struct eventqent" are supposed to work. * main/manager.c: more comment and formatting fixes, small simplifications to functions get_input() and session_do() 2006-10-18 16:45 +0000 [r45571] Matt O'Gorman * main/manager.c: rizzo compile then commit, maybe even run it too ^_^ 2006-10-18 15:49 +0000 [r45529-45561] Luigi Rizzo * main/manager.c: comment and cleanup the main thread. On passing, fix a bug: close the socket if the allocation of a structure for the new session fails. (the bugfix is a candidate for 1.4) * main/manager.c: create a new (internal, for the time being) function astman_start_ack() to start manager responses that need further lines. This removes a lot of duplicate code from the various handlers that at the moment build an ActionID string themselves. Once settled, the function should move to manager.h so it can be used by other files (chan_agent, chan_iax2, chan_sip, chan_zap, res_jabber and app_queue). I am not totally clear if there is a preferred position for the ActionID: line in a message. Some instances put it at the end, but one would argue that it is preferable to have it at the beginning. * main/manager.c: more indentation cleanup from previous commits, and remove the "busy" field from struct mansession as it was not used correctly anyways. * main/manager.c: create proper handlers for "Challenge" and "Login" actions, rather than use inline code for them. Things are more readable this way, and also error processing is more consistent. * main/manager.c: fix indentation from a commit of a couple of days ago * main/manager.c: another batch of simplifications to authenticate() (they are committed a bit at a time so it is easier to revert them in case we find a bug at a later time). 2006-10-18 12:15 +0000 [r45528] Olle Johansson * /, channels/chan_sip.c: Remove duplicate declarations... 2006-10-18 11:59 +0000 [r45463-45518] Luigi Rizzo * main/manager.c, configs/manager.conf.sample: remove unused fields and unimplemented options. * main/manager.c: first pass as simplifying authenticate(), avoiding whitespace changes * main/manager.c: more code simplifications * main/manager.c: simplify ast_strings_to_mask * main/manager.c: add a comment to remember that a block of code is completely redundant. * main/manager.c: + move the enum declaration for output formats near the head of the file, so it can be used from more places; + make the declaration of contenttype[] more robust; + remove the wrappers around __xml_translate(), since they were used only in one place, and rename to xml_translate(). This allows for a bit of simplifications. + document the output produced by the above function. * main/manager.c: merge xml_translate() and html_translate() into one function since they do similar things. Add a small form on top of the html output so request like http://foo:8088/asterisk/manager will suggest you what to do. Note: i suspect there is still a bug somewhere in the session matching code, as sometimes you have to login twice in order for the following commands to be recognised. Apart from this, the cli now is basically usable from a web form! * main/http.c: introduce uri_decode() so that '+' are translated into ' ' (e.g. browsers do this when they encode input strings from a form). * main/http.c: various code simplifications to reduce nesting depth, minor optimizations to avoid extra calls of strlen(), and some variable localization. One feature worth backporting is the move of ast_variables_destroy() to a different place in handle_uri() to avoid leaking memory in case a uri is not found. 2006-10-18 03:03 +0000 [r45453] Joshua Colp * /, main/rtp.c: Merged revisions 45452 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r45452 | file | 2006-10-17 23:02:08 -0400 (Tue, 17 Oct 2006) | 2 lines Don't segfault if you're using a channel driver that doesn't turn RTCP on ........ 2006-10-18 02:46 +0000 [r45440-45442] Russell Bryant * main/channel.c, /: Merged revisions 45441 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r45441 | russell | 2006-10-17 22:41:36 -0400 (Tue, 17 Oct 2006) | 7 lines Don't attempt to access private data members of the pthread_mutex_t object, because this does not work on all linux systems. Instead, just access the reentrancy field in the ast_mutex_info struct when DEBUG_THREADS is enabled. If DEBUG_CHANNEL_LOCKS is enabled, the developer probably has DEBUG_THREADS on as well. (issue #8139, me) ........ * configs/sip_notify.conf.sample, /: Merged revisions 45439 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r45439 | russell | 2006-10-17 22:19:07 -0400 (Tue, 17 Oct 2006) | 2 lines update entry to reboot a snom phone (issue #7850, pnlarsson) ........ 2006-10-17 23:06 +0000 [r45426] Steve Murphy * res/res_agi.c: As per bug 6779, this patch is now applied to trunk; while I was at it, I corrected a reference to a CLI command, to follow the new regime. 2006-10-17 22:32 +0000 [r45409-45411] Kevin P. Fleming * /, build_tools/prep_tarball (added): Merged revisions 45410 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r45410 | kpfleming | 2006-10-17 17:31:54 -0500 (Tue, 17 Oct 2006) | 2 lines add a project-specific script to be used during release preparation ........ * main/channel.c, /, channels/chan_sip.c, channels/chan_iax2.c, include/asterisk/stringfields.h, main/ast_expr2.c: Merged revisions 45408 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r45408 | kpfleming | 2006-10-17 17:24:10 -0500 (Tue, 17 Oct 2006) | 3 lines optimize the 'quick response' code a bit more... no more malloc() or memset() for each response expand stringfields API a bit to allow reusing the stringfield pool on a structure when needed, and remove some unnecessary code when the structure was being freed ........ 2006-10-17 21:09 +0000 [r45379-45398] Joshua Colp * main/manager.c: Warning be gone! * /, channels/chan_sip.c: Merged revisions 45378 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r45378 | file | 2006-10-17 16:30:34 -0400 (Tue, 17 Oct 2006) | 2 lines Don't create a "real" pvt structure for requests that shouldn't be able to create one. Instead use a temporary pvt and fill it with enough information so we can send a reply. ........ 2006-10-17 19:57 +0000 [r45365] Olle Johansson * channels/chan_sip.c, doc/channelvariables.txt: Issue #5484 (branch sipdiversion) - Support for Diversion header in redirects of calls with 302 redirection. (tinning) 2006-10-17 18:08 +0000 [r45351] Luigi Rizzo * main/manager.c: simplify authority_to_str() using ast_build_string() 2006-10-17 17:54 +0000 [r45335] Olle Johansson * channels/chan_sip.c: Issue #7254 - Add support of "423 Interval too brief" to outbound SIP registrations. Thanks, tardieu! 2006-10-17 17:51 +0000 [r45334] Luigi Rizzo * main/manager.c: Improve the XML formatting of responses coming from web interface. Normal responses are sequences of lines of the form "Name: value", with \r\n as line terminators and an empty line as a response terminator. Generi CLI commands, however, do not have such a clean formatting, and the existing code failed to generate valid XML for them. Obviously we can only use heuristics here, and we do the following: - accept either \r or \n as a line terminator, trimming trailing whitespace; - if a line does not have a ":" in it, assume that from this point on we have unformatted data, and use "Opaque-data:" as a name; - if a line does have a ":" in it, the Name field is not always a legal identifier, so replace non-alphanum characters with underscores; All the above is to be refined as we improve the formatting of responses from the CLI. And, all the above ought to go as a comment in the code rather than just in a commit message... 2006-10-17 17:51 +0000 [r45331-45333] Olle Johansson * /, configs/sip.conf.sample: Update of docs * channels/chan_sip.c: - Remove unneeded code that won't be reached now that we kill responses to unkonwn dialogs earlier in the process. - move debug message. 2006-10-17 17:41 +0000 [r45330] Luigi Rizzo * main/manager.c: open a temporary file to receive the output from cli commands invoked through the http interface. It is not terribly efficient but better than no output at all. Todo: use a configurable /tmp directory instead of a hardwired one. 2006-10-17 17:22 +0000 [r45328] Kevin P. Fleming * /, LICENSE: Merged revisions 45327 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r45327 | kpfleming | 2006-10-17 12:22:25 -0500 (Tue, 17 Oct 2006) | 10 lines Merged revisions 45326 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r45326 | kpfleming | 2006-10-17 12:22:01 -0500 (Tue, 17 Oct 2006) | 2 lines provide licensing language for IAXy firmware file ........ ................ 2006-10-17 17:19 +0000 [r45325] Luigi Rizzo * main/manager.c: document xml_copy_escape() and add an extra function, namely replace non-alphanum chars with underscore. This is useful when building field names in xml formatting. 2006-10-17 16:27 +0000 [r45295-45316] Olle Johansson * /: ...block this one too... Only applies to 1.4 since the fix for trunk was different. * /: Block patch from 1.4 that does not apply here. * channels/chan_sip.c: Get rid of the ignore variable that was only partially replaced by the flag. 2006-10-16 20:26 +0000 [r45234-45286] Joshua Colp * channels/chan_sip.c, configs/sip.conf.sample: In the course of a data this has been turned into an option to ignore replies, then ignore responses and finally I'm just getting rid of the option altogether and making it the default no matter what. C'est la vie! * /: Woof. * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 45280 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r45280 | file | 2006-10-16 16:06:18 -0400 (Mon, 16 Oct 2006) | 10 lines Merged revisions 45265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r45265 | file | 2006-10-16 15:59:54 -0400 (Mon, 16 Oct 2006) | 2 lines Use responses rather then replies even though they mean the same thing. ........ ................ * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 45262 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r45262 | file | 2006-10-16 15:37:34 -0400 (Mon, 16 Oct 2006) | 10 lines Merged revisions 45260 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r45260 | file | 2006-10-16 15:28:18 -0400 (Mon, 16 Oct 2006) | 2 lines Add 'ignoreoodreplies' option which will not create a pvt structure on a SIP response but instead basically drop it. ........ ................ * apps/app_directed_pickup.c: It's new directed pickup! This now features a more sane way of finding the channel to pick up (I snuck it into the tree on Friday... bet you didn't know I'd actually use it eh?). PICKUPMARK now also works in a different way, you should prefix it with _ when setting it so it gets inherited onto the channel(s) created in app_dial as directed pickup will now look for it on the target channel, not the originating channel. (BE-85) 2006-10-16 14:03 +0000 [r45224] Olle Johansson * CREDITS, /: Update 2006-10-16 14:00 +0000 [r45219] Luigi Rizzo * main/manager.c: + comment some unclear fields of struct mansession; + let some commands (Challenge, Login) be processed even if already authenticated, as it doesn't harm and prevents some incorrect error messages + remove custom code for Logoff - the existing handler was ok. Some indentation fixes may be necessary 2006-10-16 13:20 +0000 [r45194-45209] Olle Johansson * channels/chan_sip.c: When adding new functions, please add a forward declaration. I *know* it is not required, but it makes navigation easier and will help when splitting up this large source code file. Thank you! * /, channels/chan_sip.c: Importing rev 45196 from 1.4 - don't kill dialog for a bad response * channels/chan_sip.c: A B2BUA should *not* issue proxy auth. 2006-10-16 11:29 +0000 [r45151-45185] Luigi Rizzo * main/manager.c: + comment some unclear requirements for master_eventq + remove the need for an snprintf in astman_get_header() + fix comment for manager list eventq + localize one variable and minor code simplifications. * main/manager.c: protect access to first_action with actionlock. Mark with XXX one place (during command execution) where navigation should be protected with actionlock, but is not because it would block requests for a long time. To solve this properly we need to put reference counts in the struct manager_action. A suboptimal fix is to copy the record on a search and then unlock the list while we work on the copy. * main/http.c: comment some functions, and more small code simplifications * main/http.c: fix indentation of a large block after changes in previous commit (basically whitespace only). * main/http.c: simplify string parsing routines using ast_skip_*() functions. * main/http.c: don't forget to close a descriptor on a malloc failure. On passing, small rearrangement of the code to reduce indentation. There is a bit more cleanup planned for this file, so a merge to 1.4 will be done when it is all done. * main/http.c: typo: serer -> server 2006-10-14 04:36 +0000 [r45142] Steve Murphy * funcs/func_rand.c: update the doc string for both AEL and extensions.conf users. 2006-10-13 23:03 +0000 [r45126] Kevin P. Fleming * /, main/acl.c: Merged revisions 45125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r45125 | kpfleming | 2006-10-13 18:02:48 -0500 (Fri, 13 Oct 2006) | 7 lines ------------------------------------------------------------------------ r45119 | kpfleming | 2006-10-13 17:57:42 -0500 (Fri, 13 Oct 2006) | 2 lines don't drop the entire permit/deny list when an attempt is made to add an invalid entry (BE-92) ------------------------------------------------------------------------ ........ 2006-10-13 21:20 +0000 [r45105-45109] Joshua Colp * apps/app_dial.c: Inherit the context and extension until the channel is answered * /, res/res_speech.c: Merged revisions 45106 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r45106 | file | 2006-10-13 17:06:09 -0400 (Fri, 13 Oct 2006) | 2 lines Clear the quiet flag too since we are restarting a recognition again (reported on -dev by Stephan Edelman) ........ * /, res/res_speech.c: Merged revisions 45104 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r45104 | file | 2006-10-13 17:01:13 -0400 (Fri, 13 Oct 2006) | 2 lines Check return value from engine in case of failure (ie: out of licenses) (reported on -dev mailing list) ........ 2006-10-13 19:24 +0000 [r45089] Christian Richter * channels/chan_misdn.c, /: Merged revisions 45088 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r45088 | crichter | 2006-10-13 21:19:46 +0200 (Fr, 13 Okt 2006) | 1 line avoiding warning, fixing potential bug ........ 2006-10-13 18:45 +0000 [r45080] Joshua Colp * codecs/lpc10/median.c, codecs/lpc10/encode.c, codecs/lpc10/ivfilt.c, /, codecs/lpc10/bsynz.c, codecs/lpc10/prepro.c, codecs/lpc10/invert.c, codecs/lpc10/placev.c, codecs/lpc10/irc2pc.c, codecs/lpc10/decode.c, codecs/lpc10/dcbias.c, codecs/lpc10/pitsyn.c, codecs/lpc10/difmag.c, codecs/lpc10/voicin.c, codecs/lpc10/synths.c, codecs/lpc10/preemp.c, codecs/lpc10/hp100.c, codecs/lpc10/lpfilt.c, codecs/lpc10/rcchk.c, codecs/lpc10/mload.c, codecs/lpc10/lpcenc.c, codecs/lpc10/vparms.c, codecs/lpc10/dyptrk.c, codecs/lpc10/lpcini.c, codecs/lpc10/random.c, codecs/lpc10/ham84.c, codecs/lpc10/chanwr.c, codecs/lpc10/placea.c, codecs/lpc10/tbdm.c, codecs/lpc10/analys.c, codecs/lpc10/onset.c, codecs/lpc10/energy.c, codecs/lpc10/lpcdec.c, codecs/lpc10/deemp.c: Merged revisions 45079 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r45079 | file | 2006-10-13 14:42:49 -0400 (Fri, 13 Oct 2006) | 2 lines And file said... let the compiler warnings STOP! ........ 2006-10-13 18:08 +0000 [r45078] Steve Murphy * pbx/ael/ael-test/ref.ael-vtest17 (added), pbx/ael/ael-test/ael-vtest17/extensions.ael (added), pbx/ael/ael-test/ael-vtest17 (added), pbx/ael/ael-test/ref.ael-test3, pbx/pbx_ael.c: Correction for bug 8128 in trunk 2006-10-13 17:06 +0000 [r45052-45067] Joshua Colp * /, apps/app_chanspy.c: Merged revisions 45066 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r45066 | file | 2006-10-13 13:05:02 -0400 (Fri, 13 Oct 2006) | 10 lines Merged revisions 45060 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r45060 | file | 2006-10-13 13:01:22 -0400 (Fri, 13 Oct 2006) | 2 lines Turn on volume adjustment if it needs to be on (issue #8136 reported by mnicholson) ........ ................ * /, apps/app_playback.c: Merged revisions 45051 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r45051 | file | 2006-10-13 12:20:58 -0400 (Fri, 13 Oct 2006) | 2 lines Move say.conf existence check to do_say function since it is called from multiple places (issue #8144 reported by kshumard) ........ 2006-10-13 16:20 +0000 [r45050] Kevin P. Fleming * /, channels/chan_iax2.c: Merged revisions 45049 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r45049 | kpfleming | 2006-10-13 11:19:35 -0500 (Fri, 13 Oct 2006) | 10 lines Merged revisions 45048 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r45048 | kpfleming | 2006-10-13 11:18:08 -0500 (Fri, 13 Oct 2006) | 2 lines when sending a call to a peer, use the proper socket if we have multiple bindings (reported on asterisk-dev) ........ ................ 2006-10-13 16:02 +0000 [r45032-45047] Joshua Colp * /, channels/chan_sip.c: Merged revisions 45040 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r45040 | file | 2006-10-13 12:01:17 -0400 (Fri, 13 Oct 2006) | 2 lines Complete merging in RPID screen changes (issue #8101 reported by hristo, patch by oej in revision 44757) ........ * main/dnsmgr.c, /: Merged revisions 45031 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r45031 | file | 2006-10-13 11:53:22 -0400 (Fri, 13 Oct 2006) | 10 lines Merged revisions 45030 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r45030 | file | 2006-10-13 11:49:53 -0400 (Fri, 13 Oct 2006) | 2 lines Pass the right value to usleep for sleeping, and always add the background refresh item back into the scheduler if enabled since it is deleted during reload. (issue #8142 reported by p_lindheimer) ........ ................ 2006-10-13 15:47 +0000 [r45029] Kevin P. Fleming * /, configure, include/asterisk/autoconfig.h.in, configure.ac, main/utils.c: Merged revisions 45027 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r45027 | kpfleming | 2006-10-13 10:41:14 -0500 (Fri, 13 Oct 2006) | 2 lines use a configure script test for PMTU discovery control instead of just assuming it's available on Linux ........ 2006-10-13 15:42 +0000 [r45028] Christian Richter * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged revisions 45026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r45026 | crichter | 2006-10-13 16:45:39 +0200 (Fr, 13 Okt 2006) | 9 lines Merged revisions 45020 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r45020 | crichter | 2006-10-13 15:11:13 +0200 (Fr, 13 Okt 2006) | 1 line fixed some echocandisable issues when bridged. this caused a kernel panic sometimes..also some minor formatting fixes ........ ................ 2006-10-13 11:18 +0000 [r45009-45010] Luigi Rizzo * channels/chan_sip.c: Try to avoid the use of 'z' modifier in cases where it is not necessary - rather, cast the argument to int. In this case, the string is in a UDP packet and as such limited to 64k so its length can be safely represented in an int without truncation (besides, this is just a debugging message!) * channels/chan_sip.c: arguments to auth_headers() needed to be swapped here. To avoid the same mistake in the future (due to slightly confusing variable names), add a comment. On passing, remove a redundant initialization. 2006-10-13 08:23 +0000 [r45000] Christian Richter * /, channels/misdn/isdn_msg_parser.c: Merged revisions 44994 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r44994 | crichter | 2006-10-13 09:52:41 +0200 (Fr, 13 Okt 2006) | 9 lines Merged revisions 44993 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r44993 | crichter | 2006-10-13 09:40:07 +0200 (Fr, 13 Okt 2006) | 1 line fixed issue, that the hangupcause got a wrong isdn cause at RELEASE_COMPLETE ........ ................ 2006-10-12 20:41 +0000 [r44983] Matt O'Gorman * /, channels/chan_gtalk.c: Merged revisions 44982 via svnmerge from https://svn.digium.com/svn/asterisk/branches/1.4 ........ r44982 | mogorman | 2006-10-12 15:34:49 -0500 (Thu, 12 Oct 2006) | 2 lines fix for bug 7764. ........ 2006-10-12 19:16 +0000 [r44957-44973] Kevin P. Fleming * channels/chan_sip.c: eliminate compiler warning * /, channels/chan_sip.c: Merged revisions 44971 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44971 | kpfleming | 2006-10-12 14:14:24 -0500 (Thu, 12 Oct 2006) | 2 lines we can only send one 'a=ptime' attribute per media session, not one for each format ........ * include/asterisk/utils.h, /, channels/chan_sip.c, main/utils.c, main/netsock.c: Merged revisions 44956 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r44956 | kpfleming | 2006-10-12 13:38:51 -0500 (Thu, 12 Oct 2006) | 10 lines Merged revisions 44955 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r44955 | kpfleming | 2006-10-12 13:31:26 -0500 (Thu, 12 Oct 2006) | 2 lines ensure that IAX2 and SIP sockets allow UDP fragmentation when running on Linux (thanks to Brian Candler on the asterisk-dev list for the tip) ........ ................ 2006-10-12 16:57 +0000 [r44944-44946] Russell Bryant * main/manager.c, /: Merged revisions 44945 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44945 | russell | 2006-10-12 12:56:32 -0400 (Thu, 12 Oct 2006) | 2 lines fix a silly typo in a comment that I saw while reading the commit list ........ * pbx/pbx_dundi.c: put flags in an enum and remove a couple of unused defines 2006-10-12 16:11 +0000 [r44943] Joshua Colp * Makefile, /: Merged revisions 44942 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44942 | file | 2006-10-12 12:08:50 -0400 (Thu, 12 Oct 2006) | 2 lines Pass off AUDIO_LIBS so muted can link on OSX (issue #8135 reported by ssokol) ........ 2006-10-12 15:12 +0000 [r44933] Luigi Rizzo * channels/chan_sip.c: + move [almost] all instances of WWW-Authenticate/Proxy-Authenticate and friends in a function, auth_headers(), which is used to simplify the interface of do_{proxy|register}_auth(). + use PROXY_AUTH = 407, WWW_AUTH = 401 as values for enum sip_auth_type; No functional change, only code cleanup. 2006-10-12 13:04 +0000 [r44922] Nadi Sarrar * main/manager.c, /: Merged revisions 44921 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44921 | nadi | 2006-10-12 14:55:25 +0200 (Do, 12 Okt 2006) | 2 lines append_event must be called while holding the session lock ........ 2006-10-12 10:26 +0000 [r44912] Russell Bryant * /, res/res_jabber.c: Merged revisions 44911 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44911 | russell | 2006-10-12 06:24:36 -0400 (Thu, 12 Oct 2006) | 2 lines change some debug output to use LOG_DEBUG instead of verbose output ........ 2006-10-11 23:36 +0000 [r44900-44901] Luigi Rizzo * channels/chan_sip.c: reduce indentation of two large blocks * channels/chan_sip.c: operator != also works between booleans... 2006-10-11 16:57 +0000 [r44889] Jason Parker * /, main/db1-ast/Makefile: Merged revisions 44888 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44888 | qwell | 2006-10-11 11:57:06 -0500 (Wed, 11 Oct 2006) | 3 lines These are already set by the parent Makefile.. There is no need to have this here (it doesn't actually work anyways). ........ 2006-10-11 13:45 +0000 [r44876-44877] Russell Bryant * doc/linkedlists.txt (removed): Remove doc/linkedlists.txt as it is no longer needed. The top of the file reads: As of 2004-12-23, this documentation is no longer maintained. The doxygen documentation generated from linkedlists.h should be referred to in its place, as it is more complete and better maintained. * channels/chan_sip.c: Revert Luigi's accidental commit of his local changes when debugging some SIP authentication issues. This was committed in revision 44844, where the commit message was just "small formatting cleanup", so I am pretty sure he didn't mean to commit this part. 2006-10-11 13:21 +0000 [r44844-44875] Luigi Rizzo * channels/chan_sip.c: remove duplicate prototypes. Have not checked if there are more. * channels/chan_sip.c: simplify and comment handle_response_peerpoke() * channels/chan_sip.c: fix indentation of a function after previous commit (whitespace-only change) * channels/chan_sip.c: handle_response_peerpoke() does not need to return anything. (Reindentation in the next commit.) * channels/chan_sip.c: small formatting cleanup 2006-10-11 08:45 +0000 [r44840-44843] Christian Richter * channels/chan_misdn.c, /: Merged revisions 44563 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r44563 | crichter | 2006-10-06 14:53:41 +0200 (Fr, 06 Okt 2006) | 9 lines Merged revisions 44460 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r44460 | crichter | 2006-10-05 12:02:38 +0200 (Do, 05 Okt 2006) | 1 line fixed segfault which happens during hold/transfer action ........ ................ * channels/chan_misdn.c, /: Merged revisions 44562 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r44562 | crichter | 2006-10-06 14:52:01 +0200 (Fr, 06 Okt 2006) | 9 lines Merged revisions 44335 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r44335 | crichter | 2006-10-04 17:26:59 +0200 (Mi, 04 Okt 2006) | 1 line if INFORMATION Message come with keypad instead of called party number, we just use the keypad as called party number. ........ ................ * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample, channels/misdn/isdn_lib.c, channels/misdn_config.c: Merged revisions 44561 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r44561 | crichter | 2006-10-06 14:50:25 +0200 (Fr, 06 Okt 2006) | 9 lines Merged revisions 44334 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r44334 | crichter | 2006-10-04 17:13:58 +0200 (Mi, 04 Okt 2006) | 1 line added the option 'reject_cause' to make it possible to set the RELEASE_COMPLETE - cause on the 3. incoming PMP channel, which is automatically rejected because chan_misdn does not support that kind of callwaiting. Therefore chan_misdn supports now 3 incoming channels on a PMP BRI Port. misdn_lib_get_free_bc now gets the info if the requested channel is incoming or outgoing to make the 3. channel possible ........ ................ * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged revisions 44559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r44559 | crichter | 2006-10-06 12:44:34 +0200 (Fr, 06 Okt 2006) | 9 lines Merged revisions 44149 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r44149 | crichter | 2006-10-02 15:28:14 +0200 (Mo, 02 Okt 2006) | 1 line fixed the hold/retrieve/transfer issues, removed a useless bc field, added setting of frame.delivery fields, some minor code cleanups ........ ................ 2006-10-10 20:52 +0000 [r44831] Tilghman Lesher * apps/app_rpt.c: More whitespace fixes 2006-10-10 17:23 +0000 [r44820] Joshua Colp * /, channels/chan_sip.c: Merged revisions 44819 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44819 | file | 2006-10-10 13:21:44 -0400 (Tue, 10 Oct 2006) | 2 lines Move some stuff around so that a NOTIFY dialog won't hang around until the end of the world under certain circumstances ........ 2006-10-10 16:46 +0000 [r44810] Tilghman Lesher * /, funcs/func_logic.c: Merged revisions 44808 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44808 | tilghman | 2006-10-10 11:42:19 -0500 (Tue, 10 Oct 2006) | 2 lines Lost of a bit of logic when this was simplified between 1.2 and 1.4 (Bug 8117) ........ 2006-10-10 16:31 +0000 [r44789-44807] Joshua Colp * /, channels/chan_sip.c: Merged revisions 44806 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44806 | file | 2006-10-10 12:30:00 -0400 (Tue, 10 Oct 2006) | 2 lines Bail out if we have no refer structure and we get a refer response ........ * /, channels/chan_sip.c: Merged revisions 44788 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44788 | file | 2006-10-10 11:23:14 -0400 (Tue, 10 Oct 2006) | 2 lines Only set DTMF information if an RTP structure exists ........ 2006-10-10 14:54 +0000 [r44787] Christian Richter * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged revisions 44786 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r44786 | crichter | 2006-10-10 15:50:26 +0200 (Di, 10 Okt 2006) | 9 lines Merged revisions 44785 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r44785 | crichter | 2006-10-10 15:34:33 +0200 (Di, 10 Okt 2006) | 1 line (re)added support of dynamical enabling hdlc on bchannels ........ ................ 2006-10-10 08:08 +0000 [r44770-44774] Luigi Rizzo * channels/chan_sip.c: clarify the use of the standard SIP port number, 5060, and rename the old DEFAULT_SIP_PORT as STANDARD_SIP_PORT to make it clear that this is not something we can change, unlike other defaults. * channels/chan_sip.c: improve formatting of SIP packets when dumped to the verbose output stream, so it is easier to find them in the log. 2006-10-09 18:23 +0000 [r44768] Joshua Colp * funcs/func_timeout.c: Timeout values are in seconds (issue #7122 reported by jmls) 2006-10-09 16:15 +0000 [r44765] Jason Parker * /, channels/chan_skinny.c: Merged revisions 44764 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44764 | qwell | 2006-10-09 11:12:35 -0500 (Mon, 09 Oct 2006) | 4 lines Fix a problem where phones that go "missing" never got unregistered. Issue #8067, reported by pj, patch by Anthony LaMantia (with minor whitespace modifications) ........ 2006-10-09 15:52 +0000 [r44762-44763] Joshua Colp * /, channels/chan_iax2.c: Merged revisions 44759 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44759 | file | 2006-10-09 11:41:28 -0400 (Mon, 09 Oct 2006) | 2 lines Properly avoid a collision with iax2_hangup (issue #8115 reported by vazir) ........ 2006-10-09 11:20 +0000 [r44753] Olle Johansson * channels/chan_sip.c: Being pedantic... "media" is easier to understand than "data" in the function name... :-) 2006-10-09 09:04 +0000 [r44745-44752] Luigi Rizzo * channels/chan_sip.c: slightly restructure sipsock_read() removing a "goto" * channels/chan_sip.c: use S_OR in one place * channels/chan_sip.c: update_call_counter(): indentation fixes and small simplifications at the top of the function. * channels/chan_sip.c: localize some variables and reduce nesting depth (indentation will be fixed by a separate commit). * channels/chan_sip.c: small simplification to initreqprep() * channels/chan_sip.c: Simplify function parse_request() using a single loop instead of two very similar ones. * channels/chan_sip.c: do not dereference p if we know it is NULL. 2006-10-07 20:42 +0000 [r44697-44731] Olle Johansson * channels/chan_sip.c: Fix some debug output for setsockopt for TOS * channels/chan_sip.c: - move definition of global_autoframing to the same place as other globals - set initial value at load/reload - Add questionmarks for someone to fill in for doxygen * channels/chan_sip.c: Add/change doxygen and comments * configs/sip.conf.sample: Recommend using "sip reload" since it's much easier to learn and remember. * channels/chan_sip.c: Explain usage of DEFAULT_SIP_PORT * channels/chan_sip.c: Do *NOT* use DEFAULT_SIP_PORT in these comparisions, since users may change that, but the protocol clearly states that if we DO NOT mention a port it is 5060. DEFAULT_SIP_PORT is whatever we default to listen to. I believe it's the third time I revert a patch like this. 2006-10-07 14:48 +0000 [r44685-44686] Paul Cadach * /, channels/h323/ast_h323.cxx, channels/chan_h323.c, channels/h323/ast_h323.h, channels/h323/chan_h323.h: Merged revisions 44684 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44684 | pcadach | 2006-10-07 20:39:34 +0600 (Сбт, 07 Окт 2006) | 1 line Propagate caller's transfer capability too ........ * include/asterisk/callerid.h, main/callerid.c, CHANGES, funcs/func_callerid.c: Extend CALLERID() function for "pres" and "ton" values 2006-10-07 12:50 +0000 [r44641-44675] Luigi Rizzo * channels/chan_sip.c: slightly restructure the code that computes the channel's name * channels/chan_sip.c: put repeated code to set nat mode in a function. * channels/chan_sip.c: put common code in a function to avoid repetitions. * channels/chan_sip.c: remove hardwired usage of 5060, use DEFAULT_SIP_PORT instead * channels/chan_sip.c: improve and document function get_in_brackets(), introducing a helper function find_closing_quote() of more general use. * channels/chan_sip.c: when possible, use ast_set2_flags instead of ast_set/ast_clr . Also mark XXX some dubious places. 2006-10-06 21:29 +0000 [r44632] Kevin P. Fleming * /, include/asterisk/linkedlists.h: Merged revisions 44631 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44631 | kpfleming | 2006-10-06 16:28:03 -0500 (Fri, 06 Oct 2006) | 2 lines ensure that mutex locks inside list heads are initialized properly on platforms that require constructor initialization (issue #8029, patch from timrobbins) ........ 2006-10-06 21:10 +0000 [r44630] Joshua Colp * /, main/rtp.c: Merged revisions 44628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44628 | file | 2006-10-06 17:08:54 -0400 (Fri, 06 Oct 2006) | 2 lines Remove the seqno check for RFC2833, the handler is smart enough to not need it. ........ 2006-10-06 21:04 +0000 [r44616-44626] Luigi Rizzo * main/manager.c: basically fix indentation of a large function after previous simplifications. On passing, use a single exit point. (once done with the cleanup i will merge the changes into 1.4, if applicable) * main/manager.c: s cannot be null here, so remove the useless test and error-handling block. * main/manager.c: simplify logic in preparation to reduce indentation 2006-10-06 18:47 +0000 [r44606] Joshua Colp * /, main/rtp.c: Merged revisions 44605 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44605 | file | 2006-10-06 14:46:28 -0400 (Fri, 06 Oct 2006) | 2 lines When the sequence number rolls over then reset the recorded sequence number for DTMF (issue #8106 reported by bungalow) ........ 2006-10-06 17:27 +0000 [r44595] Tilghman Lesher * apps/app_rpt.c: Massive cleanup of the rpt code, updating to current coding guidelines 2006-10-06 16:56 +0000 [r44582] Joshua Colp * /, main/file.c: Merged revisions 44581 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r44581 | file | 2006-10-06 12:53:48 -0400 (Fri, 06 Oct 2006) | 10 lines Merged revisions 44580 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r44580 | file | 2006-10-06 12:52:14 -0400 (Fri, 06 Oct 2006) | 2 lines Even more frames to treat as though the remote side disappeared (issue #8097 reported by eldadran) ........ ................ 2006-10-06 16:43 +0000 [r44566-44579] Luigi Rizzo * configs/sip.conf.sample: document a bit the use of templates. They are highly convenient for writing configuration files, especially if you have many similar entries, or want to switch quickly between different configurations without having to comment/uncomment large sections of the files. * configs/sip.conf.sample: document the "contact" option a bit better. * res/res_limit.c: help old bsd-system which don't have RLIMIT_AS and use RLIMIT_VMEM for virtual memory limits. * main/manager.c, main/http.c: make sure sockets are blocking when they should be blocking. * channels/chan_sip.c, configs/sip.conf.sample: Two things: 1. slightly rearrange/simplify the parsing of the argument in sip_register. This brings in a patch that has been in Mantis (5834) for ages, and is the larger part of the commit; 2. implement the "contact" option for peers, similar to the one in users.conf: If you put a "contact" option with a non-empty argument (e.g. contact=123) in a peer section, asterisk will register with the provider as if you had a register= username:secret@host/contact line in the general section. The latter is a very small is a new feature so i am not putting it in the 1.4 branch, although the "contact" option in user.conf is already in the 1.4 branch and so it wouldn't be too strange to merge it. Note that the implementation of "contact" is much simpler than the one in 5834, and limited to a few lines in build_peer(). 2006-10-06 09:01 +0000 [r44545] Olle Johansson * channels/chan_sip.c: Remove deprecated "incominglimit" config option 2006-10-06 06:43 +0000 [r44537] Luigi Rizzo * configs/sip.conf.sample: update example commands to match current syntax (does not apply to 1.4) 2006-10-06 02:24 +0000 [r44527] Russell Bryant * configure, include/asterisk/autoconfig.h.in: regenerate the configure script to reflect the latest changes done by Luigi Rizzo 2006-10-05 20:13 +0000 [r44503-44516] Joshua Colp * apps/app_voicemail.c: Fix indenting a bit (issue #8082 reported by selsky) * /, main/file.c: Merged revisions 44502 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r44502 | file | 2006-10-05 15:57:16 -0400 (Thu, 05 Oct 2006) | 10 lines Merged revisions 44501 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r44501 | file | 2006-10-05 15:55:41 -0400 (Thu, 05 Oct 2006) | 2 lines Treat busy control frames as hangup in the file streaming core (issue #8097 reported by eldadran) ........ ................ 2006-10-05 18:29 +0000 [r44489] Steve Murphy * pbx/pbx_ael.c: These mods fix a problem pointed out by dgartang, where in certain situations, the target of a goto cannot be found, even right under your nose. This is because the current context is not updated properly, and rather than waste time and find why and where the context should have been updated, I just use my newly added 'dad' ptrs, and pop until I have either the context or extension, and use that instead. 2006-10-05 18:03 +0000 [r44487] Joshua Colp * /, channels/chan_sip.c: Merged revisions 44486 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44486 | file | 2006-10-05 14:01:51 -0400 (Thu, 05 Oct 2006) | 2 lines One more T.38 fix! Don't leave a reinvite hanging by a thread if the other side is already setup with T.38 ........ 2006-10-05 16:11 +0000 [r44477] Kevin P. Fleming * /, main/app.c: Merged revisions 44476 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44476 | kpfleming | 2006-10-05 11:10:01 -0500 (Thu, 05 Oct 2006) | 3 lines don't segfault when an argument without a close parenthesis is found stop parsing as soon as that situation occurs ........ 2006-10-05 15:42 +0000 [r44467] Luigi Rizzo * configure.ac, acinclude.m4: Basically, this commit only simplifies configure.ac and makes the mechanism more flexible, but otherwise should not affect your build even if you regenerate the "configure" script. (Most likely you need to run bootstrap.sh as you really need to re-run autoheader for reasons that i do not completely understand). If you don't regenerate "configure", of course you will see no difference. In detail: - restructure the check for mandatory modules to remove some redundant code blocks; - extend the AST_EXT_LIB_CHECK so that it can used also for checking headers; - define the AST_C_DEFINE_CHECK macro to test for #defined symbols; - for the two above macros, add a last argument that getscopied into HAVE_$1_VERSION so the source can adapt to different versions of the same libraries/header/etc - document the above; - document a problem that existed before and i did not manage to solve: the 'description' argument to AC_DEFINE does not substiture shell variables so you will not see the actual values in the comments (in autoconfig.h).. 2006-10-05 02:43 +0000 [r44451] Joshua Colp * /, channels/chan_sip.c: Merged revisions 44450 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44450 | file | 2006-10-04 22:40:40 -0400 (Wed, 04 Oct 2006) | 2 lines Don't totally bail out if T.38 was negotiated ........ 2006-10-05 01:43 +0000 [r44437] Kevin P. Fleming * utils/Makefile, /: Merged revisions 44436 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44436 | kpfleming | 2006-10-04 20:42:06 -0500 (Wed, 04 Oct 2006) | 2 lines this change was correct, the old version is no longer needed ........ 2006-10-05 01:40 +0000 [r44435] Steve Murphy * main/pbx.c, apps/app_read.c, apps/app_waitforring.c, CHANGES, apps/app_speech_utils.c: As per ToDo list, I have made it so that Wait(), WaitExten(), Congestion(), Busy(), Read(), WaitForRing(), will now either actually handle a floating point argument as advertised, or has been upgraded to accept a floating point [timeout] arg. 2006-10-05 01:30 +0000 [r44434] Kevin P. Fleming * /, channels/chan_sip.c: Merged revisions 44433 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r44433 | kpfleming | 2006-10-04 20:30:05 -0500 (Wed, 04 Oct 2006) | 10 lines Merged revisions 44432 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r44432 | kpfleming | 2006-10-04 20:27:57 -0500 (Wed, 04 Oct 2006) | 2 lines fix Polycom presence notification again ........ ................ 2006-10-04 23:52 +0000 [r44408-44423] Luigi Rizzo * configure.ac: simplify checks for OSS using AST_EXT_LIB_CHECK; remove two repeated blocks using better logic. * acinclude.m4: small formatting fix * acinclude.m4: when only checking headers, do not set $1_LIB. Also PBX_$1=0 is the default so we don't need to set it explicitly. * acinclude.m4: document, and extend a bit the macro AST_EXT_LIB_CHECK so that it can be used in more places in configure.ac * configure.ac: restore proper CPPFLAGS and LDFLAGS for FreeBSD, until a better solution is found. Please do not commit the regenerated "configure" file yet, as there are some more simplifications to be applied to configure.ac and acinclude.m4 in the next few days. For the same reason, i am postponing the commit to the 1.4 branch until the above changes are complete. * utils/Makefile: correct libraries for astman, at least so i think... * Makefile: put linker flags in ASTLDFLAGS where they belong 2006-10-04 21:20 +0000 [r44379-44394] Kevin P. Fleming * /, channels/chan_sip.c: Merged revisions 44393 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r44393 | kpfleming | 2006-10-04 16:17:30 -0500 (Wed, 04 Oct 2006) | 11 lines Merged revisions 44392 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r44392 | kpfleming | 2006-10-04 16:15:29 -0500 (Wed, 04 Oct 2006) | 3 lines remove workaround for old Polycom firmware SUBSCRIBE requests add workaround for new Polycom firmware SUBSCRIBE requests (bug is known to exist in 2.0.1 firmware) ........ ................ * include/asterisk.h, /, main/utils.c: Merged revisions 44390 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44390 | kpfleming | 2006-10-04 16:04:21 -0500 (Wed, 04 Oct 2006) | 2 lines make LOW_MEMORY builds actually work ........ * include/asterisk/utils.h, main/autoservice.c, main/dnsmgr.c, channels/chan_zap.c, res/res_snmp.c, /, apps/app_meetme.c, channels/chan_sip.c, main/utils.c, main/devicestate.c, res/res_musiconhold.c, res/res_jabber.c, apps/app_queue.c, channels/chan_iax2.c, channels/chan_oss.c, main/cdr.c, channels/chan_phone.c, main/manager.c, pbx/pbx_spool.c, res/res_smdi.c, channels/chan_skinny.c, channels/chan_h323.c, main/http.c, channels/chan_alsa.c, pbx/pbx_dundi.c, apps/app_mixmonitor.c, main/asterisk.c, channels/chan_mgcp.c: Merged revisions 44378 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44378 | kpfleming | 2006-10-04 14:47:22 -0500 (Wed, 04 Oct 2006) | 4 lines update thread creation code a bit reduce standard thread stack size slightly to allow the pthreads library to allocate the stack+data and not overflow a power-of-2 allocation in the kernel and waste memory/address space add a new stack size for 'background' threads (those that don't handle PBX calls) when LOW_MEMORY is defined ........ 2006-10-04 19:33 +0000 [r44336-44377] Steve Murphy * pbx/ael/ael-test/ael-test16 (added), pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/pbx_ael.c, pbx/ael/ael-test/ref.ael-test16 (added), pbx/ael/ael-test/ael-test16/extensions.ael: These changes resolve the problems in bug 8090, where there's a crash compiling an empty context * configs/muted.conf.sample: I've been meaning to add some explanation about muted... here it is * configs/manager.conf.sample: CLI reverbification update to this config file * apps/app_macro.c: Added a warning to the documentation for Macro in response to bug 7776 2006-10-04 00:26 +0000 [r44323] Kevin P. Fleming * Makefile, include/asterisk.h, /, main/asterisk.c, main/loader.c, main/term.c: Merged revisions 44322 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44322 | kpfleming | 2006-10-03 19:25:44 -0500 (Tue, 03 Oct 2006) | 3 lines ensure that local include files are always used avoid a duplicate function name (term_init()) ........ 2006-10-03 22:36 +0000 [r44313] Matt O'Gorman * /, channels/chan_gtalk.c, res/res_jabber.c: Merged revisions 44312 via svnmerge from https://svn.digium.com/svn/asterisk/branches/1.4 ........ r44312 | mogorman | 2006-10-03 17:35:43 -0500 (Tue, 03 Oct 2006) | 2 lines fix issue with dialing client without resource. ........ 2006-10-03 20:19 +0000 [r44299] Kevin P. Fleming * /, apps/app_queue.c: Merged revisions 44298 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r44298 | kpfleming | 2006-10-03 15:18:29 -0500 (Tue, 03 Oct 2006) | 10 lines Merged revisions 44296 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r44296 | kpfleming | 2006-10-03 15:14:13 -0500 (Tue, 03 Oct 2006) | 2 lines fix a logic error in my previous fix to the queue reload code ........ ................ 2006-10-03 20:17 +0000 [r44297] Joshua Colp * CHANGES, apps/app_queue.c: Strat becomes Strategy based on feedback from two nameless fellows 2006-10-03 18:47 +0000 [r44287] Paul Cadach * /, channels/h323/ast_h323.cxx: Merged revisions 44283,44286 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44283 | pcadach | 2006-10-04 00:30:48 +0600 (Срд, 04 Окт 2006) | 1 line Fix preparation of type and presentation of calling number ........ r44286 | pcadach | 2006-10-04 00:42:20 +0600 (Срд, 04 Окт 2006) | 1 line Change default presentation indicator to "user provided not screened" if octet 3a missed in CallingPartyNumber IE ........ 2006-10-03 18:37 +0000 [r44273-44285] Joshua Colp * /, channels/chan_sip.c: Merged revisions 44284 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44284 | file | 2006-10-03 14:35:55 -0400 (Tue, 03 Oct 2006) | 2 lines Use VideoSupport instead so it is considered a valid XML attribute name. (issue #8075 reported by renemendoza) ........ * CHANGES, apps/app_queue.c: Add 'Strat' manager field to QueueParams event. (issue #7704 reported by renemendoza) * main/channel.c, CHANGES: Add Masquerade manager event which trips when a masquerade happens (issue #7840 reported by moy) 2006-10-03 16:42 +0000 [r44263] Steve Murphy * pbx/ael/ael-test/ref.ael-test11, pbx/ael/ael-test/ref.ael-test14, pbx/ael/ael-test/ref.ael-test15, pbx/ael/ael-test/ref.ael-ntest9, pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-ntest12, pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3, pbx/pbx_ael.c, pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test5, pbx/ael/ael-test/ref.ael-test6, pbx/ael/ael-test/ref.ael-test7, pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-vtest13: These changes correspond to the changes to app_stack's Gosub() application 2006-10-03 16:15 +0000 [r44262] Joshua Colp * UPGRADE.txt: First entry! Tell people about the callerid changes with manager. 2006-10-03 15:53 +0000 [r44253] Matt O'Gorman * main/udptl.c, funcs/func_rand.c, main/say.c, apps/app_record.c, apps/app_test.c, funcs/func_strings.c, apps/app_alarmreceiver.c, apps/app_ices.c, channels/chan_iax2.c, main/loader.c, res/res_smdi.c, channels/chan_skinny.c, apps/app_zapscan.c, apps/app_zapras.c, main/http.c, channels/chan_alsa.c, apps/app_externalivr.c, cdr/cdr_odbc.c, main/db.c, main/sched.c, apps/app_dial.c, main/pbx.c, channels/chan_agent.c, apps/app_disa.c, channels/iax2-provision.c, apps/app_talkdetect.c, apps/app_db.c, res/res_monitor.c, channels/chan_misdn.c, apps/app_zapbarge.c, channels/chan_features.c, apps/app_macro.c, apps/app_voicemail.c, apps/app_meetme.c, res/res_musiconhold.c, channels/chan_gtalk.c, res/res_jabber.c, main/enum.c, cdr/cdr_csv.c, main/channel.c, channels/chan_phone.c, apps/app_osplookup.c, main/manager.c, apps/app_mp3.c, res/res_agi.c, main/logger.c, main/app.c, main/dns.c, channels/chan_mgcp.c, apps/app_nbscat.c, res/res_config_pgsql.c, channels/chan_zap.c, funcs/func_db.c, channels/chan_sip.c, apps/app_festival.c, apps/app_waitforsilence.c, res/res_adsi.c, res/res_crypto.c, apps/app_queue.c, main/rtp.c, cdr/cdr_tds.c, channels/chan_jingle.c, apps/app_directed_pickup.c, main/file.c, pbx/pbx_dundi.c, channels/chan_nbs.c, main/dsp.c: bug #8076 check option_debug before printing to debug channel. patch provided in bugnote, with minor changes. 2006-10-03 15:50 +0000 [r44252] Tilghman Lesher * apps/app_stack.c: Okay, I can't use ast_app_separate_args for that... and add some debugging for murf... 2006-10-03 15:48 +0000 [r44250-44251] Luigi Rizzo * configure.ac: comment the fact that autoconf2.59 is ok to process this file, but we want to use 2.60 in case the generated "configure" file must me committed back to the repository, so we keep differences to a minimum. * bootstrap.sh: simplify this file 2006-10-03 00:07 +0000 [r44241] Matt O'Gorman * include/asterisk/jabber.h, res/res_jabber.c: 44240 same as but without the removing of chan_jingle and such, as I hope to finish jingle support for 1.6 2006-10-02 22:02 +0000 [r44231] Tilghman Lesher * apps/app_stack.c: Use the standard parsing routines 2006-10-02 20:58 +0000 [r44200-44218] Joshua Colp * configs/queues.conf.sample, doc/channelvariables.txt, CHANGES, apps/app_queue.c: Expand setinterfacevar option to also set a variable, MEMBERNAME, which contains the member's name. (issue #8046 reported by jmls) * apps/app_dial.c, main/channel.c, apps/app_meetme.c, res/res_features.c, apps/app_dumpchan.c, CHANGES, apps/app_queue.c: Make callerid fields in Manager events more consistent. CallerIDNum for number and CallerIDName for name. (issue #7976 reported by suhler) * /, channels/chan_sip.c: Merged revisions 44215 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r44215 | file | 2006-10-02 16:11:02 -0400 (Mon, 02 Oct 2006) | 10 lines Merged revisions 44213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r44213 | file | 2006-10-02 16:07:59 -0400 (Mon, 02 Oct 2006) | 2 lines Change the fd on the I/O context in case it changed during the reload, which is indeed possible. (issue #7943 reported by eclubb) ........ ................ * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 44199 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r44199 | file | 2006-10-02 15:41:39 -0400 (Mon, 02 Oct 2006) | 10 lines Merged revisions 44198 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r44198 | file | 2006-10-02 15:39:59 -0400 (Mon, 02 Oct 2006) | 2 lines We should be using $AST_SBIN instead of hardcoding the path for the error message (issue #7942 reported by eclubb) ........ ................ 2006-10-02 19:01 +0000 [r44187-44196] Paul Cadach * configs/users.conf.sample, /, pbx/pbx_config.c: Merged revisions 44186 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44186 | pcadach | 2006-10-03 00:52:56 +0600 (Втр, 03 Окт 2006) | 1 line Missed part of userconf functionality for chan_h323 ........ * /, doc/realtime.txt: Merged revisions 44167 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44167 | pcadach | 2006-10-02 23:16:37 +0600 (Пнд, 02 Окт 2006) | 1 line Typo fix ........ * /, channels/chan_h323.c: Merged revisions 44166 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44166 | pcadach | 2006-10-02 23:15:11 +0600 (Пнд, 02 Окт 2006) | 1 line Optimization of oh323_indicate(): less locks - less problems, plus single exit point ........ 2006-10-02 17:54 +0000 [r44153-44172] Joshua Colp * main/logger.c, CHANGES, configs/logger.conf.sample: Add option to logger to rename log files with timestamp (issue #8020 reported by jmls) * /, main/io.c: Merged revisions 44169 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r44169 | file | 2006-10-02 13:25:13 -0400 (Mon, 02 Oct 2006) | 10 lines Merged revisions 44168 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r44168 | file | 2006-10-02 13:22:27 -0400 (Mon, 02 Oct 2006) | 2 lines Shrink when current_ioc is unused. It is set to -1 when unused, not 0. (issue #7941 reported by eclubb) ........ ................ * res/res_monitor.c: Get rid of the IS_NULL_STRING macro and use ast_strlen_zero instead (issue #8070 reported by wrmem) 2006-10-02 16:00 +0000 [r44152] Kevin P. Fleming * main/asterisk.c: clean up formatting and conformance to code guidelines revert Mark's change that caused a memory leak (cap_set_proc() does not free the capability structure so we always need to call cap_free()) 2006-10-02 15:40 +0000 [r44150] Joshua Colp * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Add option 'keepstats' which will keep queue statistics during a reload. (issue #7908 reported by jmls) 2006-10-02 04:17 +0000 [r44148] Tilghman Lesher * apps/app_stack.c: It makes more sense that in GosubIf that the two labels might have different arguments. 2006-10-02 02:38 +0000 [r44145-44147] Mark Spencer * channels/chan_sip.c, channels/chan_iax2.c: Don't use channel when you don't mean a channel * main/asterisk.c: Uhm yah, not sure who committed this into trunk... Anyway, I think this is what was intended... 2006-10-01 19:40 +0000 [r44136] Paul Cadach * /, channels/chan_h323.c: Merged revisions 44135 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44135 | pcadach | 2006-10-02 01:32:24 +0600 (Пнд, 02 Окт 2006) | 1 line Do not simulate any audio tones if we got PROGRESS message ........ 2006-10-01 18:30 +0000 [r44112-44126] Russell Bryant * Makefile, /: Merged revisions 44125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44125 | russell | 2006-10-01 14:30:06 -0400 (Sun, 01 Oct 2006) | 6 lines Fix a problem that cuased AST_DATA_DIR in defaults.h to be empty. The cause is that since ASTDATADIR is explicitly exported using "export ASTDATADIR" at the top of the Makefile, make no longer considers the variable "undefined", so the Makefile can't use ?= to set ASTDATADIR if not yet set. (issue #8063, reported by akohlsmith, fixed by me) ........ * /, configs/queues.conf.sample: Merged revisions 44111 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r44111 | russell | 2006-10-01 11:20:12 -0400 (Sun, 01 Oct 2006) | 11 lines Merged revisions 44110 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r44110 | russell | 2006-10-01 11:19:23 -0400 (Sun, 01 Oct 2006) | 3 lines Fix the name of the "eventmemberstatus" option in the sample queues.conf (issue #8065, adamg) ........ ................ 2006-10-01 05:37 +0000 [r44100] Tilghman Lesher * apps/app_zapateller.c: Janitor for Zapateller: convert to use argument macros 2006-09-30 19:23 +0000 [r44091] Paul Cadach * /, main/rtp.c: Merged revisions 44090 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44090 | pcadach | 2006-10-01 01:20:38 +0600 (Вск, 01 Окт 2006) | 1 line Allow one-way RTP streams (device->Asterisk) ........ 2006-09-30 16:37 +0000 [r44081] Luigi Rizzo * Makefile, main/Makefile, codecs/lpc10/Makefile: merge compile fixes from 44080: - with AST_DEVMODE, building codecs/lpc10 fails because of lots of warnings, and the configure step in editline fails as well. Fix this by removing the -Werror in these steps. - on FreeBSD (but probably on other platforms as well), the final link of asterisk fails because AST_LIBS was not exported to the subdirs Makefiles. Add a proper fix in the top-level Makefile (a possible alternative way is to add "export AST_LIBS" near the beginning of the file). With this fix, i believe that some of the platform-specific conditionals in main/Makefile are redundant (because they should be already dealt with in the top level Makefile) but i don't have a platform to check. 2006-09-30 16:15 +0000 [r44069-44079] Paul Cadach * /, channels/chan_sip.c: Merged revisions 44078 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44078 | pcadach | 2006-09-30 22:12:23 +0600 (Сбт, 30 Сен 2006) | 6 lines Fix issue #7928 correctly. Next is a comment of previous fix: Issue #7928 - Don't send both 404 and 503. Fix by phsultan with a small fix by me, myself or I. Thanks, Philippe! (This was caused by my changes to the transaction handling) ........ * /, channels/chan_sip.c: Merged revisions 44068 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44068 | pcadach | 2006-09-30 10:37:39 +0600 (Сбт, 30 Сен 2006) | 14 lines Found some buggy SIP clients (phones Planet VIP-153T firmware 1.0, Linksys PAP2 firmware 3.1.9(LSc)) which sends ACK not on OK message only (when remote party answers) but on RINGING message too, so when we send 200 OK message, we get unidentified ACK message (because INVITE acknowledged on RINGING message already), so 200 OK retransmits within its retransmission interval then call gets dropped. If someone else knows how to provide workaround for such cases, please, fix it in correct way. Thanks to ssh from #asteriskru for provide access to his box to study and fix this case. ........ 2006-09-29 22:52 +0000 [r44056-44058] Kevin P. Fleming * /, agi, utils: Merged revisions 44057 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44057 | kpfleming | 2006-09-29 17:51:53 -0500 (Fri, 29 Sep 2006) | 2 lines ignore temporary files made by the Makefiles during a build ........ * /, codecs/gsm/Makefile, configure.ac, Makefile.moddir_rules, Makefile.rules, pbx/Makefile, res/Makefile, channels/Makefile, agi/Makefile, codecs/Makefile, utils/Makefile, configure, build_tools/embed_modules.xml, codecs/ilbc/Makefile, codecs/lpc10/Makefile, main/db1-ast/Makefile: Merged revisions 44055 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44055 | kpfleming | 2006-09-29 17:47:40 -0500 (Fri, 29 Sep 2006) | 2 lines fix a few build system bugs, and convert Makefiles to be compatible with GNU make 3.80 ........ 2006-09-29 22:36 +0000 [r44054] Jason Parker * /, main/asterisk.c, main/cli.c: Merged revisions 44053 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44053 | qwell | 2006-09-29 15:35:09 -0700 (Fri, 29 Sep 2006) | 3 lines Fix a bug with the removal of 'atleast' argument to 'core verbose' and 'core debug'. Add that argument back in. ........ 2006-09-29 21:13 +0000 [r44044] Paul Cadach * /, channels/h323/ast_h323.cxx: Merged revisions 44034,44042-44043 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44034 | pcadach | 2006-09-30 02:43:13 +0600 (Сбт, 30 Сен 2006) | 1 line Fake display name by called number on incoming calls (until passing connected number/connected name is not implemented) ........ r44042 | pcadach | 2006-09-30 03:05:43 +0600 (Сбт, 30 Сен 2006) | 1 line Set TON/PRESENTATION information more carefully when no CallingNumber IE available ........ r44043 | pcadach | 2006-09-30 03:09:10 +0600 (Сбт, 30 Сен 2006) | 1 line Compile first, please ........ 2006-09-29 20:16 +0000 [r44033] Tilghman Lesher * apps/app_voicemail.c: Remove locking conflict 2006-09-29 19:16 +0000 [r44024-44025] Paul Cadach * /, channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Merged revisions 44022 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44022 | pcadach | 2006-09-30 01:06:55 +0600 (Сбт, 30 Сен 2006) | 3 lines Properly pass TON/PRESENTATION information - original H323Connection::SendSignalSetup() destroys Q.931 fields. ........ 2006-09-29 18:54 +0000 [r44013] Kevin P. Fleming * Makefile, codecs/Makefile, utils/Makefile, /, configure, include/asterisk/autoconfig.h.in, main/Makefile, codecs/gsm/Makefile, configure.ac, Makefile.moddir_rules, Makefile.rules, pbx/Makefile, channels/Makefile, main/db1-ast/Makefile: Merged revisions 43996-43997,44008,44011-44012 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43996 | kpfleming | 2006-09-29 11:47:05 -0500 (Fri, 29 Sep 2006) | 2 lines another cross-compile fix ........ r43997 | kpfleming | 2006-09-29 11:52:27 -0500 (Fri, 29 Sep 2006) | 2 lines support --without-curl in configure script ........ r44008 | kpfleming | 2006-09-29 13:25:49 -0500 (Fri, 29 Sep 2006) | 2 lines don't abuse CFLAGS and LDFLAGS for build of Asterisk components, because they are also then used for non-Asterisk components (like menuselect); use our own variables instead ........ r44011 | kpfleming | 2006-09-29 13:40:17 -0500 (Fri, 29 Sep 2006) | 2 lines missed one conversion to ASTCFLAGS ........ r44012 | kpfleming | 2006-09-29 13:49:07 -0500 (Fri, 29 Sep 2006) | 2 lines yet another place where we were not using the correct CFLAGS by default ........ 2006-09-29 18:35 +0000 [r44010] Paul Cadach * /, channels/h323/ast_h323.cxx, channels/chan_h323.c, channels/h323/ast_h323.h, channels/h323/chan_h323.h: Merged revisions 44009 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44009 | pcadach | 2006-09-30 00:30:34 +0600 (Сбт, 30 Сен 2006) | 1 line Pass TON/PRESENTATION information too ........ 2006-09-29 16:38 +0000 [r43979-43994] Kevin P. Fleming * Makefile, /: Merged revisions 43993 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43993 | kpfleming | 2006-09-29 11:38:27 -0500 (Fri, 29 Sep 2006) | 2 lines a couple more environment settings that can't leak into the menuselect build ........ * /, main/cli.c: Merged revisions 43978 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43978 | kpfleming | 2006-09-29 08:45:40 -0500 (Fri, 29 Sep 2006) | 2 lines proper fix for ast_group_t change ........ 2006-09-29 01:36 +0000 [r43954-43961] Joshua Colp * channels/chan_iax2.c: One must remember to unlock their list... thanks to Qwell for letting me into his box * main/pbx.c: Cache the application pointer so we don't have to needlessly search for it over and over. This should yield a suitable performance increase. 2006-09-28 22:43 +0000 [r43953] Kevin P. Fleming * /, include/asterisk/lock.h: Merged revisions 43952 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43952 | kpfleming | 2006-09-28 17:42:18 -0500 (Thu, 28 Sep 2006) | 2 lines eliminate compiler warning when DEBUG_CHANNEL_LOCKS is enabled and users of this header file don't also include channel.h ........ 2006-09-28 20:13 +0000 [r43945] Jason Parker * /, apps/app_queue.c: Merged revisions 43944 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43944 | qwell | 2006-09-28 13:11:22 -0700 (Thu, 28 Sep 2006) | 4 lines Fix incorrect argument order for member names, on persisted members. Issue 8047, patch by jmls. ........ 2006-09-28 18:09 +0000 [r43934] Joshua Colp * main/udptl.c, main/frame.c, /, channels/chan_sip.c, funcs/func_timeout.c, apps/app_festival.c, apps/app_alarmreceiver.c, channels/iax2-provision.c, res/res_musiconhold.c, apps/app_followme.c, channels/chan_iax2.c, res/res_monitor.c, apps/app_playback.c, include/asterisk/logger.h, res/res_smdi.c, channels/chan_misdn.c, channels/chan_skinny.c, apps/app_rpt.c, channels/chan_mgcp.c: Merged revisions 43933 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43933 | file | 2006-09-28 14:05:43 -0400 (Thu, 28 Sep 2006) | 2 lines Put in missing \ns on the end of ast_logs (issue #7936 reported by wojtekka) ........ 2006-09-28 17:38 +0000 [r43921] Kevin P. Fleming * /, apps/app_queue.c: Merged revisions 43919 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r43919 | kpfleming | 2006-09-28 12:35:42 -0500 (Thu, 28 Sep 2006) | 10 lines Merged revisions 43916 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r43916 | kpfleming | 2006-09-28 12:31:57 -0500 (Thu, 28 Sep 2006) | 2 lines fix buggy (and overly complex) loop used during reload of app_queue for static member list updating ........ ................ 2006-09-28 17:36 +0000 [r43920] Paul Cadach * /, channels/h323/ast_h323.cxx: Merged revisions 43918 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43918 | pcadach | 2006-09-28 23:34:19 +0600 (Чтв, 28 Сен 2006) | 1 line Extend call establishment timeout ........ 2006-09-28 17:32 +0000 [r43912-43917] Joshua Colp * /, channels/chan_iax2.c: Merged revisions 43915 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43915 | file | 2006-09-28 13:31:09 -0400 (Thu, 28 Sep 2006) | 2 lines Make sure the pvt exists before accessing it again as it may have gone away (issue #7562 reported by Seb7 and issue #7939 reported by sorg) ........ * /, main/cli.c: Merged revisions 43913 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43913 | file | 2006-09-28 13:14:07 -0400 (Thu, 28 Sep 2006) | 2 lines Warning be gone! ........ * channels/chan_sip.c: Add jitterbuffer information to sip list settings (issue #7945 reported by sergee) 2006-09-28 16:54 +0000 [r43902] BJ Weschke * /, apps/app_queue.c: Merged revisions 43899 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r43899 | bweschke | 2006-09-28 12:41:05 -0400 (Thu, 28 Sep 2006) | 11 lines Merged revisions 43897 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r43897 | bweschke | 2006-09-28 12:37:15 -0400 (Thu, 28 Sep 2006) | 3 lines app_queue is comparing the device names incorrectly while checking their statuses. It's internal list of interfaces includes the dial string, while the argument passed to this function does not have the dial string (/n for a local channel). This causes it to ignore the device state changes because it thinks it belongs to none of its members. (#8040 reported and patch by tim_ringenbach) ........ ................ 2006-09-28 16:43 +0000 [r43900] Kevin P. Fleming * /, main/cli.c: Merged revisions 43898 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r43898 | kpfleming | 2006-09-28 11:38:25 -0500 (Thu, 28 Sep 2006) | 10 lines Merged revisions 43895 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r43895 | kpfleming | 2006-09-28 11:32:44 -0500 (Thu, 28 Sep 2006) | 2 lines eliminate compiler warning introduced by recent changes ........ ................ 2006-09-28 16:19 +0000 [r43894] Joshua Colp * /, apps/app_meetme.c: Merged revisions 43893 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r43893 | file | 2006-09-28 12:17:36 -0400 (Thu, 28 Sep 2006) | 10 lines Merged revisions 43891 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r43891 | file | 2006-09-28 12:13:55 -0400 (Thu, 28 Sep 2006) | 2 lines Stop the stream after waitstream returns so that our formats get restored. (issue #7370 reported by kryptolus) ........ ................ 2006-09-28 16:01 +0000 [r43888] Paul Cadach * /, channels/h323/ast_h323.cxx: Merged revisions 43877 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43877 | pcadach | 2006-09-28 21:56:21 +0600 (Чтв, 28 Сен 2006) | 1 line Fix compiler warning ........ 2006-09-28 15:32 +0000 [r43865-43875] BJ Weschke * /, apps/app_queue.c: Merged revisions 43873 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r43873 | bweschke | 2006-09-28 11:29:21 -0400 (Thu, 28 Sep 2006) | 11 lines Merged revisions 43871 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r43871 | bweschke | 2006-09-28 11:18:05 -0400 (Thu, 28 Sep 2006) | 3 lines Fix race condion crash with get_member_status (#7864 - tim_ringenbach reported and patched) ........ ................ * /, apps/app_queue.c: Merged revisions 43864 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43864 | bweschke | 2006-09-28 09:24:10 -0400 (Thu, 28 Sep 2006) | 3 lines Autopause not working for queue members. (#8042 - jmls reported and patch) ........ 2006-09-28 13:02 +0000 [r43863] Paul Cadach * /, channels/h323/ast_h323.cxx, channels/h323/ast_h323.h, include/asterisk/compiler.h: Merged revisions 43861-43862 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43861 | pcadach | 2006-09-28 18:47:23 +0600 (Чтв, 28 Сен 2006) | 1 line Put attribute tag at correct place ........ r43862 | pcadach | 2006-09-28 18:58:22 +0600 (Чтв, 28 Сен 2006) | 1 line Force remote side to start media on outgoing PROGRESS message ........ 2006-09-28 11:32 +0000 [r43854-43855] Christian Richter * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged revisions 43852 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r43852 | crichter | 2006-09-28 13:03:05 +0200 (Do, 28 Sep 2006) | 9 lines Merged revisions 43764 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r43764 | crichter | 2006-09-27 14:51:03 +0200 (Mi, 27 Sep 2006) | 1 line fixed a bug which led to chan_list zombies, when the call could not be properly established in misdn_call. also removed the ACK_HDLC stuff which is not really needed. ........ ................ * channels/chan_misdn.c, /, channels/Makefile: Merged revisions 43775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43775 | crichter | 2006-09-27 18:24:51 +0200 (Mi, 27 Sep 2006) | 1 line removed the chan_misdn versioning, since asterisk has it's own ........ 2006-09-28 11:12 +0000 [r43845-43853] Paul Cadach * channels/h323/cisco-h225.h, /, channels/h323/ast_h323.cxx, main/file.c, channels/h323/cisco-h225.asn, channels/h323/cisco-h225.cxx: Merged revisions 43635,43843-43844,43846 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43635 | pcadach | 2006-09-26 03:26:12 +0600 (Втр, 26 Сен 2006) | 1 line Fix ASN1 description of non-standard Cisco extensions ........ r43843 | pcadach | 2006-09-28 12:01:37 +0600 (Чтв, 28 Сен 2006) | 1 line Don't treat unknown control frames as voice ........ r43844 | pcadach | 2006-09-28 12:02:45 +0600 (Чтв, 28 Сен 2006) | 1 line Don't warn on HOLD/UNHOLD control frames ........ r43846 | pcadach | 2006-09-28 16:51:21 +0600 (Чтв, 28 Сен 2006) | 1 line Do not open transmit channel until TCS is received ........ * channels/h323/ast_h323.cxx, channels/chan_h323.c, channels/h323/ast_h323.h, CHANGES, channels/h323/chan_h323.h, configs/h323.conf.sample: Handle HOLD/RETRIEVE notifications 2006-09-27 22:01 +0000 [r43827-43836] Joshua Colp * CHANGES: Update CHANGES to reflect libcap capability that was added. * configure, main/Makefile, configure.ac, makeopts.in, doc/security.txt, main/asterisk.c: Add ability to set high ToS bits as non-root on Linux using libcap (issue #7047 reported by maddison) * apps/app_voicemail.c: Finish up last commit * apps/app_voicemail.c: Do the directory walk dance instead of repeated stat calls as it seems to be faster (issue #7507 reported by Corydon76) 2006-09-27 20:27 +0000 [r43817] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 43816 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r43816 | tilghman | 2006-09-27 15:21:54 -0500 (Wed, 27 Sep 2006) | 10 lines Merged revisions 43815 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r43815 | tilghman | 2006-09-27 15:20:35 -0500 (Wed, 27 Sep 2006) | 2 lines Avoid inability to lock directory log message by creating the directory ahead of time. (Issue 7631) ........ ................ 2006-09-27 20:03 +0000 [r43804-43814] Jason Parker * main/pbx.c: Add BACKGROUNDSTATUS to Background() Issue #7835, original patch by bcnit - redone by me. * main/pbx.c, /, apps/app_playback.c: Merged revisions 43803 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43803 | qwell | 2006-09-27 12:44:02 -0700 (Wed, 27 Sep 2006) | 4 lines Fix an issue with PLAYBACKSTATUS not being set under certain circumstances. Fix a minor issue, to make it use the filenames that were parsed, instead of the entire argument string. Fix Background() to return -1 like Playback(), if no args are specified. ........ 2006-09-27 19:39 +0000 [r43792-43802] Joshua Colp * channels/chan_iax2.c: I *think* this is the last list in chan_iax2 * /, main/rtp.c: Merged revisions 43798 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43798 | file | 2006-09-27 15:10:59 -0400 (Wed, 27 Sep 2006) | 2 lines Compensate for out of order packets better if RFC2833 compensation is turned on. ........ * /, channels/chan_iax2.c: Merged revisions 43783 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43783 | file | 2006-09-27 13:00:31 -0400 (Wed, 27 Sep 2006) | 2 lines Get rid of two functions from a time now past (we THINK these are from pre-recursive lock time) that may be contributing to two open issues on the bug tracker (7562/7939) and that has the potential to just make bad things happen if the timing is right. ........ 2006-09-27 17:00 +0000 [r43785] Matthew Fredrickson * channels/chan_zap.c: Fix some little things 2006-09-27 16:57 +0000 [r43780] Russell Bryant * main/channel.c, /, res/res_features.c: Merged revisions 43779 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r43779 | russell | 2006-09-27 12:55:49 -0400 (Wed, 27 Sep 2006) | 50 lines Merged revisions 43778 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r43778 | russell | 2006-09-27 12:54:30 -0400 (Wed, 27 Sep 2006) | 42 lines Fix a problem that occurred if a user entered a digit that matched a bridge feature that was configured using multiple digits, and the digit that was pressed timed out in the feature digit timeout period. For example, if blind transfer is configured as '##', and a user presses just '#'. In this situation, the call would lock up and no longer pass any frames. (issue #7977 reported by festr, and issue #7982 reported by michaels and valuable input provided by mneuhauser and kuj. Fixed by me, with testing help and peer review from Joshua Colp). There are a couple of issues involved in this fix: 1) When ast_generic_bridge determines that there has been a timeout, it returned AST_BRIDGE_RETRY. Then, when ast_channel_bridge gets this result, it calls ast_generic_bridge over again with the same timestamp for the next event. This results in an endless loop of nothing until the call is terminated. This is resolved by simply changing ast_generic_bridge to return AST_BRIDGE_COMPLETE when it sees a timeout. 2) I also changed ast_channel_bridge such that if in the process of calculating the time until the next event, it knows a timeout has already occured, to immediately return AST_BRIDGE_COMPLETE instead of attempting to bridge the channels anyway. 3) In the process of testing the previous two changes, I ran into a problem in res_features where ast_channel_bridge would return because it determined that there was a timeout. However, ast_bridge_call in res_features would then determine by its own calculation that there was still 1 ms before the timeout really occurs. It would then proceed, and since the bridge broke out and did *not* return a frame, it interpreted this as the call was over and hung up the channels. The reason for this was because ast_bridge_call in res_features and ast_channel_bridge in channel.c were using different times for their calculations. channel.c uses the start_time on the bridge config, which is the time that the feature digit was recieved. However, res_features had another time, 'start', which was set right before calling ast_channel_bridge. 'start' will always be slightly after start_time in the bridge config, and sometimes enough to round up to one ms. This is fixed by making ast_bridge_call use the same time as ast_channel_bridge for the timeout calculation. ........ ................ 2006-09-27 16:49 +0000 [r43777] Matthew Fredrickson * channels/chan_zap.c: Add CLI block and unblock circuit commands for SS7. 2006-09-27 16:25 +0000 [r43776] Joshua Colp * /, channels/chan_sip.c: Merged revisions 43774 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43774 | file | 2006-09-27 12:23:12 -0400 (Wed, 27 Sep 2006) | 2 lines Make rfc2833compensate a global option. ........ 2006-09-27 12:32 +0000 [r43763] Paul Cadach * channels/chan_h323.c: Use ast_strdupa() instead of strdup(), thanks to sergee 2006-09-27 04:37 +0000 [r43754-43757] Russell Bryant * apps/app_voicemail.c: remote an unused buffer in mm_login() (issue #8038, selsky) In passing, I have cleaned up some formatting to better comply with our guidelines. I have also changed one place to use S_OR(), and a couple of places to use ast_strlen_zero() as appropriate. 2006-09-27 03:45 +0000 [r43740-43747] Steve Murphy * pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3, pbx/pbx_ael.c, pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test5, pbx/ael/ael-test/ael-test11/extensions.ael, pbx/ael/ael-test/ref.ael-test6, CHANGES, pbx/ael/ael-test/ael-test3/extensions.ael, pbx/ael/ael-test/ref.ael-test7, pbx/ael/ael-test/ael-test5/extensions.ael, pbx/ael/ael-test/ref.ael-vtest13: This commits the changes to AEL to use the gosub-with-args from Tilghman to perform macro calls. This results in substantially smaller stack footprint, which allows macro call depths in excess of 100,000 levels, rather than the limit of 7 calls deep, which the Macro app is subject to. * /, configs/extensions.ael.sample: Merged revisions 43739 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43739 | murf | 2006-09-26 20:32:47 -0600 (Tue, 26 Sep 2006) | 1 line This change to extensions.ael was to fix bug 8031; the install scripts are causing it to be copied to /etc/asterisk/extensions.ael, and because it is a fairly direct conversion of the original extensions.conf, the macro and context names clash with the existing extensions.conf. So, I put an ael- in front of all macros and contexts, and checked every goto and macro call. Also, this file compiles under aelparse. ........ 2006-09-27 01:39 +0000 [r43733] Joshua Colp * channels/chan_iax2.c: Clean up code and convert last two things (firmware/dialplan cache) to linked list macros. 2006-09-26 22:18 +0000 [r43721-43727] Jason Parker * apps/app_meetme.c: Fire a manager event when a meetme is started/stopped. Issue #7891, patch by suhler. * apps/app_queue.c: Add QueueSummary manager action. Gives "at a glance" information about a single queue, or all queues. Issue #8035, patch by rgollent, slightly modified (formatting) by me. 2006-09-26 21:01 +0000 [r43715] Russell Bryant * /, main/asterisk.c: Merged revisions 43710 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r43710 | russell | 2006-09-26 16:56:42 -0400 (Tue, 26 Sep 2006) | 17 lines (This was actually BE-65) Merged revisions 43708 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r43708 | russell | 2006-09-26 16:49:21 -0400 (Tue, 26 Sep 2006) | 7 lines Back in revision 4798, this message was changed from using ast_cli() to directly calling write(). During this change, checking if this was a remote console was removed. This caused this message about using "exit" or "quit" to exit an Asterisk console to come up in times where it did not make sense. This change restores the check to see if this is a remote console before printing the message. (fixes BE-4) ........ ................ 2006-09-26 20:51 +0000 [r43709] Joshua Colp * /, channels/chan_sip.c, include/asterisk/channel.h, .cleancount, main/cli.c: Merged revisions 43707 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r43707 | file | 2006-09-26 16:47:26 -0400 (Tue, 26 Sep 2006) | 10 lines Merged revisions 43705 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r43705 | file | 2006-09-26 16:38:06 -0400 (Tue, 26 Sep 2006) | 2 lines Use proper type to represent the group variable (issue #8025 reported by makoto) ........ ................ 2006-09-26 20:30 +0000 [r43702] Jason Parker * CHANGES: update CHANGES file to reflect codec support in chan_skinny 2006-09-26 20:26 +0000 [r43701] Russell Bryant * /, apps/app_voicemail.c: Merged revisions 43700 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r43700 | russell | 2006-09-26 16:24:39 -0400 (Tue, 26 Sep 2006) | 14 lines Merged revisions 43699 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r43699 | russell | 2006-09-26 16:23:15 -0400 (Tue, 26 Sep 2006) | 6 lines When parsing the sections of voicemail.conf that contain mailbox definitions, don't introduce a length limit on the definition by using a 256 byte temporary storage buffer. Instead, make the temporary buffer just as big as it needs to be to hold the entire mailbox definition. (fixes BE-68) ........ ................ 2006-09-26 20:20 +0000 [r43696-43698] Joshua Colp * channels/chan_local.c, /: Merged revisions 43697 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43697 | file | 2006-09-26 16:19:33 -0400 (Tue, 26 Sep 2006) | 2 lines Strip options off the argument passed for devicestate in chan_local. (issue #8034 reported by pcardozo) ........ * main/channel.c, /, main/slinfactory.c, apps/app_chanspy.c: Merged revisions 43695 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43695 | file | 2006-09-26 16:09:41 -0400 (Tue, 26 Sep 2006) | 2 lines Slight overhaul of the whisper support. 1. We need to duplicate the frame from ast_translate 2. We need to ensure we always have signed linear coming in for signed linear combining. 3. We need to ensure we are always feeding signed linear out. 4. Properly store and restore write format when beeping on the channel we are whispering on. 5. Properly discontinue the stream on the channel for the beep. (issue #8019 reported by timkelly1980) ........ 2006-09-26 19:37 +0000 [r43677-43687] Kevin P. Fleming * CHANGES: start a CHANGES file for trunk... no need to force people to have to review commit logs after branching * /, sounds/Makefile: Merged revisions 43676 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43676 | kpfleming | 2006-09-26 13:34:27 -0500 (Tue, 26 Sep 2006) | 2 lines update to use 1.4.3 core sounds, with corrected beep/beeperr/tt-monkeys files ........ 2006-09-26 18:10 +0000 [r43675] Jason Parker * main/frame.c, /, doc/rtp-packetization.txt: Merged revisions 43674 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43674 | qwell | 2006-09-26 11:08:51 -0700 (Tue, 26 Sep 2006) | 4 lines Issue #8015, patch by Dan Austin. Maximum values were incorrect, which is why this is being put in 1.4 ........ 2006-09-26 17:25 +0000 [r43667] Tilghman Lesher * apps/app_stack.c: Gosub arguments (Issue 7780) 2006-09-26 17:09 +0000 [r43666] Jason Parker * main/logger.c, configs/logger.conf.sample: Add optional queue_log_name config option for logger.conf, to change the name of the queue_log file. Issue #7363, patch by Steve Davies, slightly modified by me. 2006-09-26 16:56 +0000 [r43658-43659] Tilghman Lesher * apps/app_voicemail.c: MailboxExists should be a dialplan function, not an application (Issue 7503) * res/res_limit.c: These three are not defined on all platforms that we support 2006-09-26 15:35 +0000 [r43651] Jason Parker * /, channels/chan_skinny.c: Merged revisions 43650 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43650 | qwell | 2006-09-26 08:33:47 -0700 (Tue, 26 Sep 2006) | 11 lines Add proper codec support to chan_skinny. Works with at least ulaw, alaw, and g729a. This is technically a "new feature", but there are justifications for it. I found a bug with the recent rtp packetization changes, which caused the media setup to fail under certain circumstances, particularly when using allow=all, or having no allow= statements (globally or on the device). I could have either removed the rtp packetization features, or I could add proper codec support (which, without, I think most people would consider to be a bug anyways). ........ 2006-09-25 22:09 +0000 [r43641-43643] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 43642 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43642 | tilghman | 2006-09-25 17:07:44 -0500 (Mon, 25 Sep 2006) | 2 lines Should have moved these lines up in the merge, instead of removing them ........ * /, apps/app_voicemail.c: Merged revisions 43640 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r43640 | tilghman | 2006-09-25 17:04:47 -0500 (Mon, 25 Sep 2006) | 12 lines Merged revisions 43634 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r43634 | tilghman | 2006-09-25 16:14:41 -0500 (Mon, 25 Sep 2006) | 4 lines Two bugs when forwarding voicemail (Issue 7824): 1) delete=yes was ignored 2) maxmessages was ignored ........ ................ 2006-09-25 20:30 +0000 [r43627] Paul Cadach * /: Block revision 43626 from 1.4 tree - already here 2006-09-25 15:24 +0000 [r43617] Jason Parker * /, sounds/Makefile: Merged revisions 43616 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43616 | qwell | 2006-09-25 08:23:31 -0700 (Mon, 25 Sep 2006) | 4 lines One more fix for sounds installation - this time for portability. Reported to asterisk-dev mailing list. ........ 2006-09-25 14:49 +0000 [r43604] Steve Murphy * formats/format_ogg_vorbis.c: This tiny fix prevents asterisk from crashing if trying to play an OGG moh file. 2006-09-25 09:03 +0000 [r43571-43597] Paul Cadach * channels/h323/caps_h323.cxx, channels/h323/ast_h323.cxx, channels/chan_h323.c, channels/h323/ast_h323.h, channels/h323/chan_h323.h, configs/h323.conf.sample: Support for negotiation and receiption of Cisco's RTP DTMF * channels/h323/ast_h323.cxx: Disable fastStart if requested by remote side * /: Block revision 43582 * channels/chan_h323.c, configs/h323.conf.sample: Specify RFC2833 payload on dtmfmode option rather than dtmfcodec option (deprecated) * channels/h323/ast_h323.cxx, channels/chan_h323.c: DTMF mode is bitmask, not valued field * channels/h323/caps_h323.cxx, channels/h323/caps_h323.h: Define Cisco RTP capability * channels/h323/caps_h323.cxx: Specify non-standard data independedly on OpenH323's codec name (it can be easily changed) * channels/chan_h323.c, channels/h323/chan_h323.h: Define DTMF payload types 2006-09-24 15:01 +0000 [r43554-43565] Russell Bryant * /, channels/iax2-provision.c: Merged revisions 43564 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43564 | russell | 2006-09-24 10:58:10 -0400 (Sun, 24 Sep 2006) | 5 lines Fix a CLI command registration issue where an erroneous message claiming that "iax2 show provisioning" was already registered. This was because this command was registering itself as both the command, as well as the command it is deprecating. (issue #8022, reported by bjweeks, fixed by myself) ........ * /, channels/chan_iax2.c: Merged revisions 43553 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r43553 | russell | 2006-09-24 09:53:35 -0400 (Sun, 24 Sep 2006) | 12 lines Merged revisions 43552 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r43552 | russell | 2006-09-24 09:50:30 -0400 (Sun, 24 Sep 2006) | 4 lines Check to see if the channel that is activating the IAXPEER function is actually an IAX2 channel before proceeding to process it to avoid crashing. (issue #8017, reported by admott, fixed by myself) ........ ................ 2006-09-24 12:15 +0000 [r43539-43546] Paul Cadach * main/rtp.c: Small Cisco's RTP DTMF update * channels/chan_h323.c: Avoid possible deadlock on channel destruction * main/rtp.c: Correct behavior on Cisco's DTMF 2006-09-22 23:46 +0000 [r43525-43526] Kevin P. Fleming * /: file forgot one :-) * Makefile, /: Merged revisions 43524 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43524 | kpfleming | 2006-09-22 18:44:47 -0500 (Fri, 22 Sep 2006) | 2 lines don't output the 'build complete' message when the target being run is already going to do an installation ........ 2006-09-22 23:34 +0000 [r43522] Joshua Colp * /: You see nothing... 2006-09-22 22:13 +0000 [r43519] Jason Parker * /, channels/chan_skinny.c: Merged revisions 43518 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43518 | qwell | 2006-09-22 15:12:12 -0700 (Fri, 22 Sep 2006) | 4 lines Allow chan_skinny.so to be unloaded properly. Remove reload support, since it doesn't actually...work. ........ 2006-09-22 21:34 +0000 [r43506-43507] Steve Murphy * pbx/pbx_ael.c: This commits a change to return MODULE_LOAD_FAILURE on error, and SUCCESS (instead of 0) when all goes well for bug 8004 * pbx/pbx_ael.c: As per bug 8004, we now return AST_MODULE_LOAD_DECLINE when we can't read extensions.ael 2006-09-22 20:33 +0000 [r43495-43500] Paul Cadach * channels/chan_h323.c: Move from h.323 to h323 command prefix * channels/chan_h323.c: Fix compilation warnings * channels/h323/compat_h323.h: Use own factory for our OpalMediaFormats too * channels/h323/caps_h323.cxx, channels/h323/compat_h323.h: Fix our capability's factory 2006-09-22 17:26 +0000 [r43493] Jason Parker * /, main/cli.c: Merged revisions 43492 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43492 | qwell | 2006-09-22 10:25:05 -0700 (Fri, 22 Sep 2006) | 2 lines Make sure we explicitly set the CLI command to not be deprecated, if it isn't. ........ 2006-09-22 16:43 +0000 [r43488-43490] Kevin P. Fleming * /, sounds/Makefile: Merged revisions 43489 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43489 | kpfleming | 2006-09-22 11:42:46 -0500 (Fri, 22 Sep 2006) | 2 lines use rebuilt extra sounds ........ * main/channel.c, /: Merged revisions 43486 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43486 | kpfleming | 2006-09-22 10:51:13 -0500 (Fri, 22 Sep 2006) | 2 lines all the Linux systems I have don't use '__m_count' for this field, so I don't know where this came from... ........ 2006-09-22 15:50 +0000 [r43483-43485] Russell Bryant * channels/chan_misdn.c, /: Merged revisions 43482 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43482 | russell | 2006-09-22 11:42:44 -0400 (Fri, 22 Sep 2006) | 3 lines return AST_MODULE_LOAD_DECLIDE if mISDN could not be configured (issue #8006, Mithraen) ........ 2006-09-22 14:58 +0000 [r43479-43480] Luigi Rizzo * include/asterisk/threadstorage.h: compatibility fix: use "attribute_XXX" instead of *__attribute__ ((XXX)) so we can handle compiler/os dependencies in our compiler.h * channels/chan_sip.c: style fix: move variable declaration at the beginning of the block. 2006-09-22 14:04 +0000 [r43478] Russell Bryant * main/frame.c, /: Merged revisions 43477 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43477 | russell | 2006-09-22 10:02:58 -0400 (Fri, 22 Sep 2006) | 3 lines Suppress a compiler warning about the use of a potentially uninitialized variable. It couldn't actually happen, though. ........ 2006-09-22 04:54 +0000 [r43472] Paul Cadach * channels/h323/caps_h323.cxx: Add missing include 2006-09-22 03:09 +0000 [r43470] Jason Parker * /, channels/chan_skinny.c: Merged revisions 43469 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43469 | qwell | 2006-09-21 20:01:16 -0700 (Thu, 21 Sep 2006) | 4 lines First shot at unload_module in chan_skinny.. More to come. ........ 2006-09-21 23:55 +0000 [r43467] Matt O'Gorman * /, include/asterisk/jabber.h, channels/chan_gtalk.c, res/res_jabber.c: Merged revisions 43466 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43466 | mogorman | 2006-09-21 18:50:56 -0500 (Thu, 21 Sep 2006) | 2 lines updates for better compontent support ........ 2006-09-21 23:29 +0000 [r43463-43465] Tilghman Lesher * /, res/res_odbc.c, configs/res_odbc.conf.sample: Merged revisions 43464 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43464 | tilghman | 2006-09-21 18:24:41 -0500 (Thu, 21 Sep 2006) | 2 lines Twould help if we actually documented how the new features in res_odbc actually work. (Oops) ........ * res/res_limit.c (added): Set process limits without restarting Asterisk 2006-09-21 22:53 +0000 [r43461] Joshua Colp * channels/chan_sip.c, channels/chan_iax2.c: Oh look more changes, but these are my own! (Clean up module load functions) 2006-09-21 22:44 +0000 [r43460] Jason Parker * channels/chan_zap.c: Suppress compiler warnings 2006-09-21 22:32 +0000 [r43459] Joshua Colp * channels/chan_alsa.c: Clean up chan_alsa load module function (issue #8000 reported by Mithraen) 2006-09-21 22:23 +0000 [r43458] Tilghman Lesher * include/asterisk/acl.h, doc/ip-tos.txt, channels/chan_sip.c, doc/mp3.txt, doc/ael.txt, doc/channelvariables.txt, main/acl.c: And some deprecated APIs and modifications to documentation 2006-09-21 22:23 +0000 [r43455-43457] Joshua Colp * /, channels/chan_oss.c: Merged revisions 43456 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43456 | file | 2006-09-21 18:21:40 -0400 (Thu, 21 Sep 2006) | 2 lines Some more clean up in the load function for chan_oss (issue #8002 reported by Mithraen with minor mods by moi) ........ * /, channels/chan_mgcp.c: Merged revisions 43454 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43454 | file | 2006-09-21 18:12:09 -0400 (Thu, 21 Sep 2006) | 2 lines Clean up chan_mgcp's module load function (issue #8001 reported by Mithraen with mods by moi) ........ 2006-09-21 21:59 +0000 [r43452] Tilghman Lesher * doc/ip-tos.txt, channels/chan_local.c, channels/chan_sip.c, res/res_features.c, channels/chan_agent.c, res/res_convert.c, res/res_crypto.c, res/res_musiconhold.c, channels/chan_iax2.c, channels/chan_oss.c, channels/chan_skinny.c, channels/chan_features.c, res/res_agi.c, channels/chan_h323.c, channels/chan_alsa.c, apps/app_settransfercapability.c (removed), res/res_indications.c, pbx/pbx_config.c, res/res_odbc.c, channels/chan_mgcp.c: Lots more removal of deprecated things 2006-09-21 21:22 +0000 [r43451] Kevin P. Fleming * /, main/Makefile, build_tools/strip_nonapi (added): Merged revisions 43450 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43450 | kpfleming | 2006-09-21 16:21:29 -0500 (Thu, 21 Sep 2006) | 2 lines add another attempt to strip non-API symbols from the final binary... script will need to be extended to work on non-Linux systems ........ 2006-09-21 21:17 +0000 [r43442-43449] Tilghman Lesher * main/udptl.c, main/pbx.c, main/frame.c, main/translate.c, apps/app_queue.c, main/config.c, main/rtp.c, apps/app_setcdruserfield.c (removed), main/cli.c, main/channel.c, main/manager.c, main/file.c, main/http.c, main/logger.c, main/astmm.c, main/image.c, main/asterisk.c: Remove deprecated CLI apps from the core * /, apps/app_url.c: Merged revisions 43445 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43445 | tilghman | 2006-09-21 15:22:43 -0500 (Thu, 21 Sep 2006) | 2 lines Fix documentation to reflect how Url() really works ........ * apps/app_setcallerid.c, apps/app_voicemail.c: More removal of deprecated stuff * main/pbx.c, main/manager.c, UPGRADE.txt: Remove 1.4 changes from UPGRADE.txt, remove deprecated callerid field, remove deprecated SetGlobalVar app * /, apps/app_rpt.c: Merged revisions 43441 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43441 | tilghman | 2006-09-21 14:43:32 -0500 (Thu, 21 Sep 2006) | 2 lines Oops, missed the merge breakage ........ 2006-09-21 19:42 +0000 [r43440] Kevin P. Fleming * makeopts.in: fix this so chan_zap links properly again 2006-09-21 19:35 +0000 [r43439] Tilghman Lesher * funcs/func_language.c (removed), funcs/func_moh.c (removed), apps/app_lookupcidname.c (removed), funcs/func_md5.c, apps/app_hasnewvoicemail.c (removed), funcs/func_blacklist.c (added), apps/app_random.c (removed), funcs/func_vmcount.c (added), res/res_realtime.c (added), apps/app_lookupblacklist.c (removed), apps/app_realtime.c (removed), apps/app_queue.c: Remove deprecated apps and funcs 2006-09-21 19:27 +0000 [r43437] Joshua Colp * apps/app_dial.c, main/channel.c, /, channels/chan_sip.c, include/asterisk/rtp.h, include/asterisk/channel.h, main/rtp.c: SS7 marked the start of an open season for trunk again but here's something minor - abstract early bridging into the technology so that we don't always assume they use RTP and try it. 2006-09-21 19:22 +0000 [r43436] Kevin P. Fleming * configure: regenerated at PCadach's request 2006-09-21 19:18 +0000 [r43429-43434] Paul Cadach * acinclude.m4: Check for 64-bit OpenH323/PWLib versions too, thanks to Mithraen (please, re-build configure script) * channels/h323/caps_h323.cxx: Declare our own media formats to not rely on OpenH323 configuration * channels/h323/caps_h323.cxx, channels/h323/ast_h323.cxx, channels/chan_h323.c, channels/h323/caps_h323.h: Introduce Cisco G.726-32 capability (g726aal2 form) 2006-09-21 18:42 +0000 [r43427-43428] Matthew Fredrickson * configure: Update configure * channels/chan_zap.c, build_tools/menuselect-deps.in, include/asterisk/autoconfig.h.in, configure.ac, makeopts.in, configs/zapata.conf.sample: Merge in SS7 changes.... need to still cleanup zapata.conf 2006-09-21 17:06 +0000 [r43411-43423] Tilghman Lesher * /, apps/app_rpt.c: Merged revisions 43422 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r43422 | tilghman | 2006-09-21 12:04:40 -0500 (Thu, 21 Sep 2006) | 10 lines Merged revisions 43420 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r43420 | tilghman | 2006-09-21 12:01:48 -0500 (Thu, 21 Sep 2006) | 2 lines Whitespace change... really just an excuse to test repotools ........ ................ * /: Last merge should not have brought in the 1.2 props * /, configure, configure.ac, cdr/cdr_tds.c: Merged revisions 43410 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r43410 | tilghman | 2006-09-21 11:31:59 -0500 (Thu, 21 Sep 2006) | 10 lines Merged revisions 43409 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r43409 | tilghman | 2006-09-21 11:18:19 -0500 (Thu, 21 Sep 2006) | 2 lines TDS 0.64 updates ........ ................ 2006-09-21 16:09 +0000 [r43403-43406] Kevin P. Fleming * /, main/Makefile: Merged revisions 43405 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43405 | kpfleming | 2006-09-21 11:08:03 -0500 (Thu, 21 Sep 2006) | 2 lines remove this change... it requires binutils 2.17 ........ * /: remove extraneous property 2006-09-20 23:20 +0000 [r43397] Jason Parker * /, build_tools/make_version: Merged revisions 43396 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43396 | qwell | 2006-09-20 16:19:25 -0700 (Wed, 20 Sep 2006) | 2 lines fix minor typo in the way version is handled ........ 2006-09-20 23:02 +0000 [r43393] Kevin P. Fleming * /: this has been manually merged 2006-09-20 Kevin P. Fleming * Asterisk 1.4.0-beta1 released.