diff options
| author | lmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b> | 2010-06-01 16:15:28 +0000 |
|---|---|---|
| committer | lmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b> | 2010-06-01 16:15:28 +0000 |
| commit | 3a7ec8a8019b60b90c4e5f9e5ca3cdf84b74d325 (patch) | |
| tree | 69ef53fdf7f0e43c952cb8c613b1273b1f16303e | |
| parent | d14b1214fc1f2054f4d7216728258b2addd3cd98 (diff) | |
| parent | 5d439abcfe593f7ea0bb9bc62b3e7a28909f5cf2 (diff) | |
Creating tag for the release of asterisk-1.6.2.9-rc1
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.2.9-rc1@266655 f38db490-d61c-443f-a65b-d21fe96a405b
| -rw-r--r-- | .lastclean | 1 | ||||
| -rw-r--r-- | .version | 1 | ||||
| -rw-r--r-- | ChangeLog | 25022 |
3 files changed, 0 insertions, 25024 deletions
diff --git a/.lastclean b/.lastclean deleted file mode 100644 index 7facc8993..000000000 --- a/.lastclean +++ /dev/null @@ -1 +0,0 @@ -36 diff --git a/.version b/.version deleted file mode 100644 index 686fa1eb0..000000000 --- a/.version +++ /dev/null @@ -1 +0,0 @@ -1.6.2.9-rc1 diff --git a/ChangeLog b/ChangeLog deleted file mode 100644 index 67278177d..000000000 --- a/ChangeLog +++ /dev/null @@ -1,25022 +0,0 @@ -2010-06-01 Leif Madsen <lmadsen@digium.com> - - * Asterisk 1.6.2.9-rc1 Released. - -2010-06-01 15:20 +0000 [r266598] Tilghman Lesher <tlesher@digium.com> - - * main/asterisk.c, /: Merged revisions 266592 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r266592 | tilghman | 2010-06-01 10:18:59 -0500 (Tue, 01 Jun 2010) - | 18 lines Merged revisions 266585 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r266585 | tilghman | 2010-06-01 10:17:46 -0500 (Tue, 01 Jun 2010) - | 11 lines Prevent CLI prompt from distorting output of lines - shorter than the prompt. Uses the VT100 method of clearing the - line from the cursor position to the end of the line: Esc-0K - (closes issue #17160) Reported by: coolmig Patches: - 20100531__issue17160.diff.txt uploaded by tilghman (license 14) - Tested by: coolmig ........ ................ - -2010-05-31 16:07 +0000 [r266570] Paul Belanger <paul.belanger@polybeacon.com> - - * res/res_agi.c: Fix typo in documentation (closes issue #17395) - Reported by: pabelanger Patches: res_agi.c.patch uploaded by - pabelanger (license 224) - -2010-05-30 04:45 +0000 [r266439] Tilghman Lesher <tlesher@digium.com> - - * contrib/init.d/rc.debian.asterisk, /: Merged revisions 266438 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r266438 | tilghman | 2010-05-29 23:44:28 -0500 - (Sat, 29 May 2010) | 9 lines Merged revisions 266437 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r266437 | tilghman | 2010-05-29 23:43:28 -0500 (Sat, 29 - May 2010) | 2 lines Reverting patch and reopening issue #16784, - as patch breaks color display. ........ ................ - -2010-05-28 20:55 +0000 [r266338] Tilghman Lesher <tlesher@digium.com> - - * main/asterisk.c, /: Merged revisions 266337 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r266337 | - tilghman | 2010-05-28 15:53:04 -0500 (Fri, 28 May 2010) | 1 line - Only report swap on platforms which can examine those statistics - ........ - -2010-05-28 17:57 +0000 [r266293] David Vossel <dvossel@digium.com> - - * /, channels/chan_sip.c: Merged revisions 266292 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r266292 | - dvossel | 2010-05-28 12:55:38 -0500 (Fri, 28 May 2010) | 9 lines - fixes crash when creation of UDPTL fails (closes issue #17264) - Reported by: falves11 Patches: issue_17264_reviewboard_fix.diff - uploaded by dvossel (license 671) - issue_17264_1.6.2_reviewboard_fix.diff uploaded by dvossel - (license 671) Tested by: falves11 ........ - -2010-05-26 21:19 +0000 [r266154] Tilghman Lesher <tlesher@digium.com> - - * utils/extconf.c, main/asterisk.c, /, main/logger.c: Merged - revisions 266146 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r266146 | tilghman | 2010-05-26 16:17:46 -0500 (Wed, 26 May 2010) - | 21 lines Merged revisions 266142 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r266142 | tilghman | 2010-05-26 16:11:44 -0500 (Wed, 26 May 2010) - | 14 lines Use sigaction for signals which should persist past - the initial trigger, not signal. If you call signal() in a - Solaris signal handler, instead of just resetting the signal - handler, it causes the signal to refire, because the signal is - not marked as handled prior to the signal handler being called. - This effectively causes Solaris to immediately exceed the - threadstack in recursive signal handlers and crash. (closes issue - #17000) Reported by: rmcgilvr Patches: - 20100526__issue17000.diff.txt uploaded by tilghman (license 14) - Tested by: rmcgilvr ........ ................ - -2010-05-26 18:37 +0000 [r266007] David Vossel <dvossel@digium.com> - - * /, channels/chan_sip.c: Merged revisions 266006 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r266006 | - dvossel | 2010-05-26 13:32:51 -0500 (Wed, 26 May 2010) | 8 lines - fixes failed SIP Directed pickup resulting in dead channel - (closes issue #17339) Reported by: one47 Patches: - sip_magic_pickup2 uploaded by one47 (license 23) Tested by: - one47, dvossel ........ - -2010-05-26 16:31 +0000 [r265895-265959] Tilghman Lesher <tlesher@digium.com> - - * res/res_config_pgsql.c, /: Merged revisions 265923 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r265923 | tilghman | 2010-05-26 11:23:28 -0500 - (Wed, 26 May 2010) | 14 lines Merged revisions 265910 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r265910 | tilghman | 2010-05-26 11:21:00 -0500 (Wed, 26 May 2010) - | 7 lines Not finding rows in the DB does not rise to the level - of a warning. (closes issue #17062) Reported by: drookie Patches: - 20100525__issue17062.diff.txt uploaded by tilghman (license 14) - ........ ................ - - * configs/res_pgsql.conf.sample, res/res_config_pgsql.c, /: Merged - revisions 265894 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r265894 | - tilghman | 2010-05-26 11:14:48 -0500 (Wed, 26 May 2010) | 8 lines - Construct socket name, according to the Postgres docs, and - document as such. (closes issue #17392) Reported by: dps Patches: - 20100525__issue17392.diff.txt uploaded by tilghman (license 14) - Tested by: dps ........ - -2010-05-26 15:52 +0000 [r265890] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Recorded merge of revisions 265842 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r265842 | mmichelson | 2010-05-26 09:41:55 -0500 (Wed, - 26 May 2010) | 9 lines Re-enable "always" option for videosupport - option in sip.conf. (closes issue #17016) Reported by: twilson - Patches: 17016.patch uploaded by mmichelson (license 60) Tested - by: devmod ........ - -2010-05-26 00:33 +0000 [r265748] Tilghman Lesher <tlesher@digium.com> - - * /, configure, include/asterisk/autoconfig.h.in, configure.ac, - pbx/pbx_lua.c: Merged revisions 265747 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r265747 | - tilghman | 2010-05-25 19:29:40 -0500 (Tue, 25 May 2010) | 8 lines - Use configure to determine the prefixes and include directories - properly. This ensures cross-platform compatibility, even among - Linux distributions, which don't always put headers in the same - place. (closes issue #17391) Reported by: loloski ........ - -2010-05-25 21:05 +0000 [r265699] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 265698 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r265698 | - mmichelson | 2010-05-25 15:59:04 -0500 (Tue, 25 May 2010) | 12 - lines Properly use peer's outboundproxy for outbound REGISTERs. - The logic used in transmit_register to get the outboundproxy for - a peer was flawed since this value would be overridden shortly - afterwards when create_addr was called. In addition, this also - fixes some logic used when parsing users.conf so that the peer - name is placed in the internally-generated register string so - that an outboundproxy set in the Asterisk GUI will be used for - outbound REGISTERs. ........ - -2010-05-25 17:15 +0000 [r265615] David Vossel <dvossel@digium.com> - - * channels/chan_dahdi.c: fixes build issue with zaptel (closes - issue 0017394) Reported by: aragon Patches: half_buffer_fix.diff - uploaded by dvossel (license 671) Tested by: aragon - -2010-05-25 17:06 +0000 [r265612] Matthew Nicholson <mnicholson@digium.com> - - * apps/app_queue.c, /: Merged revisions 265611 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r265611 | mnicholson | 2010-05-25 12:00:11 -0500 (Tue, 25 May - 2010) | 15 lines Merged revisions 265610 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r265610 | mnicholson | 2010-05-25 11:48:19 -0500 (Tue, 25 May - 2010) | 8 lines Don't mark the cdr records of unanswered queue - calls with "NOANSWER". This restores the behavior prior to - r258670. (closes issue #17334) Reported by: jvandal Patches: - queue-cdr-fixes1.diff uploaded by mnicholson (license 96) Tested - by: aragon, jvandal ........ ................ - -2010-05-24 23:52 +0000 [r265521] Terry Wilson <twilson@digium.com> - - * include/asterisk/options.h, main/asterisk.c, Makefile, - doc/manager_1_1.txt, doc/tex/manager.tex, main/manager.c: Merged - revisions 265320,265467 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r265320 | - twilson | 2010-05-24 14:06:40 -0500 (Mon, 24 May 2010) | 14 lines - Add the FullyBooted AMI event It is possible to connect to the - manager interface before all Asterisk modules are loaded. To - ensure that an application does not send AMI actions that might - require a module that has not yet loaded, the application can - listen for the FullyBooted manager event. It will be sent upon - connection if all modules have been loaded, or as soon as loading - is complete. The event: Event: FullyBooted Privilege: system,all - Status: Fully Booted Review: - https://reviewboard.asterisk.org/r/639/ ........ r265467 | - twilson | 2010-05-24 17:21:58 -0500 (Mon, 24 May 2010) | 1 line - Merge the rest of the FullyBooted patch ........ - -2010-05-24 22:07 +0000 [r265450-265452] Mark Michelson <mmichelson@digium.com> - - * /, channels/h323/ast_h323.cxx: Merged revisions 265451 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r265451 | mmichelson | 2010-05-24 17:05:15 -0500 (Mon, - 24 May 2010) | 8 lines Print openh323 log to the Asterisk - console. (closes issue #17109) Reported by: under Patches: - logstream.diff uploaded by under (license 914) ........ - - * /, channels/chan_sip.c: Merged revisions 265449 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r265449 | - mmichelson | 2010-05-24 16:44:30 -0500 (Mon, 24 May 2010) | 11 - lines Allow type=user SIP endpoints to be loaded properly from - realtime. (closes issue #16021) Reported by: Guggemand Patches: - realtime-type-fix.patch uploaded by Guggemand (license 897) - (altered by me slightly to avoid ref leaks) Tested by: Guggemand - ........ - -2010-05-24 19:30 +0000 [r265364] David Vossel <dvossel@digium.com> - - * main/channel.c, /: Merged revisions 265273 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r265273 | - dvossel | 2010-05-24 11:10:09 -0500 (Mon, 24 May 2010) | 2 lines - fixes segfault when using generic plc ........ - -2010-05-24 18:30 +0000 [r265318] Tilghman Lesher <tlesher@digium.com> - - * main/asterisk.c, /: Merged revisions 265316 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r265316 | - tilghman | 2010-05-24 13:19:08 -0500 (Mon, 24 May 2010) | 7 lines - On systems with a LOT of RAM, a signed integer sometimes printed - negative. (closes issue #16837) Reported by: jlpedrosa Patches: - 20100504__issue16837.diff.txt uploaded by tilghman (license 14) - ........ - -2010-05-21 21:57 +0000 [r264998-265172] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Fix memory hogging behavior of app_queue. From - reviewboard: This review request is for the patch on issue 17081. - A user reported that he saw increasing numbers of allocations - stemming from app_queue.c when he would run the "queue show" CLI - command. The user reported that he was using approximately 40 - realtime queues and as he ran the CLI command more and more, the - memory usage would shoot up. As it turns out, there was a memory - leak and a separate usage of memory that, while not really a - leak, was very irresponsible. Both memory problems can be - attributed to the function init_queue(). When the "queue show" - command is run, all realtime queues have the init_queue() - function called on the in-memory queue. The idea is to place the - queue in its default state and then overwrite options specified - in the realtime backend as we read them. The first problem, the - memory leak, had to do with the fact that the string field for - the name of the first periodic announcement file was being - re-created every time init_queue was called. This patch corrects - the behavior by only calling ast_str_create if the memory has not - already been allocated. The other problem is a bit more - complicated. The majority of the strings in the call_queue - structure were changed to use the ast_string_fields API for 1.6.0 - and beyond. init_queue resets all string fields on the queue to - their default values. Then, later in the realtime queue loading - process, these string fields are set to their configured values. - For those unfamiliar with string fields, frequent resizing of a - string like this is not what the string fields API is designed - for. The result of this constant resizing is that as the queue - gets loaded, eventually space for the string runs out and so a - new memory pool, at twice the size of the previously allocated - one, is created for the string fields. The reporter of issue - 17081 wrote a script that ran the "queue show" CLI command 2100 - times. By the end, each of his 40 queues was taking about a - megabyte of memory apiece just for their string fields. My fix - for this problem is to revert the call_queue structure from using - string fields. In my patch here, I have moved the queue back to - using fixed-sized buffers. I ran the script provided by the - reporter of 17081 and determined that I no longer saw the - steadily-increasing memory usage that I had seen before applying - the patch. (closes issue #17081) Reported by: wliegel Patches: - 17081v2.patch uploaded by mmichelson (license 60) Tested by: - wliegel, mmichelson Review: - https://reviewboard.asterisk.org/r/651/ - - * apps/app_queue.c, include/asterisk/file.h, /: Merged revisions - 265090 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r265090 | mmichelson | 2010-05-21 16:08:51 -0500 (Fri, 21 May - 2010) | 15 lines Merged revisions 265089 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r265089 | mmichelson | 2010-05-21 15:59:14 -0500 (Fri, 21 May - 2010) | 8 lines Don't hang up on a queue caller if the file we - attempt to play does not exist. This also fixes a documentation - mistake in file.h that made my original attempt to correct this - problem not work correctly. (closes issue #17061) Reported by: - RoadKill ........ ................ - - * /, channels/chan_sip.c: Merged revisions 265087 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r265087 | - mmichelson | 2010-05-21 15:38:14 -0500 (Fri, 21 May 2010) | 7 - lines Be sure to set the sin_family on the proxy when allocating. - (closes issue #17157) Reported by: stuarth ........ - - * /, include/asterisk/channel.h: Merged revisions 265000 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r265000 | mmichelson | 2010-05-21 11:54:21 -0500 - (Fri, 21 May 2010) | 9 lines Merged revisions 264999 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r264999 | mmichelson | 2010-05-21 11:53:53 -0500 (Fri, - 21 May 2010) | 3 lines Fix grammatical error in comment. ........ - ................ - - * main/channel.c, main/autoservice.c, /, - include/asterisk/channel.h: Merged revisions 264997 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r264997 | mmichelson | 2010-05-21 11:44:27 -0500 - (Fri, 21 May 2010) | 38 lines Merged revisions 264996 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r264996 | mmichelson | 2010-05-21 11:28:34 -0500 (Fri, 21 May - 2010) | 32 lines Allow ast_safe_sleep to defer specific frames - until after the sleep has concluded. From reviewboard Background: - A Digium customer discovered a somewhat odd bug. The setup is - that parties A and B are bridged, and party A places party B on - hold. While party B is listening to hold music, he mashes a bunch - of DTMF. Party A takes party B off hold while this is happening, - but party B continues to hear hold music. I could reproduce this - about 1 in 5 times. The issue: When DTMF features are enabled and - a user presses keys, the channel that the DTMF is streamed to is - placed in an ast_safe_sleep for 100 ms, the duration of the - emulated tone. If an AST_CONTROL_UNHOLD frame is read from the - channel during the sleep, the frame is dropped. Thus the unhold - indication is never made to the channel that was originally - placed on hold. The fix: Originally, I discussed with Kevin - possible ways of fixing the specific problem reported. However, - we determined that the same type of problem could happen in other - situations where ast_safe_sleep() is used. Using autoservice as a - model, I modified ast_safe_sleep_conditional() to defer specific - frame types so they can be re-queued once the sleep has finished. - I made a common function for determining if a frame should be - deferred so that there are not two identical switch blocks to - maintain. Review: https://reviewboard.asterisk.org/r/674/ - ........ ................ - -2010-05-20 23:34 +0000 [r264829] Richard Mudgett <rmudgett@digium.com> - - * /, main/callerid.c: Merged revisions 264828 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r264828 | rmudgett | 2010-05-20 18:29:43 -0500 (Thu, 20 May 2010) - | 13 lines Merged revisions 264820 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r264820 | rmudgett | 2010-05-20 18:23:21 -0500 (Thu, 20 May 2010) - | 6 lines ast_callerid_parse() had a path that left name - uninitialized. Several callers of ast_callerid_parse() do not - initialize the name parameter before calling thus there is the - potential to use an uninitialized pointer. ........ - ................ - -2010-05-20 22:24 +0000 [r264753-264783] Tilghman Lesher <tlesher@digium.com> - - * main/pbx.c, /: Merged revisions 264779 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r264779 | - tilghman | 2010-05-20 17:23:32 -0500 (Thu, 20 May 2010) | 8 lines - Let ExtensionState resolve dynamic hints. (closes issue #16623) - Reported by: tilghman Patches: 20100116__issue16623.diff.txt - uploaded by tilghman (license 14) Tested by: lmadsen ........ - - * apps/app_stack.c, /: Merged revisions 264752 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r264752 | - tilghman | 2010-05-20 16:28:53 -0500 (Thu, 20 May 2010) | 7 lines - Error message fix. (closes issue #17356) Reported by: kenner - Patches: app_stack.c.diff uploaded by kenner (license 1040) - ........ - -2010-05-19 22:10 +0000 [r264453] Mark Michelson <mmichelson@digium.com> - - * include/asterisk/_private.h, include/asterisk/options.h, - main/asterisk.c, main/loader.c, main/channel.c, /, - channels/chan_sip.c: Merged revisions 264452 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r264452 | - mmichelson | 2010-05-19 16:29:08 -0500 (Wed, 19 May 2010) | 86 - lines Fix transcode_via_sln option with SIP calls and improve PLC - usage. From reviewboard: The problem here is a bit complex, so - try to bear with me... It was noticed by a Digium customer that - generic PLC (as configured in codecs.conf) did not appear to - actually be having any sort of benefit when packet loss was - introduced on an RTP stream. I reproduced this issue myself by - streaming a file across an RTP stream and dropping approx. 5% of - the RTP packets. I saw no real difference between when PLC was - enabled or disabled when using wireshark to analyze the RTP - streams. After analyzing what was going on, it became clear that - one of the problems faced was that when running my tests, the - translation paths were being set up in such a way that PLC could - not possibly work as expected. To illustrate, if packets are lost - on channel A's read stream, then we expect that PLC will be - applied to channel B's write stream. The problem is that generic - PLC can only be done when there is a translation path that moves - from some codec to SLINEAR. When I would run my tests, I found - that every single time, read and write translation paths would be - set up on channel A instead of channel B. There appeared to be no - real way to predict which channel the translation paths would be - set up on. This is where Kevin swooped in to let me know about - the transcode_via_sln option in asterisk.conf. It is supposed to - work by placing a read translation path on both channels from the - channel's rawreadformat to SLINEAR. It also will place a write - translation path on both channels from SLINEAR to the channel's - rawwriteformat. Using this option allows one to predictably set - up translation paths on all channels. There are two problems with - this, though. First and foremost, the transcode_via_sln option - did not appear to be working properly when I was placing a SIP - call between two endpoints which did not share any common - formats. Second, even if this option were to work, for PLC to be - applied, there had to be a write translation path that would go - from some format to SLINEAR. It would not work properly if the - starting format of translation was SLINEAR. The one-line change - presented in this review request in chan_sip.c fixed the first - issue for me. The problem was that in sip_request_call, the - jointcapability of the outbound channel was being set to the - format passed to sip_request_call. This is nativeformats of the - inbound channel. Because of this, when - ast_channel_make_compatible was called by app_dial, both channels - already had compatibly read and write formats. Thus, no - translation path was set up at the time. My change is to set the - jointcapability of the sip_pvt created during sip_request_call to - the intersection of the inbound channel's nativeformats and the - configured peer capability that we determined during the earlier - call to create_addr. Doing this got the translation paths set up - as expected when using transcode_via_sln. The changes presented - in channel.c fixed the second issue for me. First and foremost, - when Asterisk is started, we'll read codecs.conf to see the value - of the genericplc option. If this option is set, and ast_write is - called for a frame with no data, then we will attempt to fill in - the missing samples for the frame. The implementation uses a - channel datastore for maintaining the PLC state and for creating - a buffer to store PLC samples in. Even when we receive a frame - with data, we'll call plc_rx so that the PLC state will have - knowledge of the previous voice frame, which it can use as a - basis for when it comes time to actually do a PLC fill-in. So, - reviewers, now I ask for your help. First off, there's the one - line change in chan_sip that I have put in. Is it right? By my - logic it seems correct, but I'm sure someone can tell me why it - is not going to work. This is probably the change I'm least - concerned about, though. What concerns me much more is the set of - changes in channel.c. First off, am I even doing it right? When I - run tests, I can clearly see that when PLC is activated, I see a - significant increase in RTP traffic where I would expect it to - be. However, in my humble opinion, the audio sounds kind of - crappy whenever the PLC fill-in is done. It sounds worse to me - than when no PLC is used at all. I need someone to review the - logic I have used to be sure that I'm not misusing anything. As - far as I can see my pointer arithmetic is correct, and my use of - AST_FRIENDLY_OFFSET should be correct as well, but I'm sure - someone can point out somewhere where I've done something - incorrectly. As I was writing this review request up, I decided - to give the code a test run under valgrind, and I find that for - some reason, calls to plc_rx are causing some invalid reads. - Apparently I'm reading past the end of a buffer somehow. I'll - have to dig around a bit to see why that is the case. If it's - obvious to someone reviewing, speak up! Finally, I have one other - proposal that is not reflected in my code review. Since without - transcode_via_sln set, one cannot predict or control where a - translation path will be up, it seems to me that the current - practice of using PLC only when transcoding to SLINEAR is not - useful. I recommend that once it has been determined that the - method used in this code review is correct and works as expected, - then the code in translate.c that invokes PLC should be removed. - Review: https://reviewboard.asterisk.org/r/622/ ........ - -2010-05-19 20:31 +0000 [r264405] David Vossel <dvossel@digium.com> - - * main/udptl.c, /: Merged revisions 264400 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r264400 | - dvossel | 2010-05-19 15:30:33 -0500 (Wed, 19 May 2010) | 11 lines - fixes infinite loop during udptl.c's decode_open_type When - decode_length returns the length there is a check to see if that - length is negative, if so the decode loop breaks as this means - the limit has been reached. The problem here is that length is an - unsigned int, so length can never be negative. This resulted in - an infinite loop. (issue #17352) ........ - -2010-05-19 20:27 +0000 [r264336-264388] Matthew Nicholson <mnicholson@digium.com> - - * main/udptl.c, /: Merged revisions 264379 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r264379 | - mnicholson | 2010-05-19 15:26:27 -0500 (Wed, 19 May 2010) | 4 - lines Cast an unsigned int to a signed int when comparing it with - 0. (AST-377) ........ - - * apps/app_speech_utils.c, /: Merged revisions 264335 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r264335 | mnicholson | 2010-05-19 15:02:57 -0500 - (Wed, 19 May 2010) | 12 lines Merged revisions 264334 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r264334 | mnicholson | 2010-05-19 15:01:38 -0500 (Wed, 19 May - 2010) | 5 lines Set quieted flag when receiving a dtmf tone - during playback in speechbackground. (closes issue #16966) - Reported by: asackheim ........ ................ - -2010-05-19 19:25 +0000 [r264332] David Vossel <dvossel@digium.com> - - * /, channels/chan_sip.c: Merged revisions 264331 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r264331 | - dvossel | 2010-05-19 14:21:04 -0500 (Wed, 19 May 2010) | 13 lines - fixes crash in check_rtp_timeout During deadlock avoidance the - sip dialog pvt is locked and unlocked. When this occurs we have - no guarantee the pvt's owner is still valid. We were trying to - access the pvt's owner after this without checking to see if it - still existed first. (closes issue #17271) Reported by: under - Patches: check_rtp_timeout.diff uploaded by under (license 914) - Tested by: dvossel ........ - -2010-05-19 17:49 +0000 [r264205-264250] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/options.h, /, configure, - include/asterisk/autoconfig.h.in, configure.ac: Merged revisions - 264249 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r264249 | tilghman | 2010-05-19 12:48:31 -0500 (Wed, 19 May 2010) - | 24 lines Merged revisions 264248 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r264248 | tilghman | 2010-05-19 12:41:29 -0500 (Wed, 19 May 2010) - | 17 lines Internal timing is now on by default, if you're using - DAHDI 2.3 or above. The reason for ensuring DAHDI 2.3 or above is - that this version ensures that a timer is always available, - whereas in previous versions, it was possible for DAHDI to be - loaded, but have no drivers to actually generate timing. If - internal_timing was turned on in this circumstance, a complete - lack of audio would result. This is the reason why - internal_timing was not on by default. However, now that DAHDI - ensures the availability of a timer, there is no reason for this - setting to be off (and in fact, it solves a great many initial - user problems). (closes issue #15932) Reported by: dimas Patches: - 20100519__issue15932.diff.txt uploaded by tilghman (license 14) - Tested by: tilghman ........ ................ - - * main/dsp.c, /: Merged revisions 264204 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r264204 | - tilghman | 2010-05-19 11:42:20 -0500 (Wed, 19 May 2010) | 9 lines - Keep track of digit duration, when we're decoding inband to pass - DTMF frames. (closes issue #17235) Reported by: frawd Patches: - new_dtmf_dsp_len.patch uploaded by frawd (license 610) - 20100518__issue17235.diff.txt uploaded by tilghman (license 14) - Tested by: frawd ........ - -2010-05-19 14:47 +0000 [r264115] David Vossel <dvossel@digium.com> - - * main/rtp.c, /: Merged revisions 264114 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r264114 | - dvossel | 2010-05-19 09:38:02 -0500 (Wed, 19 May 2010) | 13 lines - fixes crash during dtmf During the processing of Cisco dtmf the - dtmf samples were not being calculated correctly. In an attempt - to determine what sample rate was being used, a NULL frame was - processed which caused a crash. This patch resolves this. (closes - issue #17248) Reported by: falves11 Patches: issue_17248.diff - uploaded by dvossel (license 671) ........ - -2010-05-19 08:15 +0000 [r264032] Alec L Davis <sivad.a@paradise.net.nz> - - * /, configs/indications.conf.sample: Merged revisions 264031 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r264031 | alecdavis | 2010-05-19 20:09:14 +1200 (Wed, 19 - May 2010) | 8 lines fix incorrectly typed indications for [nz] - stutter and dialrecall (closes issue #17359) Reported by: - alecdavis Patches: bug17359.diff.txt uploaded by alecdavis - (license 585) ........ - -2010-05-19 06:41 +0000 [r263951] Tilghman Lesher <tlesher@digium.com> - - * main/dsp.c, /: Merged revisions 263950 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r263950 | tilghman | 2010-05-19 01:41:04 -0500 (Wed, 19 May 2010) - | 15 lines Merged revisions 263949 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r263949 | tilghman | 2010-05-19 01:32:27 -0500 (Wed, 19 May 2010) - | 8 lines Because progress is called multiple times, across - several frames, we must persist states when detecting multitone - sequences. (closes issue #16749) Reported by: dant Patches: - dsp.c-bug16749-1.patch uploaded by dant (license 670) Tested by: - dant ........ ................ - -2010-05-18 22:49 +0000 [r263906] David Vossel <dvossel@digium.com> - - * main/strings.c, /: Merged revisions 263904 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r263904 | - dvossel | 2010-05-18 17:48:51 -0500 (Tue, 18 May 2010) | 9 lines - fixes segfault on logging (closes issue #17331) Reported by: - under Patches: utils.diff uploaded by under (license 914) - segfault_on_logging.diff uploaded by dvossel (license 671) Tested - by: under, dvossel ........ - -2010-05-18 19:41 +0000 [r263809] Jeff Peeler <jpeeler@digium.com> - - * apps/app_directory.c, /: Merged revisions 263807 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r263807 | jpeeler | 2010-05-18 14:27:34 -0500 - (Tue, 18 May 2010) | 17 lines Merged revisions 263769 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r263769 | jpeeler | 2010-05-18 13:54:58 -0500 (Tue, 18 May 2010) - | 10 lines Modify directory name reading to be interrupted with - operator or pound escape. In the case of accidentally entering - the wrong first three letters for the reading, users could be - very frustrated if the name listing is very long. This allows - interrupting the reading by pressing 0 or #. 0 will attempt to - execute a configured operator (o) extension and # will exit and - proceed in the dialplan. ABE-2200 ........ ................ - -2010-05-17 22:10 +0000 [r263642] Mark Michelson <mmichelson@digium.com> - - * /, main/devicestate.c: Merged revisions 263640 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r263640 | mmichelson | 2010-05-17 17:08:01 -0500 (Mon, 17 May - 2010) | 16 lines Merged revisions 263639 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r263639 | mmichelson | 2010-05-17 17:00:28 -0500 (Mon, 17 May - 2010) | 10 lines Fix logic error when checking for a devstate - provider. When using strsep, if one of the list of specified - separators is not found, it is the first parameter to strsep - which is now NULL, not the pointer returned by strsep. This issue - isn't especially severe in that the worst it is likely to do is - waste some cycles when a device with no '/' and no ':' is passed - to ast_device_state. ........ ................ - -2010-05-17 19:37 +0000 [r263587-263590] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 263589 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r263589 | tilghman | 2010-05-17 14:31:15 -0500 (Mon, 17 May 2010) - | 9 lines With IMAP backend, messages in INBOX were counted twice - for MWI. (closes issue #17135) Reported by: edhorton Patches: - 20100513__issue17135.diff.txt uploaded by tilghman (license 14) - 17135_2.diff uploaded by ebroad (license 878) Tested by: - edhorton, ebroad ........ - - * main/app.c: Don't close 'n', just close 'above_n'. (closes issue - #17345) Reported by: wdoekes - -2010-05-17 14:41 +0000 [r263376-263458] Leif Madsen <lmadsen@digium.com> - - * main/manager.c, /: Merged revisions 263457 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r263457 | lmadsen | 2010-05-17 09:37:35 -0500 (Mon, 17 May 2010) - | 19 lines Recorded merge of revisions 263456 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r263456 | lmadsen | 2010-05-17 09:35:18 -0500 (Mon, 17 May 2010) - | 11 lines Manager cookies are not compatible with RFC2109. The - Version field in the cookies we're setting contain quotes around - the version number which is not compatible with RFC2109 and - breaks some implementations. (closes issue #17231) Reported by: - ecarruda Patches: manager_rfc2109-trunk-v1.patch uploaded by - ecarruda (license 559) manager_rfc2109-1.6.2-v1.patch uploaded by - ecarruda (license 559) Tested by: ecarruda, russell ........ - ................ - - * sounds/Makefile, /: Merged revisions 263375 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r263375 | lmadsen | 2010-05-17 09:05:33 -0500 (Mon, 17 May 2010) - | 16 lines Merged revisions 263374 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r263374 | lmadsen | 2010-05-17 09:04:57 -0500 (Mon, 17 May 2010) - | 8 lines Update link to new version of core sounds. The latest - version of the core sounds files 1.4.19 now includes the missing - queue-minute sound file which is called by app_queue but which - has been missing. (closes issue #17123) Reported by: n8ideas - ........ ................ - -2010-05-17 13:03 +0000 [r263293] David Vossel <dvossel@digium.com> - - * CHANGES, channels/chan_dahdi.c: backport of DAHDI dynamic buffer - policy dialstring option - -2010-05-15 23:41 +0000 [r263202] Paul Belanger <paul.belanger@polybeacon.com> - - * /, codecs/gsm/Makefile: Merged revisions 252488 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r252488 | - tilghman | 2010-03-15 12:27:08 -0400 (Mon, 15 Mar 2010) | 9 lines - Make the Makefile logic more explicit and move the Snow Leopard - logic down to where it's not executed on non-Darwin systems. - (closes issue #17028) Reported by: pabelanger Patches: - issue17028_20100315.patch uploaded by seanbright (license 71) - 20100315__issue17028.diff.txt uploaded by tilghman (license 14) - Tested by: tilghman, pabelanger ........ - -2010-05-13 22:13 +0000 [r263070] Richard Mudgett <rmudgett@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 263069 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r263069 | rmudgett | 2010-05-13 17:01:36 -0500 (Thu, 13 May 2010) - | 1 line Fix inverted logic in cli command: ss7 set debug on/off - ........ - -2010-05-13 15:36 +0000 [r262898] Russell Bryant <russell@digium.com> - - * channels/chan_console.c, /: Merged revisions 262897 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r262897 | russell | 2010-05-13 10:36:12 -0500 (Thu, 13 May 2010) - | 4 lines Fix an off by one error that causes a crash. Thanks to - Raymond Burke for pointing it out. ........ - -2010-05-12 20:01 +0000 [r262801] Paul Belanger <paul.belanger@polybeacon.com> - - * 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 19:53 +0000 [r262797-262799] Leif Madsen <lmadsen@digium.com> - - * res/ael/pval.c, /: Merged revisions 262798 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r262798 | - lmadsen | 2010-05-12 14:53:10 -0500 (Wed, 12 May 2010) | 7 lines - Revert previous WARNING message removal. Marquis42 suggested a - better method of doing what I wanted because I ended up removing - the WARNING message for all instances when really I just wanted - to remove it for the 'return' keyword, not everything. (issue - #17145) ........ - - * res/ael/pval.c, /: Merged revisions 262796 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r262796 | - lmadsen | 2010-05-12 14:31:42 -0500 (Wed, 12 May 2010) | 4 lines - Remove unnecessary WARNING message in ael/pval.c (closes issue - #17145) Reported by: okrief ........ - -2010-05-12 18:03 +0000 [r262746] David Vossel <dvossel@digium.com> - - * /, 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 [r262516-262659] Tilghman Lesher <tlesher@digium.com> - - * /, 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 19:58 +0000 [r262425] Jason Parker <jparker@digium.com> - - * /, 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 19:41 +0000 [r262415-262420] Paul Belanger <paul.belanger@polybeacon.com> - - * 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 - ........ - - * /, 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 [r262340] Tilghman Lesher <tlesher@digium.com> - - * 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 19:06 +0000 [r262237-262241] David Vossel <dvossel@digium.com> - - * /, apps/app_directed_pickup.c: Merged revisions 262240 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r262240 | dvossel | 2010-05-10 14:06:08 -0500 (Mon, 10 - May 2010) | 9 lines fixes PickupChan application (closes issue - #16863) Reported by: schern Patches: app_directed_pickup.c.patch - uploaded by schern (license 995) for_trunk.diff uploaded by - cjacobsen (license 1029) Tested by: Graber, cjacobsen, lathama, - rickead2000, dvossel ........ - - * 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 16:39 +0000 [r262155] Tilghman Lesher <tlesher@digium.com> - - * /, 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 [r261916-262105] Tilghman Lesher <tlesher@digium.com> - - * 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 - ........ - - * /, funcs/func_odbc.c: Merged revisions 261917 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r261917 | - tilghman | 2010-05-07 15:54:35 -0500 (Fri, 07 May 2010) | 8 lines - Double free crash (closes issue #17245) Reported by: - thedavidfactor Patches: 20100426__issue17245.diff.txt uploaded by - tilghman (license 14) Tested by: murraytm ........ - - * /, 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 ........ - -2010-05-07 16:08 +0000 [r261868] Leif Madsen <lmadsen@digium.com> - - * UPGRADE-1.6.txt, /: Merged revisions 261867 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r261867 | - lmadsen | 2010-05-07 11:05:24 -0500 (Fri, 07 May 2010) | 6 lines - Update UPGRADE-1.6.txt stating insecure=very has been removed. - (closes issue #17282) Reported by: stuarth Tested by: stuarth - ........ - -2010-05-06 20:13 +0000 [r261739] Jeff Peeler <jpeeler@digium.com> - - * 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 [r261612] Jason Parker <jparker@digium.com> - - * 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 15:43 +0000 [r261563] Tilghman Lesher <tlesher@digium.com> - - * /, 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 ........ - -2010-06-01 Leif Madsen <lmadsen@digium.com> - - * Asterisk 1.6.2.8 Released. - -2010-05-26 Leif Madsen <lmadsen@digium.com> - - * Asterisk 1.6.2.8-rc2 Released. - -2010-05-26 10:56 -0500 [r265891] Matt Nicholson <mnicholson@digium.com> - - * Merged r265610 from 1.4: - - Don't mark the cdr records of unanswered queue calls with "NOANSWER". - This restores the behavior prior to r258670. - - (closes issue #17334) - Reported by: jvandal - Patches: - queue-cdr-fixes1.diff uploaded by mnicholson (license 96) - Tested by: aragon, jvandal - -2010-05-06 Leif Madsen <lmadsen@digium.com> - - * Asterisk 1.6.2.8-rc1 Released - -2010-05-06 14:07 +0000 [r261498-261499] Russell Bryant <russell@digium.com> - - * tests/test_heap.c: Add test case that ensures the heap handles - arbitrary removals properly. (issue #17277) Reported by: - cappucinoking Patches: test_heap.diff uploaded by cappucinoking - (license 1036) Tested by: cappucinoking, russell - - * /, 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 07:43 +0000 [r261453] Tzafrir Cohen <tzafrir.cohen@xorcom.com> - - * 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 19:08 +0000 [r261233-261315] Paul Belanger <paul.belanger@polybeacon.com> - - * /, 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/ ........ ................ - - * apps/app_queue.c, /: Merged revisions 261232 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r261232 | - pabelanger | 2010-05-05 11:42:07 -0400 (Wed, 05 May 2010) | 10 - lines 'queue reset stats' erroneously clears wrapuptime - configuration. Resets each member's lastcall to 0 now. (closes - issue #17262, #16519) Reported by: rain Patches: - wrapuptime_reset_fix.diff uploaded by rain (license 327) Tested - by: rain ........ - -2010-05-04 23:55 +0000 [r261098] Tilghman Lesher <tlesher@digium.com> - - * 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 19:01 +0000 [r260927] Jeff Peeler <jpeeler@digium.com> - - * 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 16:58 +0000 [r260884] Matthew Nicholson <mnicholson@digium.com> - - * configs/sip.conf.sample, include/asterisk/frame.h, - main/channel.c, /, channels/chan_sip.c: Merged revisions 254450 - via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r254450 | kpfleming | 2010-03-25 10:27:31 -0500 (Thu, 25 - Mar 2010) | 49 lines Improve handling of T.38 re-INVITEs that - arrive before a T.38-capable application is executing on a - channel. This patch addresses an issue found during working with - end-users using res_fax. If an incoming call is answered in the - dialplan, or jumps to the 'fax' extension due to reception of a - CNG tone (with faxdetect enabled), and then the remote endpoint - sends a T.38 re-INVITE, it is possible for the channel's T.38 - state to be 'T38_STATE_NEGOTIATING' when the application starts - up. Unfortunately, even if the application wants to use T.38, it - can't respond to the peer's negotiation request, because the - AST_CONTROL_T38_PARAMETERS control frame that chan_sip sent - originally has been lost, and the application needs the content - of that frame to be able to formulate a reply. This patch adds a - new 'request' type to AST_CONTROL_T38_PARAMETERS, - AST_T38_REQUEST_PARMS. If the application sends this request, - chan_sip will re-send the original control frame (with - AST_T38_REQUEST_NEGOTIATE as the request type), and the - application can respond as normal. If this occurs within the five - second timeout in chan_sip, the automatic cancellation of the - peer reinvite will be stopped, and the application will 'own' the - negotiation process from that point onwards. This also improves - the code path in chan_sip to allow sip_indicate(), when called - for AST_CONTROL_T38_PARAMETERS, to be able to return a non-zero - response, which should have been in place before since the - control frame *can* fail to be processed properly. It also - modifies ast_indicate() to return whatever result the channel - driver returned for this control frame, rather than converting - all non-zero results into '-1'. Finally, the new request type - intentionally returns a positive value, so that an application - that sends AST_T38_REQUEST_PARMS can know for certain whether the - channel driver accepted it and will be replying with a control - frame of its own, or whether it was ignored (if the - sip_indicate()/ast_indicate() path had properly supported failure - responses before, this would not be necessary). This patch also - modifies res_fax to take advantage of the new request. In - addition, this patch makes sip_t38_abort() actually lock the - private structure before doing its work... bad programmer, no - donut. This patch also enhances chan_sip's 'faxdetect' support to - allow triggering on T.38 re-INVITEs received as well as CNG tone - detection. Review: https://reviewboard.asterisk.org/r/556/ - ........ - -2010-05-04 15:51 +0000 [r260746-260805] Jason Parker <jparker@digium.com> - - * /, 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 17:42 +0000 [r260743] Paul Belanger <paul.belanger@polybeacon.com> - - * 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 [r260571] Leif Madsen <lmadsen@digium.com> - - * 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 22:48 +0000 [r260441] Jeff Peeler <jpeeler@digium.com> - - * 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 20:22 +0000 [r260373] Mark Michelson <mmichelson@digium.com> - - * 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 - ........ ................ - -2010-04-30 06:22 +0000 [r260281-260303] Tilghman Lesher <tlesher@digium.com> - - * /, main/app.c: Merged revisions 260292 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r260292 | - tilghman | 2010-04-30 01:19:35 -0500 (Fri, 30 Apr 2010) | 13 - lines Don't allow file descriptors to go above 64k, when we're - closing them in a fork(2). This saves time, when, even though the - system allows the process limit to be that high, the practical - limit is much lower. (closes issue #17223) Reported by: - dbackeberg Patches: 20100423__issue17223.diff.txt uploaded by - tilghman (license 14) Tested by: dbackeberg ........ - - * configs/extensions.conf.sample, /: Merged revisions 260280 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r260280 | tilghman | 2010-04-30 00:23:56 -0500 (Fri, 30 - Apr 2010) | 7 lines Logic fixups for a sample FREENUM dialplan - context. (closes issue #17263) Reported by: pprindeville Patches: - freenum-dialplan.patch#3 uploaded by pprindeville (license 347) - ........ - -2010-04-29 23:13 +0000 [r260234] Richard Mudgett <rmudgett@digium.com> - - * 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 [r260156] Tilghman Lesher <tlesher@digium.com> - - * 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 15:35 +0000 [r260051] David Vossel <dvossel@digium.com> - - * main/audiohook.c, /, include/asterisk/audiohook.h: 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 22:36 +0000 [r259959] Mark Michelson <mmichelson@digium.com> - - * /, 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 21:26 +0000 [r259899] David Vossel <dvossel@digium.com> - - * 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 [r259854] Leif Madsen <lmadsen@digium.com> - - * 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 20:34 +0000 [r259781-259851] Jason Parker <jparker@digium.com> - - * /, 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 [r259681] Jeff Peeler <jpeeler@digium.com> - - * 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 22:46 +0000 [r259616] Richard Mudgett <rmudgett@digium.com> - - * 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 [r259528] Leif Madsen <lmadsen@digium.com> - - * 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 21:25 +0000 [r259356-259486] Jason Parker <jparker@digium.com> - - * main/editline/configure.in, /, main/editline/configure, - main/editline/Makefile.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. ........ - - * /: Unblock revision 259439. - - * /, 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 19:03 +0000 [r259310] Richard Mudgett <rmudgett@digium.com> - - * 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 [r259103-259109] Mark Michelson <mmichelson@digium.com> - - * 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 ........ ................ - -2010-04-26 16:00 +0000 [r258935] Leif Madsen <lmadsen@digium.com> - - * /, channels/chan_sip.c: Merged revisions 258934 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r258934 | - lmadsen | 2010-04-26 10:59:34 -0500 (Mon, 26 Apr 2010) | 7 lines - Small error in the T.140 RTP port verbose log. (closes issue - #16988) Reported by: frawd Patches: chan_sip_sdp_verbose_fix.diff - uploaded by frawd (license 610) Tested by: russell ........ - -2010-04-25 18:14 +0000 [r258779] Tilghman Lesher <tlesher@digium.com> - - * 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 22:15 +0000 [r258676] Matthew Nicholson <mnicholson@digium.com> - - * 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. - ................ - -2010-04-22 21:58 +0000 [r258516-258672] Russell Bryant <russell@digium.com> - - * /, main/event.c: Merged revisions 258632 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk For 1.6.2, only - merge the bug fixes, not the unit test. ........ r258632 | - russell | 2010-04-22 16:06:53 -0500 (Thu, 22 Apr 2010) | 22 lines - Add ast_event subscription unit test and fix some ast_event API - bugs. This patch introduces another test in test_event.c that - exercises most of the subscription related ast_event API calls. I - made some minor additions to the existing event allocation test - to increase API coverage by the test code. Finally, I made a list - in a comment of API calls not yet touched by the test module as a - to-do list for future test development. During the development of - this test code, I discovered a number of bugs in the event API. - 1) subscriptions to AST_EVENT_ALL were not handled appropriately - in a couple of different places. The API allows a subscription to - all event types, but with IE parameters, just as if it was a - subscription to a specific event type. However, the parameters - were being ignored. This affected ast_event_check_subscriber() - and event distribution to subscribers. 2) Some of the logic in - ast_event_check_subscriber() for checking subscriptions against - query parameters was wrong. Review: - https://reviewboard.asterisk.org/r/617/ ........ - - * /, doc/tex/channelvariables.tex: Merged revisions 258515 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r258515 | russell | 2010-04-22 12:36:34 -0500 (Thu, 22 - Apr 2010) | 2 lines Add MEETMEBOOKID from r256019. ........ - -2010-04-21 22:11 +0000 [r258436] Jeff Peeler <jpeeler@digium.com> - - * 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 19:44 +0000 [r258384-258386] Leif Madsen <lmadsen@digium.com> - - * doc/tex/asterisk.tex: Remove missed line in previous merge. - (issue #17220) - - * configure: Forgot to merge the updated configure script. (issue - #17220) - - * doc/tex/localchannel.tex, doc/tex/enum.tex, makeopts.in, - doc/tex/asterisk.tex, Makefile, /, doc/tex/Makefile, - configure.ac, doc/tex/phoneprov.tex, doc/tex, doc/tex/ael.tex, - build_tools/prep_tarball: Merged revisions 258351 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r258351 | lmadsen | 2010-04-21 14:18:35 -0500 (Wed, 21 Apr 2010) - | 20 lines Add ability to generate ASCII documentation from the - TeX files. These changes add the ability to run 'make - asterisk.txt' just like the existing 'make asterisk.pdf' commands - to generate a text document from the TeX files we have in the - doc/tex/ directory. I've also updated a few of the .tex files - because they weren't properly escaping certain characters so they - would show up as Unicode characters (like [U+021C]). Made changes - to the configure scripts so it would detect the catdvi program - which is required to convert the .dvi file generated by latex. - I've also added a few lines to the build_tools/prep_tarball - script so that the text documentation gets generated and added to - future tarballs of Asterisk releases. (closes issue #17220) - Reported by: lmadsen Patches: asterisk.txt.patch uploaded by - lmadsen (license 10) asterisk.txt.patch-v4 uploaded by pabelanger - (license 224) Tested by: lmadsen, pabelanger ........ - -2010-04-21 18:19 +0000 [r258314] David Vossel <dvossel@digium.com> - - * /, 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 19:03 +0000 [r258148-258150] Leif Madsen <lmadsen@digium.com> - - * /, configs/cli_aliases.conf.sample: Merged revisions 258149 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r258149 | lmadsen | 2010-04-20 14:02:49 -0500 (Tue, 20 - Apr 2010) | 1 line Add 'soft hangup' alias per Steve Johnson on - asterisk-users. ........ - - * configs/extensions.conf.sample, /: Merged revisions 258147 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r258147 | lmadsen | 2010-04-20 13:38:39 -0500 (Tue, 20 - Apr 2010) | 8 lines Add example dialplan for dialing ISN numbers - (http://www.freenum.org). Minor tweaks and documentation added by - me. (closes issue #17058) Reported by: pprindeville Patches: - freenum.patch#5 uploaded by pprindeville (license 347) Tested by: - lmadsen ........ - -2010-04-20 18:04 +0000 [r258108] Jeff Peeler <jpeeler@digium.com> - - * 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 ........ ................ - -2010-04-20 18:02 +0000 [r258107] Leif Madsen <lmadsen@digium.com> - - * contrib/scripts/sip-friends.sql, /: Merged revisions 258106 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r258106 | lmadsen | 2010-04-20 13:01:28 -0500 (Tue, 20 - Apr 2010) | 7 lines Add missing 'useragent' field to - sip-friends.sql file. (closes issue #17171) Reported by: thehar - Patches: sip-friends.patch uploaded by thehar (license 831) - Tested by: pabelanger, thehar ........ - -2010-04-19 21:58 +0000 [r257948-257950] Jason Parker <jparker@digium.com> - - * main/indications.c, /: Merged revisions 257949 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r257949 | - qwell | 2010-04-19 16:57:56 -0500 (Mon, 19 Apr 2010) | 1 line - Change log message to match severity. ........ - - * main/indications.c, /: Merged revisions 257947 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r257947 | - qwell | 2010-04-19 16:49:30 -0500 (Mon, 19 Apr 2010) | 6 lines - Don't consider a missing indications.conf to be a critical error. - There were many changes in revision 176627 which would avoid the - error that a missing config would have caused. Other than this, - there are no other config files (including asterisk.conf, - surprisingly) that are required. ........ - -2010-04-19 18:30 +0000 [r257850] Terry Wilson <twilson@digium.com> - - * /, 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 [r257771] Tilghman Lesher <tlesher@digium.com> - - * 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 21:47 +0000 [r257740] Dwayne M. Hubbard <dwayne.hubbard@gmail.com> - - * 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 21:34 +0000 [r257510-257597] Tilghman Lesher <tlesher@digium.com> - - * 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 19:42 +0000 [r257344-257428] Leif Madsen <lmadsen@digium.com> - - * 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 [r257265] Tilghman Lesher <tlesher@digium.com> - - * 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 19:20 +0000 [r257210] Tilghman Lesher <tlesher@digium.com> - - * /, 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/ ........ - -2010-04-13 18:43 +0000 [r257184] Matthew Nicholson <mnicholson@digium.com> - - * 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 16:38 +0000 [r257068] Tilghman Lesher <tlesher@digium.com> - - * 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 17:30 +0000 [r256822-256902] Leif Madsen <lmadsen@digium.com> - - * 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 ........ - ................ - - * apps/app_voicemail.c, /: Merged revisions 256860 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r256860 | lmadsen | 2010-04-12 11:16:43 -0500 (Mon, 12 Apr 2010) - | 3 lines Remove silly debug message that is not useful. (issue - #17159) ........ - - * /, main/logger.c: Merged revisions 256821 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r256821 | - lmadsen | 2010-04-12 09:39:37 -0500 (Mon, 12 Apr 2010) | 8 lines - CLI command logger set level auto complete. A simple patch to - enable auto tab complete. (closes issue #17152) Reported by: - pabelanger Patches: 0017152.patch uploaded by pabelanger (license - 224) ........ - -2010-04-08 22:03 +0000 [r256483] Tilghman Lesher <tlesher@digium.com> - - * main/app.c: Backport /proc/%d/fd method of closing file - descriptors to 1.6.2. - -2010-04-06 19:40 +0000 [r256373] Tilghman Lesher <tlesher@digium.com> - - * /, 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 18:53 +0000 [r256268-256368] Richard Mudgett <rmudgett@digium.com> - - * channels/chan_dahdi.c: CallerID channel DAHDI port FXS are empty - after the first call. The bug is exposed if MFC/R2 support is - built into asterisk (i.e., openr2.h is present in the include - path). Code that unconditionally clears the CallerID name and - number is included. Also fixed a malformed if test in mkintf() - added by issue 15883. Converted the if statement to a switch - statement for clarity. Regression of the issue 15883 fix. (closes - issue #16968) Reported by: grecco Patches: issue16968.patch - uploaded by rmudgett (license 664) (closes issue #16747) Reported - by: viniciusfontes - - * 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 <lmadsen@digium.com> - - * Asterisk 1.6.2.7 Released - -2010-04-29 Leif Madsen <lmadsen@digium.com> - - * Asterisk 1.6.2.7-rc3 Released - -2010-04-29 10:31 +0000 [r260053] David Vossel <dvossel@digium.com> - - * 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 [r259899] David Vossel <dvossel@digium.com> - - * 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 <lmadsen@digium.com> - - * Asterisk 1.6.2.7-rc2 Released - -2010-04-13 [r257210] Tilghman Lesher <tlesher@digium.com> - - 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/ - -2010-04-05 Leif Madsen <lmadsen@digium.com> - - * Asterisk 1.6.2.7-rc1 Released - -2010-04-05 15:15 +0000 [r256162] Leif Madsen <lmadsen@digium.com> - - * 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 23:56 +0000 [r256013-256020] Russell Bryant <russell@digium.com> - - * /, apps/app_meetme.c: Merged revisions 256019 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r256019 | - russell | 2010-04-02 18:55:57 -0500 (Fri, 02 Apr 2010) | 10 lines - Export MEETMEBOOKID and fix pin-less conferences with realtime - conferences (closes issue #16866) Reported by: DEA Patches: - rt-meetme-options.txt uploaded by DEA (license 3) Tested by: DEA - Review: https://reviewboard.asterisk.org/r/582/ ........ - - * 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 [r255955] Tilghman Lesher <tlesher@digium.com> - - * 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 [r255676-255816] Tilghman Lesher <tlesher@digium.com> - - * /, 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 17:49 +0000 [r255505] Leif Madsen <lmadsen@digium.com> - - * configs/sip.conf.sample, apps/app_dial.c: 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 20:58 +0000 [r255326-255413] Russell Bryant <russell@digium.com> - - * /, 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 19:28 +0000 [r255023-255067] Leif Madsen <lmadsen@digium.com> - - * configs/sip.conf.sample, /: Merged revisions 255066 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r255066 | lmadsen | 2010-03-26 14:27:56 -0500 (Fri, 26 Mar 2010) - | 6 lines Replace some documentation from 1.6.x back into trunk. - This documentation associated wth tlsbindaddr is still useful so - lets synchronize it between trunk and 1.6.x branches. (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 [r254770-254805] Jason Parker <jparker@digium.com> - - * 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. ........ - ................ - - * 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 20:09 +0000 [r254721] Russell Bryant <russell@digium.com> - - * 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 17:47 +0000 [r254556] Mark Michelson <mmichelson@digium.com> - - * 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 [r254548] Sean Bright <sean@malleable.com> - - * channels/chan_sip.c: Initialize stream to avoid a compilation - error. - -2010-03-25 17:12 +0000 [r254542] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: Fix potential crashes from trying to - reference nonexistent RTP streams. - -2010-03-25 16:26 +0000 [r254499] Terry Wilson <twilson@digium.com> - - * /, 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 16:22 +0000 [r254482] Mark Michelson <mmichelson@digium.com> - - * 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 15:21 +0000 [r254447] Leif Madsen <lmadsen@digium.com> - - * /, 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 17:19 +0000 [r254288] Jeff Peeler <jpeeler@digium.com> - - * 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 22:05 +0000 [r254131] Tzafrir Cohen <tzafrir.cohen@xorcom.com> - - * 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 [r254068] Jeff Peeler <jpeeler@digium.com> - - * 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 19:55 +0000 [r253801] Matthew Nicholson <mnicholson@digium.com> - - * /, 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 [r253714-253760] Tilghman Lesher <tlesher@digium.com> - - * /, 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 17:33 +0000 [r253595-253620] Russell Bryant <russell@digium.com> - - * cdr/cdr_pgsql.c, main/stdtime/localtime.c, main/tcptls.c, /, - main/features.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 08:05 +0000 [r253492] Alec L Davis <sivad.a@paradise.net.nz> - - * 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 17:54 +0000 [r253257-253346] Leif Madsen <lmadsen@digium.com> - - * /, apps/app_userevent.c: Merged revisions 253345 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r253345 | lmadsen | 2010-03-18 12:52:35 -0500 (Thu, 18 Mar 2010) - | 7 lines Change usage of pipe to comma in UserEvent docs. Change - the example usage of pipe as a separator to comma in the - UserEvent documentation. (closes issue #16961) Reported by: - jlpedrosa ........ - - * 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 <twilson@digium.com> - - * 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 00:41 +0000 [r253029-253033] Leif Madsen <lmadsen@digium.com> - - * main/xmldoc.c, /: Merged revisions 253032 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r253032 | - lmadsen | 2010-03-16 19:40:51 -0500 (Tue, 16 Mar 2010) | 1 line - Fix a typo. ........ - - * 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 23:54 +0000 [r252978] Tilghman Lesher <tlesher@digium.com> - - * 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 19:38 +0000 [r252850] Sean Bright <sean@malleable.com> - - * res/res_clialiases.c, /: Merged revisions 252848 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r252848 | seanbright | 2010-03-16 15:36:24 -0400 (Tue, 16 Mar - 2010) | 10 lines Include an extra newline after "Aliased CLI - command" to get back the prompt. The other issue mentioned in - this bug will be more difficult to resolve since we have no idea - (right now) of knowing if the command that is aliased has been - installed yet. (issue #16978) Reported by: jw-asterisk Tested by: - seanbright ........ - -2010-03-16 19:02 +0000 [r252770] Russell Bryant <russell@digium.com> - - * 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 [r252763] Leif Madsen <lmadsen@digium.com> - - * 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 [r252626] Sean Bright <sean@malleable.com> - - * /, 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 [r252622] Tilghman Lesher <tlesher@digium.com> - - * 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 [r252535] Leif Madsen <lmadsen@digium.com> - - * 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 05:04 +0000 [r252365-252444] Tilghman Lesher <tlesher@digium.com> - - * /, 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 17:48 +0000 [r252317] Sean Bright <sean@malleable.com> - - * 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 00:32 +0000 [r252137-252178] Terry Wilson <twilson@digium.com> - - * main/rtp.c: Remove unusued field - - * configs/sip.conf.sample, 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: 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 22:05 +0000 [r252090] Moises Silva <moises.silva@gmail.com> - - * channels/chan_dahdi.c, /: Merged revisions 252088 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r252088 | moy | 2010-03-12 16:57:40 -0500 (Fri, 12 Mar 2010) | 1 - line add missing mfcr2_skip_category setting ........ - -2010-03-12 19:50 +0000 [r251994] Tilghman Lesher <tlesher@digium.com> - - * 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 19:49 +0000 [r251991] Richard Mudgett <rmudgett@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 251946 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r251946 | rmudgett | 2010-03-12 13:05:40 -0600 (Fri, 12 Mar 2010) - | 1 line Doxegen this chan_dahdi lock. ........ - -2010-03-11 21:08 +0000 [r251879-251887] Tilghman Lesher <tlesher@digium.com> - - * 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 ........ - -2010-03-11 Leif Madsen <lmadsen@digium.com> - - * Asterisk 1.6.2.6 released - -2010-03-05 Leif Madsen <lmadsen@digium.com> - - * Asterisk 1.6.2.6-rc2 released - -2010-03-05 Tilghman Lesher <tlesher@digium.com> - - * /, 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 <lmadsen@digium.com> - - * Asterisk 1.6.2.6-rc1 released - -2010-03-03 21:24 +0000 [r250610] Leif Madsen <lmadsen@digium.com> - - * 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 19:13 +0000 [r250484] Jeff Peeler <jpeeler@digium.com> - - * 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 [r250253-250396] David Vossel <dvossel@digium.com> - - * 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 21:10 +0000 [r249953-250052] Leif Madsen <lmadsen@digium.com> - - * 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 - ........ - - * apps/app_directory.c, /: Merged revisions 249950 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r249950 | lmadsen | 2010-03-02 14:49:48 -0500 (Tue, 02 Mar 2010) - | 4 lines Fix literal values wrapped in documentation. (closes - issue #16145) Reported by: tilghman ........ - -2010-03-02 19:50 +0000 [r249952] Alec L Davis <sivad.a@paradise.net.nz> - - * UPGRADE-1.6.txt, 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 19:26 +0000 [r249916-249933] Leif Madsen <lmadsen@digium.com> - - * /, main/features.c: Merged revisions 249925 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r249925 | - lmadsen | 2010-03-02 14:24:43 -0500 (Tue, 02 Mar 2010) | 6 lines - Add missing description of the PARKINGLOT variable in XML - documentation. (closes issue #16743) Reported by: snuffy Patches: - parkingdoc.diff uploaded by snuffy (license 35) ........ - - * /, pbx/pbx_dundi.c: Merged revisions 249912 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r249912 | - lmadsen | 2010-03-02 14:21:19 -0500 (Tue, 02 Mar 2010) | 6 lines - Convert some DUNDI functions to XML documentation. (closes issue - #16798) Reported by: snuffy Patches: xml_dundi.diff uploaded by - snuffy (license 35) ........ - -2010-03-02 19:12 +0000 [r249895] David Vossel <dvossel@digium.com> - - * channels/chan_console.c, channels/chan_gtalk.c, - 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: 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 19:09 +0000 [r249894] Leif Madsen <lmadsen@digium.com> - - * /, apps/app_confbridge.c: Merged revisions 249892 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r249892 | lmadsen | 2010-03-02 14:02:56 -0500 (Tue, 02 Mar 2010) - | 1 line Fix several XML documentation validate errors. ........ - -2010-03-02 09:05 +0000 [r249844] Alec L Davis <sivad.a@paradise.net.nz> - - * 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 19:40 +0000 [r249675] Sean Bright <sean@malleable.com> - - * 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 18:47 +0000 [r249625] Tilghman Lesher <tlesher@digium.com> - - * 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 17:25 +0000 [r249580] Jeff Peeler <jpeeler@digium.com> - - * 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 [r249407-249493] Tilghman Lesher <tlesher@digium.com> - - * 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 23:04 +0000 [r249321] Alec L Davis <sivad.a@paradise.net.nz> - - * 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 14:10 +0000 [r249238] Kevin P. Fleming <kpfleming@digium.com> - - * 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 18:49 +0000 [r249190] Tilghman Lesher <tlesher@digium.com> - - * 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 17:06 +0000 [r249104] Mark Michelson <mmichelson@digium.com> - - * /, 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 23:12 +0000 [r248955] Jeff Peeler <jpeeler@digium.com> - - * 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 [r248949] Mark Michelson <mmichelson@digium.com> - - * /, 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 21:25 +0000 [r248864] Tilghman Lesher <tlesher@digium.com> - - * 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 [r248797] Jeff Peeler <jpeeler@digium.com> - - * 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 [r248643] Tilghman Lesher <tlesher@digium.com> - - * /, 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 16:37 +0000 [r248398] David Vossel <dvossel@digium.com> - - * /, 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 19:07 +0000 [r248011] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_console.c, main/loader.c, /: Merged revisions - 228798 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ 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 19:00 +0000 [r248005] Moises Silva <moises.silva@gmail.com> - - * channels/chan_dahdi.c, /: Merged revisions 248003 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r248003 | moy | 2010-02-19 13:38:34 -0500 (Fri, 19 Feb 2010) | 1 - line mfcr2 issue 0016844 - Fix portability bit fields and make - mfcr2_immediate_accept work again, reported and patched by - korihor ........ - -2010-02-19 18:45 +0000 [r248004] Richard Mudgett <rmudgett@digium.com> - - * 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 - ................ ................ - -2010-02-19 17:41 +0000 [r247916] David Vossel <dvossel@digium.com> - - * /, channels/chan_sip.c: Merged revisions 247915 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r247915 | - dvossel | 2010-02-19 11:40:26 -0600 (Fri, 19 Feb 2010) | 7 lines - handle_request_invite revise comment, fix coding guideline issues - I'm working with this code right now trying to analyze a - deadlock. This change is just to clean up a few things before I - make a more complex patch. ........ - -2010-02-18 23:15 +0000 [r247792-247845] Tilghman Lesher <tlesher@digium.com> - - * 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 21:25 +0000 [r247737-247776] David Vossel <dvossel@digium.com> - - * /, bridges/bridge_softmix.c: Merged revisions 247770 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r247770 | dvossel | 2010-02-18 15:23:48 -0600 (Thu, 18 Feb 2010) - | 9 lines fixes confbridge crash when no timing module is loaded. - (closes issue #16471) Reported by: kjotte Patches: M16471.diff - uploaded by junky (license 177) Tested by: kjotte, junky ........ - - * apps/app_queue.c, /: Merged revisions 247736 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r247736 | - dvossel | 2010-02-18 14:58:41 -0600 (Thu, 18 Feb 2010) | 7 lines - fixes Queue with C option crash (closes issue #16475) Reported - by: okrief Patches: queue_crash.diff uploaded by dvossel (license - 671) ........ - -2010-02-18 19:41 +0000 [r247653] Matthew Nicholson <mnicholson@digium.com> - - * /, 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 16:58 +0000 [r247506-247512] Leif Madsen <lmadsen@digium.com> - - * 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 [r247426] Russell Bryant <russell@digium.com> - - * 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 21:32 +0000 [r246989-247337] Mark Michelson <mmichelson@digium.com> - - * include/asterisk/strings.h, main/strings.c, /: 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/strings.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 21:03 +0000 [r246900-246982] David Vossel <dvossel@digium.com> - - * 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: fixes merge error with Monitor calculation fix - - * 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 23:45 +0000 [r246713] Tilghman Lesher <tlesher@digium.com> - - * 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 23:33 +0000 [r246547] David Vossel <dvossel@digium.com> - - * 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 [r246464] Jason Parker <jparker@digium.com> - - * main/channel.c: Fix some silly formatting that made my head hurt. - -2010-02-10 21:28 +0000 [r246199-246207] Tilghman Lesher <tlesher@digium.com> - - * /, 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. ........ - - * /, configure, include/asterisk/autoconfig.h.in, configure.ac, - res/res_agi.c: Merged revisions 246030 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r246030 | - tilghman | 2010-02-10 10:01:28 -0600 (Wed, 10 Feb 2010) | 12 - lines Solaris doesn't like outputting a NULL to a %s in format - strings. Detect all platforms that don't like that, either, and - ensure that when documentation is missing, we pass a non-NULL - pointer when outputting the corresponding documentation. (closes - issue #16689) Reported by: bklang Patches: - 20100209__issue16689__with_tests.diff.txt uploaded by tilghman - (license 14) Review: https://reviewboard.asterisk.org/r/497/ - ........ - -2010-02-10 17:51 +0000 [r246117] David Vossel <dvossel@digium.com> - - * 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 16:58 +0000 [r246073] Jeff Peeler <jpeeler@digium.com> - - * 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 [r245948-246025] Tilghman Lesher <tlesher@digium.com> - - * /, 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 23:11 +0000 [r245794] David Vossel <dvossel@digium.com> - - * 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 [r245732] Tilghman Lesher <tlesher@digium.com> - - * /, 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 17:43 +0000 [r245728] Matthew Nicholson <mnicholson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 245727 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r245727 | - mnicholson | 2010-02-09 11:40:04 -0600 (Tue, 09 Feb 2010) | 2 - lines This commit removes an extra newline in T.38 generated SDP - packets. This bug was caused by the fix introduced in r243860. - (closes issue #16766) Reported by: raivisr Patches: - t38-sdp-newline-fix1.diff uploaded by mnicholson (license 96) - Tested by: raivisr ........ - -2010-02-09 16:26 +0000 [r245683] Kevin P. Fleming <kpfleming@digium.com> - - * /, 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 23:47 +0000 [r245626] Russell Bryant <russell@digium.com> - - * /, 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 [r245581] Tilghman Lesher <tlesher@digium.com> - - * 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 20:43 +0000 [r245500] Jason Parker <jparker@digium.com> - - * 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 19:27 +0000 [r245097] Jeff Peeler <jpeeler@digium.com> - - * 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 [r244930] Sean Bright <sean@malleable.com> - - * 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 19:28 +0000 [r244555] Mark Michelson <mmichelson@digium.com> - - * 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 18:47 +0000 [r244508] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_dahdi.c, /, main/ast_expr2f.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 22:29 +0000 [r244445] David Vossel <dvossel@digium.com> - - * 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 [r244395] Tilghman Lesher <tlesher@digium.com> - - * 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 <lmadsen@digium.com> - - * Release Asterisk 1.6.2.2 - - * 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 <lmadsen@digium.com> - - * Release Asterisk 1.6.2.1 - -2010-01-08 Leif Madsen <lmadsen@digium.com> - - * Release Asterisk 1.6.2.1-rc1 - -2010-01-07 21:17 +0000 [r238499] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_console.c, channels/chan_oss.c, main/poll.c, - channels/chan_usbradio.c, include/asterisk/utils.h, /, - channels/chan_sip.c, channels/chan_alsa.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 20:17 +0000 [r238362-238416] David Vossel <dvossel@digium.com> - - * 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 09:49 +0000 [r238349] Tzafrir Cohen <tzafrir.cohen@xorcom.com> - - * 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 [r238234] Tilghman Lesher <tlesher@digium.com> - - * /, 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 20:40 +0000 [r238137-238185] Jeff Peeler <jpeeler@digium.com> - - * /, 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 15:22 +0000 [r238013] Russell Bryant <russell@digium.com> - - * /, 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 06:54 +0000 [r237969] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_sip.c: Merged revisions 237968 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r237968 | - tilghman | 2010-01-06 00:53:23 -0600 (Wed, 06 Jan 2010) | 2 lines - Whoa, duplicate setting (dead code). ........ - -2010-01-05 23:10 +0000 [r237924] Kinsey Moore <kmoore@digium.com> - - * apps/app_test.c: Add a wait to ensure TestServer thinks it has - finished sending the final digit. This was previously committed - to 1.4, 1.6.0, 1.6.1, and trunk just after 1.6.2 was created (and - missed). 1.6.2 also needs this patch to resolve the bug. (closes - issue #16550) Reported by: opticron Patches: apptest.diff - uploaded by opticron (license 267) - -2010-01-05 23:09 +0000 [r237840-237921] David Vossel <dvossel@digium.com> - - * 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 17:25 +0000 [r237743] Russell Bryant <russell@digium.com> - - * /, 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) - ........ ................ - -2010-01-05 16:09 +0000 [r237657] Michiel van Baak <michiel@vanbaak.info> - - * apps/app_mixmonitor.c, /: Merged revisions 237656 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r237656 | mvanbaak | 2010-01-05 17:08:12 +0100 (Tue, 05 Jan 2010) - | 6 lines Make CLI command 'mixmonitor start|stop <channel> work - again. (closes issue #16534) Reported by: jlaguilar Fix as - suggested by jlaguilar in the bugreport ........ - -2010-01-04 21:52 +0000 [r237409-237577] Tilghman Lesher <tlesher@digium.com> - - * /, 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 16:50 +0000 [r237328] David Vossel <dvossel@digium.com> - - * 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 16:27 +0000 [r237326] Jeff Peeler <jpeeler@digium.com> - - * /, 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 [r237322] Tilghman Lesher <tlesher@digium.com> - - * 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 10:03 +0000 [r237139] Olle Johansson <oej@edvina.net> - - * /, 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 [r236985] Tilghman Lesher <tlesher@digium.com> - - * 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 [r236899-236905] Jeff Peeler <jpeeler@digium.com> - - * /, 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. ........ - - * main/cli.c, /: Merged revisions 236893 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r236893 | - jpeeler | 2009-12-30 14:34:41 -0600 (Wed, 30 Dec 2009) | 11 lines - Fix compiling with LOW_MEMORY. Modified handle_verbose to be - LOW_MEMORY aware. (closes issue #16381) Reported by: - michael_iedema Patches: ast_complete_source_filename.patch - uploaded by michael iedema (license 942) modified by me ........ - -2009-12-30 17:56 +0000 [r236804-236850] Tilghman Lesher <tlesher@digium.com> - - * /, 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 22:13 +0000 [r236716] Jason Parker <jparker@digium.com> - - * 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 [r236670] Tilghman Lesher <tlesher@digium.com> - - * 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 [r236513-236635] Sean Bright <sean@malleable.com> - - * 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 [r236437] Tilghman Lesher <tlesher@digium.com> - - * 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 15:32 +0000 [r236361] Kevin P. Fleming <kpfleming@digium.com> - - * 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 18:27 +0000 [r236189-236303] Tilghman Lesher <tlesher@digium.com> - - * 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 17:04 +0000 [r236064] David Vossel <dvossel@digium.com> - - * /, 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 19:58 +0000 [r235944] Jeff Peeler <jpeeler@digium.com> - - * 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 [r235826] Tilghman Lesher <tlesher@digium.com> - - * /, 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 08:58 +0000 [r235775] Alec L Davis <sivad.a@paradise.net.nz> - - * 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 23:04 +0000 [r235665] Jeff Peeler <jpeeler@digium.com> - - * 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 [r235576-235659] Tilghman Lesher <tlesher@digium.com> - - * /, 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 23:22 +0000 [r235522] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 235521 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r235521 | - file | 2009-12-17 19:21:07 -0400 (Thu, 17 Dec 2009) | 3 lines - Remove some old code for going to the 'fax' extension when a T.38 - switchover occurs. This would have already happened when we - detected the CNG tone so this was basically a noop. ........ - -2009-12-17 Leif Madsen <lmadsen@digium.com> - - * Release Asterisk 1.6.2.0 - -2009-12-09 Leif Madsen <lmadsen@digium.com> - - * Release Asterisk 1.6.2.0-rc8 - -2009-12-08 18:33 +0000 [r233731] Tilghman Lesher <tlesher@digium.com> - - * 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 18:04 +0000 [r233694] Russell Bryant <russell@digium.com> - - * formats/format_sln16.c, formats/format_wav_gsm.c, - formats/format_siren7.c, formats/format_ilbc.c, - formats/format_vox.c, formats/format_pcm.c, - formats/format_h263.c, formats/format_g723.c, - formats/format_h264.c, formats/format_siren14.c, - formats/format_jpeg.c, formats/format_g726.c, - formats/format_gsm.c, formats/format_g729.c, /, - formats/format_sln.c, formats/format_wav.c, - formats/format_ogg_vorbis.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 07:41 +0000 [r233689] TransNexus OSP Development <support@transnexus.com> - - * apps/app_osplookup.c: Fixed compile error with OSP Toolkit 3.6. - -2009-12-07 23:54 +0000 [r233615] Atis Lezdins <atis@iq-labs.net> - - * 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 [r233473-233612] David Vossel <dvossel@digium.com> - - * /, 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 [r233396] Matthew Nicholson <mnicholson@digium.com> - - * /, 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 [r233281] David Vossel <dvossel@digium.com> - - * 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 21:07 +0000 [r233240] Matthias Nick <mnick@digium.com> - - * 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 17:36 +0000 [r233165] David Vossel <dvossel@digium.com> - - * 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 17:23 +0000 [r233130] Russell Bryant <russell@digium.com> - - * 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 15:57 +0000 [r233049] Matthias Nick <mnick@digium.com> - - * 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 [r232866] Tilghman Lesher <tlesher@digium.com> - - * 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 15:14 +0000 [r232813] David Ruggles <thedavidfactor@gmail.com> - - * 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 00:20 +0000 [r232675-232678] Tilghman Lesher <tlesher@digium.com> - - * res/res_musiconhold.c: Oops, really remove it this time - - * 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 23:28 +0000 [r232658] David Vossel <dvossel@digium.com> - - * CHANGES, /, UPGRADE.txt: Merged revisions 232657 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r232657 | dvossel | 2009-12-02 17:27:45 -0600 (Wed, 02 Dec 2009) - | 6 lines update CHANGES and UPGRADE.txt for early media behavior - change between 1.6.1 and 1.6.2 (closes issue #16212) Reported by: - miki ........ - -2009-12-02 22:05 +0000 [r232579-232585] Jeff Peeler <jpeeler@digium.com> - - * 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 17:11 +0000 [r232359] Joshua Colp <jcolp@digium.com> - - * /, 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 17:01 +0000 [r232352] David Vossel <dvossel@digium.com> - - * /, 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 16:43 +0000 [r232348] Joshua Colp <jcolp@digium.com> - - * /, 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 [r232270] David Vossel <dvossel@digium.com> - - * 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 [r232232] Joshua Colp <jcolp@digium.com> - - * /, 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 00:52 +0000 [r232094] Jeff Peeler <jpeeler@digium.com> - - * 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 23:40 +0000 [r232011-232015] Russell Bryant <russell@digium.com> - - * /, 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 22:03 +0000 [r231930] Jeff Peeler <jpeeler@digium.com> - - * 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 [r231870] David Vossel <dvossel@digium.com> - - * 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 Leif Madsen <lmadsen@digium.com> - - * Release Asterisk 1.6.2.0-rc7 - -2009-12-01 15:48 +0000 [r231743] Matthew Nicholson <mnicholson@digium.com> - - * /, 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 21:59 +0000 [r231695-231696] Kevin P. Fleming <kpfleming@digium.com> - - * 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. ........ - - * pbx/pbx_config.c: Backport a tiny fix from trunk that makes GCC - 4.4.x happier. - -2009-11-30 21:36 +0000 [r231689] Matthew Nicholson <mnicholson@digium.com> - - * 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 20:47 +0000 [r231605] Joshua Colp <jcolp@digium.com> - - * /, 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 [r231505-231558] David Vossel <dvossel@digium.com> - - * 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 [r231302] Tilghman Lesher <tlesher@digium.com> - - * 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 15:48 +0000 [r231191] Matthew Nicholson <mnicholson@digium.com> - - * /, 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 20:36 +0000 [r231136] Tilghman Lesher <tlesher@digium.com> - - * 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 [r231098] Jeff Peeler <jpeeler@digium.com> - - * /, 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 15:48 +0000 [r230884] Joshua Colp <jcolp@digium.com> - - * configs/sip.conf.sample, /, channels/chan_sip.c: 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 15:38 +0000 [r230796-230880] Kevin P. Fleming <kpfleming@digium.com> - - * /, 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 22:36 +0000 [r230727] David Vossel <dvossel@digium.com> - - * 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 21:07 +0000 [r230629] Matthew Nicholson <mnicholson@digium.com> - - * /, 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 17:31 +0000 [r230510-230585] David Vossel <dvossel@digium.com> - - * main/audiohook.c, /, include/asterisk/audiohook.h: 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/ ........ ................ - -2009-11-16 16:41 +0000 [r230250-230384] Kevin P. Fleming <kpfleming@digium.com> - - * /, apps/app_fax.c: Merged revisions 230381 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r230381 | - kpfleming | 2009-11-16 10:40:25 -0600 (Mon, 16 Nov 2009) | 1 line - Fix another buglet in T.38 session teardown at the end of FAX - sessions. ........ - - * /, apps/app_fax.c: Merged revisions 230343 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r230343 | - kpfleming | 2009-11-16 06:51:59 -0600 (Mon, 16 Nov 2009) | 2 - lines Ensure that only one end of a T.38 session initiates - teardown at completion. ........ - - * channels/chan_iax2.c, /: Merged revisions 230247 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r230247 | kpfleming | 2009-11-15 11:23:02 -0600 - (Sun, 15 Nov 2009) | 12 lines Merged revisions 230246 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r230246 | kpfleming | 2009-11-15 11:19:06 -0600 (Sun, 15 Nov - 2009) | 6 lines Correct mistaken option name in error message. - The configuration option for allowing hosts to make - non-token-based calls is 'calltokenoptional', not - 'calltokenignore'. (reported on asterisk-users) ........ - ................ - -2009-11-13 22:01 +0000 [r229969-230148] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 230145 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r230145 | file | 2009-11-13 16:00:44 -0600 (Fri, 13 Nov 2009) | - 15 lines Merged revisions 230144 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r230144 | file | 2009-11-13 16:00:19 -0600 (Fri, 13 Nov 2009) | 8 - lines Respect the maddr parameter in the Via header. (closes - issue #14446) Reported by: frawd Patches: via_maddr.patch - uploaded by frawd (license 610) Tested by: frawd ........ - ................ - - * channels/chan_local.c, /: Merged revisions 230039 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r230039 | file | 2009-11-13 13:44:53 -0600 (Fri, - 13 Nov 2009) | 16 lines Merged revisions 230038 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r230038 | file | 2009-11-13 13:44:07 -0600 (Fri, 13 Nov 2009) | 9 - lines Fix a crash caused by two threads thinking they should both - free the chan_local private structure when only one should. - (closes issue #15314) Reported by: sroberts Patches: - Issue15314_Move_Nulling_owner.patch uploaded by davidw (license - 780) Tested by: davidw, lottc ........ ................ - - * configs/extensions.conf.sample, /, apps/app_chanisavail.c: Merged - revisions 229966 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r229966 | file | 2009-11-13 11:20:26 -0600 (Fri, 13 Nov 2009) | - 13 lines Merged revisions 229965 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r229965 | file | 2009-11-13 11:19:59 -0600 (Fri, 13 Nov 2009) | 6 - lines Document a limitation in the AVAILSTATUS variable from - ChanIsAvail and provide a workaround for it that does not change - existing behavior. (closes issue #14426) Reported by: macli - ........ ................ - -2009-11-13 Leif Madsen <lmadsen@digium.com> - - * Release Asterisk 1.6.2.0-rc6 - -2009-11-13 15:57 +0000 [r229915] Joshua Colp <jcolp@digium.com> - - * /, 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 [r229752] Jason Parker <jparker@digium.com> - - * 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 16:47 +0000 [r229671] David Vossel <dvossel@digium.com> - - * 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 20:49 +0000 [r229570] David Ruggles <thedavidfactor@gmail.com> - - * 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 19:56 +0000 [r229492-229502] David Brooks <dbrooks@digium.com> - - * 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 [r229366] Tilghman Lesher <tlesher@digium.com> - - * 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 [r229359] David Ruggles <thedavidfactor@gmail.com> - - * 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 21:33 +0000 [r229354] Tilghman Lesher <tlesher@digium.com> - - * 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 [r229285] Joshua Colp <jcolp@digium.com> - - * /, 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 17:52 +0000 [r229232] David Vossel <dvossel@digium.com> - - * 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 17:39 +0000 [r229231] David Ruggles <thedavidfactor@gmail.com> - - * 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 15:47 +0000 [r229101] Matthew Nicholson <mnicholson@digium.com> - - * UPGRADE-1.6.txt, main/editline/makelist.in, UPGRADE.txt: Reset - props that were accidently deleted in 229088. - -2009-11-10 15:28 +0000 [r229094] David Vossel <dvossel@digium.com> - - * 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 15:16 +0000 [r229088] Matthew Nicholson <mnicholson@digium.com> - - * UPGRADE-1.6.txt, main/editline/makelist.in, channels/chan_sip.c, - UPGRADE.txt: Reverted revision 202007. (closes issue #16175) - Reported by: paul-tg Tested by: paul-tg - -2009-11-10 11:25 +0000 [r229078] Gavin Henry <ghenry@suretecsystems.com> - - * 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 22:59 +0000 [r229017] Terry Wilson <twilson@digium.com> - - * 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 22:17 +0000 [r229012] David Vossel <dvossel@digium.com> - - * 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 #15848) 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 <lmadsen@digium.com> - - * Release Asterisk 1.6.2.0-rc5 - -2009-11-09 15:40 +0000 [r228900] Leif Madsen <lmadsen@digium.com> - - * 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 14:48 +0000 [r228859] Matthew Nicholson <mnicholson@digium.com> - - * /, 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 [r228694] David Vossel <dvossel@digium.com> - - * 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 20:26 +0000 [r228649] Matthew Nicholson <mnicholson@digium.com> - - * 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 18:43 +0000 [r228551] Joshua Colp <jcolp@digium.com> - - * /, 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 <lmadsen@digium.com> - - * Release Asterisk 1.6.2.0-rc4 - -2009-11-06 17:54 +0000 [r228504] Joshua Colp <jcolp@digium.com> - - * 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 [r228421-228447] David Vossel <dvossel@digium.com> - - * 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 [r228413] Joshua Colp <jcolp@digium.com> - - * /, 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 15:43 +0000 [r228269-228340] David Vossel <dvossel@digium.com> - - * /, 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 ........ - -2009-11-05 22:13 +0000 [r228198] Tilghman Lesher <tlesher@digium.com> - - * /, 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) ........ - -2009-11-05 21:27 +0000 [r228195] Jeff Peeler <jpeeler@digium.com> - - * 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 21:27 +0000 [r228194] Tilghman Lesher <tlesher@digium.com> - - * /, 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 19:42 +0000 [r228148] David Brooks <dbrooks@digium.com> - - * 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 19:20 +0000 [r228093] Jason Parker <jparker@digium.com> - - * 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 17:14 +0000 [r228017] Tilghman Lesher <tlesher@digium.com> - - * 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 23:53 +0000 [r227947] Jeff Peeler <jpeeler@digium.com> - - * 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 21:09 +0000 [r227760-227831] Matthew Nicholson <mnicholson@digium.com> - - * 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 19:28 +0000 [r227733-227748] Joshua Colp <jcolp@digium.com> - - * /, 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 [r227375] Jason Parker <jparker@digium.com> - - * 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 [r227364-227371] Leif Madsen <lmadsen@digium.com> - - * 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 18:15 +0000 [r227280] Richard Mudgett <rmudgett@digium.com> - - * 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 17:14 +0000 [r227239] David Vossel <dvossel@digium.com> - - * /, 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 ........ - -2009-11-03 15:40 +0000 [r227170] Joshua Colp <jcolp@digium.com> - - * /, 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 15:25 +0000 [r227165] Leif Madsen <lmadsen@digium.com> - - * 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 13:51 +0000 [r227156] Olle Johansson <oej@edvina.net> - - * Makefile, /, 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 21:06 +0000 [r226978] David Brooks <dbrooks@digium.com> - - * 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 18:12 +0000 [r226893] Joshua Colp <jcolp@digium.com> - - * 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 [r226815] Tilghman Lesher <tlesher@digium.com> - - * /, 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 18:18 +0000 [r226540] Joshua Colp <jcolp@digium.com> - - * doc/tex/localchannel.tex, channels/chan_local.c, /: 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 21:32 +0000 [r226486] Tzafrir Cohen <tzafrir.cohen@xorcom.com> - - * build_tools/get_documentation, /: remove empty awk pattern (//) - Solaris 10 nawk doesn't like the empty pattern such as '//' for - 'always'. Just remove that. No pattern at all always matches. - Merged revisions 226453 via svnmerge from - http://svn.digium.com/svn/asterisk/trunk - -2009-10-28 20:13 +0000 [r226379-226385] Leif Madsen <lmadsen@digium.com> - - * 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 18:06 +0000 [r226170-226308] Tilghman Lesher <tlesher@digium.com> - - * /, 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 17:12 +0000 [r226101] Terry Wilson <twilson@digium.com> - - * 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 ........ - -2009-10-27 00:16 +0000 [r226055] Tzafrir Cohen <tzafrir.cohen@xorcom.com> - - * /, 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 19:42 +0000 [r225914] Jeff Peeler <jpeeler@digium.com> - - * /, 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 15:56 +0000 [r225871] Kevin P. Fleming <kpfleming@digium.com> - - * 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. (closes issue #16127) - -2009-10-23 14:46 +0000 [r225651] David Vossel <dvossel@digium.com> - - * /, 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 14:08 +0000 [r225585] Kevin P. Fleming <kpfleming@digium.com> - - * 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 22:24 +0000 [r225516] Leif Madsen <lmadsen@digium.com> - - * README, /: Merged revisions 225515 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r225515 | - lmadsen | 2009-10-22 17:24:03 -0500 (Thu, 22 Oct 2009) | 8 lines - Update README documentation. Update the README documentation to - correctly describe which CLI command you should use when - attempting to get help from the CLI. (closes issue #16064) - Reported by: thedavidfactor Patches: readme.patch uploaded by - thedavidfactor (license 903) ........ - -2009-10-22 21:55 +0000 [r225489] David Vossel <dvossel@digium.com> - - * apps/app_externalivr.c, include/asterisk/tcptls.h, main/tcptls.c, - /, channels/chan_sip.c: 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 [r225488] Leif Madsen <lmadsen@digium.com> - - * 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 [r225363] Tilghman Lesher <tlesher@digium.com> - - * 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 22:00 +0000 [r225035-225308] David Vossel <dvossel@digium.com> - - * 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/ ........ ................ - - * configs/sip.conf.sample, channels/chan_iax2.c, - configs/iax.conf.sample, /, channels/chan_sip.c: 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/ ........ ................ - -2009-10-20 22:11 +0000 [r224859] Tilghman Lesher <tlesher@digium.com> - - * main/audiohook.c, funcs/func_speex.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 17:50 +0000 [r224777] Joshua Colp <jcolp@digium.com> - - * /, 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 ........ ................ - -2009-10-20 00:00 +0000 [r224674] Kevin P. Fleming <kpfleming@digium.com> - - * 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 19:54 +0000 [r224571] Joshua Colp <jcolp@digium.com> - - * 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 ........ ................ - -2009-10-19 19:41 +0000 [r224563] Kevin P. Fleming <kpfleming@digium.com> - - * formats/format_siren14.c, /: Merged revisions 224562 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r224562 | kpfleming | 2009-10-19 14:40:26 -0500 (Mon, 19 Oct - 2009) | 1 line Remove useless debugging message. ........ - -2009-10-19 00:13 +0000 [r224447-224451] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 224448 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r224448 | tilghman | 2009-10-18 19:05:56 -0500 (Sun, 18 Oct 2009) - | 3 lines Allow ODBC storage to be queried with multiple - mailboxes, and remove multiple goto's. This corrects an issue - reported on the -users list. ........ - - * configs/res_odbc.conf.sample, /: Merged revisions 224446 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r224446 | tilghman | 2009-10-18 18:41:30 -0500 (Sun, 18 - Oct 2009) | 2 lines Clarify that "forcecommit" is NOT an alias - for "autocommit", but instead controls the default disposition of - uncommitted transactions. ........ - -2009-10-17 01:58 +0000 [r224334] Jeff Peeler <jpeeler@digium.com> - - * 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 20:58 +0000 [r224264] Richard Mudgett <rmudgett@digium.com> - - * 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 [r224181] Jeff Peeler <jpeeler@digium.com> - - * 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 [r223835] Jeff Peeler <jpeeler@digium.com> - - * 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 21:01 +0000 [r223757] David Vossel <dvossel@digium.com> - - * 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 14:37 +0000 [r223655] Kevin P. Fleming <kpfleming@digium.com> - - * /, 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 17:32 +0000 [r223490] Russell Bryant <russell@digium.com> - - * 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 23:12 +0000 [r223406] Jeff Peeler <jpeeler@digium.com> - - * 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 21:01 +0000 [r223333] Kevin P. Fleming <kpfleming@digium.com> - - * /, 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 18:53 +0000 [r223286] Matthew Nicholson <mnicholson@digium.com> - - * 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 18:29 +0000 [r223257] Mark Michelson <mmichelson@digium.com> - - * 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 17:55 +0000 [r223208] David Vossel <dvossel@digium.com> - - * /, 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 [r223173] Matthew Nicholson <mnicholson@digium.com> - - * 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 17:09 +0000 [r223089-223133] David Vossel <dvossel@digium.com> - - * /, 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 20:00 +0000 [r222883] Russell Bryant <russell@digium.com> - - * 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 19:41 +0000 [r222874] David Vossel <dvossel@digium.com> - - * 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 16:51 +0000 [r222695-222802] Richard Mudgett <rmudgett@digium.com> - - * 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 18:06 +0000 [r222549] Jason Parker <jparker@digium.com> - - * /, configs/queues.conf.sample: Merged revisions 222548 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r222548 | qwell | 2009-10-07 13:04:56 -0500 (Wed, 07 Oct - 2009) | 5 lines Remove 'keepstats' queue option from sample - config, as it's no longer used. - https://reviewboard.asterisk.org/r/115/ (closes issue #15820) - Reported by: kshumard ........ - -2009-10-07 18:00 +0000 [r222547] Sean Bright <sean@malleable.com> - - * funcs/func_strings.c: Fix merge error. - -2009-10-07 17:45 +0000 [r222544] David Vossel <dvossel@digium.com> - - * /, 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 23:59 +0000 [r222354-222466] Jeff Peeler <jpeeler@digium.com> - - * 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, and 1.6.2) (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) - - * 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 19:41 +0000 [r222311] Tilghman Lesher <tlesher@digium.com> - - * cdr/cdr_pgsql.c, res/res_config_pgsql.c, /: Merged 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 19:27 +0000 [r222304] Jeff Peeler <jpeeler@digium.com> - - * 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 19:22 +0000 [r222289] Tilghman Lesher <tlesher@digium.com> - - * 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 ........ - -2009-10-06 Leif Madsen <lmadsen@digium.com> - - * Released Asterisk 1.6.2.0-rc3 - -2009-10-06 01:39 +0000 [r222113-222187] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_console.c, res/res_musiconhold.c, apps/app_queue.c, - channels/chan_iax2.c, main/astobj2.c, res/res_odbc.c, - res/res_clialiases.c, /, channels/chan_sip.c, - funcs/func_dialgroup.c, include/asterisk/astobj2.h, - res/res_phoneprov.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/ ........ ................ - - * configs/sip.conf.sample, main/udptl.c, /, channels/chan_sip.c, - configs/udptl.conf.sample, UPGRADE.txt: 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 17:35 +0000 [r222032] David Vossel <dvossel@digium.com> - - * 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 [r221923-221974] Tilghman Lesher <tlesher@digium.com> - - * main/astobj2.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. ........ ................ - - * /, 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 01:35 +0000 [r221879] Richard Mudgett <rmudgett@digium.com> - - * 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 00:07 +0000 [r221744-221780] Tilghman Lesher <tlesher@digium.com> - - * 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 19:35 +0000 [r221698] David Vossel <dvossel@digium.com> - - * /, 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 16:57 +0000 [r221660] Matthew Nicholson <mnicholson@digium.com> - - * /, 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 16:25 +0000 [r221622] Kevin P. Fleming <kpfleming@digium.com> - - * 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 23:07 +0000 [r221477-221485] Matthew Nicholson <mnicholson@digium.com> - - * /, 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 ........ - - * configs/sip.conf.sample, /, channels/chan_sip.c: 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 21:46 +0000 [r221371-221472] Matthias Nick <mnick@digium.com> - - * 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 19:15 +0000 [r221304] Terry Wilson <twilson@digium.com> - - * configs/sip.conf.sample, main/rtp.c, /, channels/chan_sip.c, - include/asterisk/rtp.h: 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 [r221204] Tilghman Lesher <tlesher@digium.com> - - * 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 14:57 +0000 [r221089] Sean Bright <sean@malleable.com> - - * 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 [r221027-221047] Tilghman Lesher <tlesher@digium.com> - - * /, 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 - ........ - - * include/asterisk/smdi.h, include/asterisk/optional_api.h - (removed), apps/app_voicemail.c, include/asterisk/agi.h, - include/asterisk/monitor.h: Remove optional_api from 1.6.2 - branch, since it is not currently working. This is a blocking - issue for the 1.6.2 release. (closes issue #15914) Reported by: - mbeckwell Branch: - http://svn.digium.com/svn/asterisk/team/tilghman/optional_api_162 - Tested by: mbeckwell - - * /, 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 20:24 +0000 [r220905-220934] Matthew Nicholson <mnicholson@digium.com> - - * 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 - - * /, apps/app_confbridge.c: Merged revisions 220904 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r220904 | mnicholson | 2009-09-29 14:49:02 -0500 (Tue, 29 Sep - 2009) | 5 lines Fix options 'm' and 's'. They were swapped in the - code. Also document the fact that app_confbridge does not - automatically answer the channel. (closes issue #15964) Reported - by: shrift ........ - -2009-09-29 17:06 +0000 [r220836] Jeff Peeler <jpeeler@digium.com> - - * 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 [r220725] Sean Bright <sean@malleable.com> - - * /, 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. ........ - ................ - -2009-09-28 19:11 +0000 [r220722] Jeff Peeler <jpeeler@digium.com> - - * /, channels/chan_sip.c: Merged revisions 220718 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r220718 | - jpeeler | 2009-09-28 14:10:10 -0500 (Mon, 28 Sep 2009) | 10 lines - Fix building of registration entry in build_peer when using - callbackextension Check for remotesecret option was - unintentionally always true, which therefore caused the secret - option to never be used. Thanks to dvossel for pointing out the - exact fix. (closes issue #15943) Reported by: tpsast ........ - -2009-09-27 20:45 +0000 [r220632] Michiel van Baak <michiel@vanbaak.info> - - * funcs/func_callerid.c, /: Merged revisions 220629 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r220629 | mvanbaak | 2009-09-27 22:40:16 +0200 (Sun, 27 Sep 2009) - | 3 lines add name argument for the CALLERID dialplan function to - the xml documentation. Pointed out to me on IRC by snuff-home. - Thanks ........ - -2009-09-26 15:12 +0000 [r220589] Tilghman Lesher <tlesher@digium.com> - - * /, 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 [r220369] David Vossel <dvossel@digium.com> - - * 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 [r220292] Tilghman Lesher <tlesher@digium.com> - - * 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 [r220103-220221] Sean Bright <sean@malleable.com> - - * 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 08:43 +0000 [r220031] Michiel van Baak <michiel@vanbaak.info> - - * 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 07:45 +0000 [r219989] Tilghman Lesher <tlesher@digium.com> - - * 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 21:48 +0000 [r219821] Tilghman Lesher <tlesher@digium.com> - - * 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 17:01 +0000 [r219722] David Vossel <dvossel@digium.com> - - * 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 [r219669] Tilghman Lesher <tlesher@digium.com> - - * /, 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 03:14 +0000 [r219590] Russell Bryant <russell@digium.com> - - * 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 23:21 +0000 [r219452-219521] David Vossel <dvossel@digium.com> - - * 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 [r219415] Tilghman Lesher <tlesher@digium.com> - - * 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 22:38 +0000 [r219376] David Vossel <dvossel@digium.com> - - * /, channels/chan_sip.c: Merged revisions 219371 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r219371 | - dvossel | 2009-09-17 17:37:28 -0500 (Thu, 17 Sep 2009) | 9 lines - fixes deadlock when performing directed pickup w Invite/replaces - (closes issue #15340) Reported by: lmsteffan Patches: - deadlock.patch uploaded by lmsteffan (license 779) Tested by: - lmsteffan ........ - -2009-09-17 22:37 +0000 [r219370] Joshua Colp <jcolp@digium.com> - - * /, 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 22:06 +0000 [r219307] David Vossel <dvossel@digium.com> - - * /, 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 [r219267] Joshua Colp <jcolp@digium.com> - - * /, 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 <lmadsen@digium.com> - - * Released Asterisk 1.6.2.0-rc2 - -2009-09-17 15:38 +0000 [r219194] Matthew Nicholson <mnicholson@digium.com> - - * 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 [r219063] Tilghman Lesher <tlesher@digium.com> - - * 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 19:40 +0000 [r218938] David Brooks <dbrooks@digium.com> - - * 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 19:29 +0000 [r218937] Mark Michelson <mmichelson@digium.com> - - * /, 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 19:25 +0000 [r218934] Joshua Colp <jcolp@digium.com> - - * /, 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 13:38 +0000 [r218802] Russell Bryant <russell@digium.com> - - * 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 [r218733] Tilghman Lesher <tlesher@digium.com> - - * 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) - ........ ................ - -2009-09-15 19:24 +0000 [r218688] David Vossel <dvossel@digium.com> - - * /, 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 [r218590] Matthew Nicholson <mnicholson@digium.com> - - * /, 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 16:06 +0000 [r218582] Tilghman Lesher <tlesher@digium.com> - - * 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 15:59 +0000 [r218576] Jeff Peeler <jpeeler@digium.com> - - * 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 15:42 +0000 [r218507-218575] Mark Michelson <mmichelson@digium.com> - - * /, 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. ........ - - * /, 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 15:05 +0000 [r218503] Kevin P. Fleming <kpfleming@digium.com> - - * 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 [r218364] Tilghman Lesher <tlesher@digium.com> - - * 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 18:18 +0000 [r218300] Joshua Colp <jcolp@digium.com> - - * /, 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 15:20 +0000 [r218238] Matthew Nicholson <mnicholson@digium.com> - - * /, 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 22:12 +0000 [r218219] Tzafrir Cohen <tzafrir.cohen@xorcom.com> - - * 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 - -2009-09-13 05:59 +0000 [r218151] Moises Silva <moises.silva@gmail.com> - - * channels/chan_dahdi.c, /: Merged revisions 218150 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r218150 | moy | 2009-09-13 01:51:46 -0400 (Sun, 13 Sep 2009) | 1 - line get rid of mfcr2 monitor thread condition, is problematic - ........ - -2009-09-11 06:00 +0000 [r217926-218055] Tilghman Lesher <tlesher@digium.com> - - * 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 21:21 +0000 [r217821] David Vossel <dvossel@digium.com> - - * 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 19:56 +0000 [r217739] mnick <mnick@localhost>: - - * 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 18:40 +0000 [r217665] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 216805 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r216805 | - oej | 2009-09-07 18:08:08 +0200 (MÃ¥n, 07 Sep 2009) | 2 lines - Since it's possible to have more than 999 calls, I'm changing the - call counter roof to something higher. ........ - -2009-09-10 18:19 +0000 [r217647] Tilghman Lesher <tlesher@digium.com> - - * 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 15:14 +0000 [r217632] Moises Silva <moises.silva@gmail.com> - - * channels/chan_dahdi.c, /: Merged revisions 217524 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r217524 | moy | 2009-09-09 17:48:04 -0400 (Wed, 09 Sep 2009) | 1 - line ast_log replaced for ast_verbose in MFCR2 event - notifications ........ - -2009-09-10 12:09 +0000 [r217594] Olle Johansson <oej@edvina.net> - - * /, 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 20:37 +0000 [r217519] Tzafrir Cohen <tzafrir.cohen@xorcom.com> - - * /, 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 - https://origsvn.digium.com/svn/asterisk/trunk - -2009-09-09 10:58 +0000 [r217369] Olle Johansson <oej@edvina.net> - - * /, 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 [r217299] Sean Bright <sean@malleable.com> - - * /, 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 20:33 +0000 [r217217] Tilghman Lesher <tlesher@digium.com> - - * /, 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 [r217077] Kevin P. Fleming <kpfleming@digium.com> - - * /, 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 [r217036] Tilghman Lesher <tlesher@digium.com> - - * /, 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 [r216994] David Vossel <dvossel@digium.com> - - * /, 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 16:43 +0000 [r216647-216845] Olle Johansson <oej@edvina.net> - - * /, 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 in the INVITE, set it to default value - (not unset minimum = -1) Patch by oej closes issue #15621 - Reported by: fnordian Tested by: atis ........ - - * CHANGES, UPGRADE.txt: Add docs - - * configs/sip.conf.sample, apps/app_playback.c, main/pbx.c, /, - channels/chan_sip.c, apps/app_disa.c: 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 19:42 +0000 [r216598] David Vossel <dvossel@digium.com> - - * /, 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 [r216597] Sean Bright <sean@malleable.com> - - * 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 [r216550-216553] Tilghman Lesher <tlesher@digium.com> - - * /, 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. ........ - - * UPGRADE-1.6.txt, main/pbx.c, /: 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 15:11 +0000 [r216469-216509] Michiel van Baak <michiel@vanbaak.info> - - * /, 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 13:57 +0000 [r216267-216436] Russell Bryant <russell@digium.com> - - * /, 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 06:14 +0000 [r216225] Michiel van Baak <michiel@vanbaak.info> - - * 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 19:44 +0000 [r216014-216099] Russell Bryant <russell@digium.com> - - * /, 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 18:42 +0000 [r216007] David Vossel <dvossel@digium.com> - - * 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 14:21 +0000 [r215887-215929] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 215891 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r215891 | - oej | 2009-09-03 15:02:41 +0200 (Tor, 03 Sep 2009) | 10 lines Add - known internal IP address when autodomain=yes (closes issue - #14573) Reported by: pj Patches: sip-internip-autodomain1.diff - uploaded by mnicholson (license 96) modified by oej Tested by: pj - ........ - - * main/rtp.c, channels/chan_sip.c: Fix bad reports in "sip show - channelstats". Not directly mergeable in svn trunk, needs more - tests, therefore committed directly to 1.6.2. (closes issue - #15819) Reported by: klaus3000 Patches: - asterisk-1.6.2-beta4-sipshowchannelstats-patch-0.2.txt uploaded - by klaus3000 (license 65) Tested by: klaus3000, oej - -2009-09-03 06:02 +0000 [r215841] Michiel van Baak <michiel@vanbaak.info> - - * doc/manager_1_1.txt, /: Merged revisions 215838 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r215838 | - mvanbaak | 2009-09-03 07:57:23 +0200 (Thu, 03 Sep 2009) | 5 lines - Document that SIPshowpeer and SKINNYshowline now include the - configured parkinglot in their response. Prodded by snuff-work on - #asterisk-dev IRC channel ........ - -2009-09-03 03:44 +0000 [r215802] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_sip.c: Merged revisions 215801 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r215801 | - tilghman | 2009-09-02 22:43:51 -0500 (Wed, 02 Sep 2009) | 5 lines - Default the callback extension to "s". This is a regression. - (closes issue #15764) Reported by: elguero Change-type: bugfix - ........ - -2009-09-03 00:34 +0000 [r215795] Terry Wilson <twilson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 215758 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r215758 | twilson | 2009-09-02 18:31:04 -0500 (Wed, 02 Sep 2009) - | 25 lines Merged revisions 215682 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r215682 | twilson | 2009-09-02 16:41:22 -0500 (Wed, 02 Sep 2009) - | 18 lines Re-send non-100 provisional responses to prevent - cancellation From section 13.3.1.1 of RFC 3261: If the UAS - desires an extended period of time to answer the INVITE, it will - need to ask for an "extension" in order to prevent proxies from - canceling the transaction. A proxy has the option of canceling a - transaction when there is a gap of 3 minutes between responses in - a transaction. To prevent cancellation, the UAS MUST send a - non-100 provisional response at every minute, to handle the - possibility of lost provisional responses. (closes issue #11157) - Reported by: rjain Tested by: twilson Review: - https://reviewboard.asterisk.org/r/315/ ........ ................ - -2009-09-02 21:46 +0000 [r215683] David Vossel <dvossel@digium.com> - - * /, channels/chan_sip.c: Merged revisions 215681 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r215681 | - dvossel | 2009-09-02 16:39:31 -0500 (Wed, 02 Sep 2009) | 10 lines - port string to int conversion using sscanf There are several - instances where a port is parsed from a uri or some other source - and converted to an int value using atoi(), if for some reason - the port string is empty, then a standard port is used. This - logic is used over and over, so I created a function to handle it - in a safer way using sscanf(). ........ - -2009-09-02 21:37 +0000 [r215647-215680] Michiel van Baak <michiel@vanbaak.info> - - * /, channels/chan_sip.c, channels/chan_skinny.c: Merged revisions - 215665 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r215665 | - mvanbaak | 2009-09-02 23:23:17 +0200 (Wed, 02 Sep 2009) | 5 lines - add Parkinglot info to sip show peer <foo> and skinny show line - <foo> If we had this from the start, debugging the 'parking not - using configured parkinglot' bug would have been easier. ........ - - * /, main/features.c: Merged revisions 215622 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r215622 | - mvanbaak | 2009-09-02 22:21:51 +0200 (Wed, 02 Sep 2009) | 4 lines - - lock channel before looking for a channel variable - Init the - parkings list member of struct parkinglot. Thanks Sean for the - explanation why this should be here. ........ - -2009-09-02 18:52 +0000 [r215569-215570] Tilghman Lesher <tlesher@digium.com> - - * /, main/Makefile, main/app.c: Merged revisions 215567 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r215567 | tilghman | 2009-09-02 13:37:25 -0500 (Wed, 02 - Sep 2009) | 9 lines Close up to the soft open file limit (same on - Linux, but varies drastically on OS X). Also, a Makefile fix for - Darwin (OS X). (closes issue #14542) Reported by: jtodd Patches: - 20090901__issue14542.diff.txt uploaded by tilghman (license 14) - Tested by: jtodd, tilghman Change-type: bugfix ........ - - * /, channels/chan_sip.c: Merged revisions 215222 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r215222 | - tilghman | 2009-09-01 16:19:40 -0500 (Tue, 01 Sep 2009) | 3 lines - Fix register such that lines with a transport string, but without - an authuser, parse correctly. (AST-228) ........ - -2009-09-02 17:35 +0000 [r215523] David Vossel <dvossel@digium.com> - - * /, channels/chan_sip.c: Merged revisions 215522 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r215522 | - dvossel | 2009-09-02 12:26:40 -0500 (Wed, 02 Sep 2009) | 11 lines - SIP uri parsing cleanup Now, the scheme passed to parse_uri can - either be a single scheme, or a list of schemes ',' delimited. - This gets rid of the whole problem of having to create two - buffers and calling parse_uri twice to check for separate - schemes. Review: https://reviewboard.asterisk.org/r/343/ ........ - -2009-09-02 16:35 +0000 [r215512] Michiel van Baak <michiel@vanbaak.info> - - * /, channels/chan_skinny.c: Merged revisions 215479 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r215479 | mvanbaak | 2009-09-02 18:20:23 +0200 (Wed, 02 Sep 2009) - | 3 lines like in chan_sip's sip_new skinny should copy the - configured parkinglot from a line to the newly created channel. - This makes callparking honor the configured parkinglot for skinny - lines as well. ........ - -2009-09-02 16:09 +0000 [r215467] David Vossel <dvossel@digium.com> - - * /, channels/chan_sip.c: Merged revisions 215466 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r215466 | - dvossel | 2009-09-02 11:08:00 -0500 (Wed, 02 Sep 2009) | 16 lines - SIP support for keep-alive event keep-alive events are used by - Sipura/Linksys for NAT keepalive. There currently don't appear to - be any problems with NAT, but everytime a keep-alive event is - received, Asterisk responds with a "489 Bad event". This error - may indicate to a user that NAT problems exist just because this - even is not supported. Now, rather than respond with an error, - the packet is consumed and a "200 ok" is sent just to indicate we - received the packet. (issue #15084) Patches: - chan_sip.keepalive.v1.diff uploaded by IgorG (license 20) - ........ - -2009-09-02 16:07 +0000 [r215465] Michiel van Baak <michiel@vanbaak.info> - - * /, channels/chan_sip.c: Merged revisions 215462 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r215462 | - mvanbaak | 2009-09-02 17:56:46 +0200 (Wed, 02 Sep 2009) | 12 - lines Honor configured parkinglot when parking and retrieving - parked calls Thank oej for pointing out the fact that sip_new did - not copy parkinglot from the peer into the newly created channel. - (closes issue #15538) Reported by: gracedman Patches: - 2009090100_sipnewparkinglot-161.diff.txt uploaded by mvanbaak - (license 7) With mod by me to also fix callparking as well (this - uploaded patch only fixed retrieving a parked call) Tested by: - gracedman, mvanbaak ........ - -2009-09-02 01:49 +0000 [r215376] Dwayne M. Hubbard <dwayne.hubbard@gmail.com> - - * /, apps/app_softhangup.c: Merged revisions 215338 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r215338 | dhubbard | 2009-09-01 20:16:59 -0500 - (Tue, 01 Sep 2009) | 18 lines Merged revisions 215270 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r215270 | dhubbard | 2009-09-01 18:04:52 -0500 (Tue, 01 Sep 2009) - | 12 lines Use strrchr() so SoftHangup will correctly truncate - multi-hyphen channel names In general channel names are in the - form Foo/Bar-Z, but the channel name could have multiple hyphens - and look like Foo/B-a-r-Z. Use strrchr to truncate the channel - name at the last hyphen. (closes issue #15810) Reported by: - dhubbard Patches: dw-softhangup-1.4.patch uploaded by dhubbard - (license 733) ........ ................ - -2009-09-01 20:00 +0000 [r215165] Kevin P. Fleming <kpfleming@digium.com> - - * main/frame.c, /: Merged revisions 215161 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r215161 | - kpfleming | 2009-09-01 14:50:48 -0500 (Tue, 01 Sep 2009) | 3 - lines Ensure that frame dumps of AST_CONTROL_T38_PARAMETERS - frames are properly decoded. ........ - -2009-08-31 16:22 +0000 [r214822-214960] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_local.c, /: Merged revisions 214945 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r214945 | tilghman | 2009-08-31 11:18:33 -0500 - (Mon, 31 Aug 2009) | 14 lines Merged revisions 214940 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r214940 | tilghman | 2009-08-31 11:16:52 -0500 (Mon, 31 Aug 2009) - | 7 lines Also unlock the "other" channel, when returning, due to - glare. (closes issue #15787) Reported by: tim_ringenbach Patches: - chan_local.diff uploaded by tim ringenbach (license 540) Tested - by: tim_ringenbach ........ ................ - - * Makefile, /: Merged revisions 214898 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r214898 | - tilghman | 2009-08-30 17:10:35 -0500 (Sun, 30 Aug 2009) | 2 lines - Force Darwin on ppc platforms to compile with a target level that - supports aliasing. ........ - - * /, configure, include/asterisk/autoconfig.h.in, configure.ac, - pbx/pbx_lua.c: Merged revisions 214819 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r214819 | - tilghman | 2009-08-30 01:43:04 -0500 (Sun, 30 Aug 2009) | 4 lines - If lua is detected with the lua5.1 prefix (or not), adjust the - include path accordingly. Based upon feedback to a release - announcement on the -users list. See - http://lists.digium.com/pipermail/asterisk-users/2009-August/236954.html - ........ - -2009-08-29 Leif Madsen <lmadsen@digium.com> - - * Asterisk 1.6.2.0-rc1 released. - -2009-08-28 20:17 +0000 [r214707] Tilghman Lesher <tlesher@digium.com> - - * main/channel.c, /: Merged revisions 214702 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r214702 | tilghman | 2009-08-28 15:14:39 -0500 (Fri, 28 Aug 2009) - | 15 lines Merged revisions 214701 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r214701 | tilghman | 2009-08-28 15:13:32 -0500 (Fri, 28 Aug 2009) - | 8 lines Modify comment to be a bit more accurate. We have kept - this comment around long enough, that it's pretty clear that - we're keeping the code, because changing the code would require a - pretty fundamental architectural shift. We've also taken - criticism in some quarters, because it was believed that it was - referring to the code being nasty. No, the code isn't nasty, just - the operation itself is rather odd. Fixed for eternity (probably - not). ........ ................ - -2009-08-28 20:05 +0000 [r214700] Kevin P. Fleming <kpfleming@digium.com> - - * makeopts.in, Makefile, /, configure, - include/asterisk/autoconfig.h.in, configure.ac: Merged revisions - 214696 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r214696 | - kpfleming | 2009-08-28 15:01:21 -0500 (Fri, 28 Aug 2009) | 9 - lines Ensure that CFLAGS and/or LDFLAGS provided to configure - script are preserved. Cross-compilation environments want to - provide 'defaults' for compiler and linker options, and - frequently do this by specifying CFLAGS and LDFLAGS in the - environment or as command-line arguments to the configure script. - This patch modifies the configure script and Makefile to preserve - these settings and ensure they are used in the build process. - ........ - -2009-08-28 18:43 +0000 [r214653] Mark Michelson <mmichelson@digium.com> - - * /, include/asterisk/sched.h: Merged revisions 214650 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r214650 | mmichelson | 2009-08-28 13:41:23 -0500 (Fri, 28 Aug - 2009) | 3 lines Fix some incorrect documentation of sched_thread - functions. ........ - -2009-08-27 21:49 +0000 [r214202-214521] Tilghman Lesher <tlesher@digium.com> - - * autoconf/libcurl.m4 (added), /, configure, - include/asterisk/autoconfig.h.in, configure.ac: Merged revisions - 214518 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r214518 | tilghman | 2009-08-27 16:46:46 -0500 (Thu, 27 Aug 2009) - | 14 lines Merged revisions 214517 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r214517 | tilghman | 2009-08-27 16:45:34 -0500 (Thu, 27 Aug 2009) - | 7 lines Use autoconf to detect libcurl, as this enables - cross-compilation checks, something we didn't allow before. - (closes issue #15714) Reported by: pprindeville Patches: - 20090813__issue15714.diff.txt uploaded by tilghman (license 14) - Tested by: pprindeville ........ ................ - - * main/manager.c, /: Merged revisions 214514 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r214514 | - tilghman | 2009-08-27 16:26:37 -0500 (Thu, 27 Aug 2009) | 7 lines - Ensure that we check for the special value - CONFIG_STATUS_FILEINVALID. (closes issue #15786) Reported by: - a_villacis Patches: - asterisk-1.6.2.0-beta4-manager-fix-crash-on-include-nonexistent-file.patch - uploaded by a villacis (license 660) (Plus a few of my own, to - catch the remaining places within manager.c where it could have - been a problem) ........ - - * autoconf/ast_ext_lib.m4, /, configure, - include/asterisk/autoconfig.h.in, configure.ac: Merged revisions - 214466 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r214466 | tilghman | 2009-08-27 12:28:01 -0500 (Thu, 27 Aug 2009) - | 9 lines Merged revisions 214436 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r214436 | tilghman | 2009-08-27 11:53:58 -0500 (Thu, 27 Aug 2009) - | 2 lines One more build system change, to make the descriptions - look better, if we have better information. ........ - ................ - - * autoconf/ast_ext_lib.m4, /, configure, - include/asterisk/autoconfig.h.in: Merged revisions 214360 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r214360 | tilghman | 2009-08-27 11:12:03 -0500 - (Thu, 27 Aug 2009) | 10 lines Merged revisions 214357 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r214357 | tilghman | 2009-08-27 11:03:50 -0500 (Thu, 27 Aug 2009) - | 3 lines Make autoheader descriptions render correctly in our - autoconfig.h file. (Figured out while working with issue #14906) - ........ ................ - - * /, channels/chan_sip.c: Merged revisions 214199 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r214199 | - tilghman | 2009-08-26 11:53:03 -0500 (Wed, 26 Aug 2009) | 6 lines - Typo fix ("SIP/2.0 XXX" is 11 chars, not 10) (closes issue - #15362) Reported by: klaus3000 Patches: - chan_sip.c_logmessagefix_patch.txt uploaded by klaus3000 (license - 65) ........ - -2009-08-26 16:39 +0000 [r214196] David Vossel <dvossel@digium.com> - - * main/channel.c, /: Merged revisions 214195 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r214195 | dvossel | 2009-08-26 11:38:53 -0500 (Wed, 26 Aug 2009) - | 25 lines Merged revisions 214194 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r214194 | dvossel | 2009-08-26 11:36:42 -0500 (Wed, 26 Aug 2009) - | 19 lines ast_write() ignores ast_audiohook_write() results In - ast_write(), if a channel has a list of audiohooks, those lists - are written to and the resulting frame is what ast_write() should - continue with. The problem was the returned audiohook frame was - not being handled at all, and the original frame passed into it - did not contain the mixed audio, so essentially audio was being - lost. One result of this was chan_spy's whisper mode no longer - worked. To complicate the issue, frames passed into ast_write may - either be a single frame, or a list of frames. So, as the list of - frames is processed in the audiohook_write, the returned frames - had to be added to a new list. (closes issue #15660) Reported by: - corruptor Tested by: dvossel ........ ................ - -2009-08-25 22:43 +0000 [r213903-214155] Tilghman Lesher <tlesher@digium.com> - - * /, configure, include/asterisk/autoconfig.h.in, configure.ac: - Merged revisions 214152 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r214152 | - tilghman | 2009-08-25 17:39:51 -0500 (Tue, 25 Aug 2009) | 4 lines - Not all versions of gnu-linux use glibc, which contains iconv. - Some (especially embedded systems) don't have iconv at all. - (closes issue #15169) Reported by: pprindeville ........ - - * /, main/say.c: Merged revisions 214071 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r214071 | tilghman | 2009-08-25 14:32:48 -0500 (Tue, 25 Aug 2009) - | 17 lines Merged revisions 214068-214069 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r214068 | tilghman | 2009-08-25 14:26:50 -0500 (Tue, 25 Aug 2009) - | 6 lines Fix pronunciation of German dates. (closes issue - #15273) Reported by: Benjamin Kluck Patches: say_c.patch uploaded - by Benjamin Kluck (license 803) ........ r214069 | tilghman | - 2009-08-25 14:28:42 -0500 (Tue, 25 Aug 2009) | 2 lines I should - always compile before committing... ........ ................ - - * /, pbx/pbx_dundi.c: Merged revisions 213975 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r213975 | - tilghman | 2009-08-25 01:51:12 -0500 (Tue, 25 Aug 2009) | 6 lines - DUNDILOOKUP function in 1.6 should use comma delimiters. (closes - issue #15322) Reported by: chappell Patches: - dundilookup-0015322.patch uploaded by chappell (license 8) - ........ - - * main/pbx.c, /: Merged revisions 213971 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r213971 | tilghman | 2009-08-25 01:35:37 -0500 (Tue, 25 Aug 2009) - | 14 lines Merged revisions 213970 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r213970 | tilghman | 2009-08-25 01:34:44 -0500 (Tue, 25 Aug 2009) - | 7 lines Improve error message by informing user exactly which - function is missing a parethesis. (closes issue #15242) Reported - by: Nick_Lewis Patches: pbx.c-funcparenthesis.patch2 uploaded by - dbrooks (license 790) pbx.c-funcparenthesis-1.4.diff uploaded by - loloski (license 68) ........ ................ - - * Makefile, /: Merged revisions 213904 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r213904 | - tilghman | 2009-08-24 21:54:07 -0500 (Mon, 24 Aug 2009) | 6 lines - The DTD should be installed in the same path as the rest of the - XML documentation. (closes issue #15344) Reported by: tzafrir - Patches: makefile_appdocs_dtd.diff uploaded by tzafrir (license - 46) ........ - - * Makefile, /: Merged revisions 213900 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r213900 | tilghman | 2009-08-24 21:41:17 -0500 (Mon, 24 Aug 2009) - | 11 lines Merged revisions 213899 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r213899 | tilghman | 2009-08-24 21:40:22 -0500 (Mon, 24 Aug 2009) - | 4 lines Use the default runlevels for Debian derivatives, - instead of making up our own. (closes issue #14730) Reported by: - pkempgen ........ ................ - -2009-08-24 16:49 +0000 [r213836] Jeff Peeler <jpeeler@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 213833 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r213833 | jpeeler | 2009-08-24 11:43:57 -0500 (Mon, 24 Aug 2009) - | 14 lines Fix storage of greetings when using IMAP_STORAGE The - store macro was not getting called preventing storage of IMAP - greetings at all. This has been corrected along with fixing - checking if the imapgreetings option is turned on to store the - greeting in IMAP. Lastly, the attachment filename was incorrectly - using the full path instead of just the basename, which was - causing problems with retrieval of the greeting. (closes issue - #14950) Reported by: noahisaac (closes issue #15729) Reported by: - lmadsen ........ - -2009-08-24 04:54 +0000 [r213791] Moises Silva <moises.silva@gmail.com> - - * channels/chan_dahdi.c, /: Merged revisions 213790 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r213790 | moy | 2009-08-24 00:46:28 -0400 (Mon, 24 Aug 2009) | 1 - line improve handling of openr2_chan_disconnect_call API failure, - unlikely, but happened on openr2 library bug ........ - -2009-08-21 22:54 +0000 [r213739] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_sip.c: Merged revisions 213738 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r213738 | - tilghman | 2009-08-21 17:36:39 -0500 (Fri, 21 Aug 2009) | 2 lines - Clarifying comments in sip_register, and removing a dead section - ........ - -2009-08-21 22:23 +0000 [r213721] David Vossel <dvossel@digium.com> - - * /, channels/chan_sip.c: Merged revisions 213716 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r213716 | - dvossel | 2009-08-21 17:22:11 -0500 (Fri, 21 Aug 2009) | 10 lines - Register request line contains wrong address when user domain and - register host differ (closes issue #15539) Reported by: - Nick_Lewis Patches: chan_sip.c-registraraddr.patch uploaded by - Nick (license 657) register_domain_fix_1.6.2 uploaded by dvossel - (license 671) Tested by: Nick_Lewis, dvossel ........ - -2009-08-21 21:44 +0000 [r213698] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 213697 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r213697 | kpfleming | 2009-08-21 16:39:51 -0500 (Fri, 21 Aug - 2009) | 12 lines Ensure that realtime mailboxes properly report - status on subscription. This patch modifies app_voicemail's - response to mailbox status subscriptions (via the internal event - system) to ensure that a subscription triggers an explicit poll - of the mailbox, so the subscriber can get an immediate cached - event with that status. Previously, the cache was only populated - with the status of non-realtime mailboxes. (closes issue #15717) - Reported by: natmlt ........ - -2009-08-21 21:12 +0000 [r213636] David Vossel <dvossel@digium.com> - - * /, channels/chan_sip.c: Merged revisions 213635 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r213635 | - dvossel | 2009-08-21 16:02:50 -0500 (Fri, 21 Aug 2009) | 5 lines - fixes sip register parsing when user@domain is used (issue - #15008) (issue #15672) ........ - -2009-08-21 16:55 +0000 [r213563] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk.h, /: Merged revisions 213560 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r213560 | tilghman | 2009-08-21 11:53:52 -0500 (Fri, 21 Aug 2009) - | 14 lines Merged revisions 213559 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r213559 | tilghman | 2009-08-21 11:52:53 -0500 (Fri, 21 Aug 2009) - | 7 lines Permit DEBUG_FD_LEAKS to be used with C++ source files. - (closes issue #15698) Reported by: slavon Patches: - 20090817__issue15698.diff.txt uploaded by tilghman (license 14) - Tested by: slavon, tilghman ........ ................ - -2009-08-21 16:06 +0000 [r213497] Jason Parker <jparker@digium.com> - - * /, configs/queues.conf.sample: Merged revisions 213494 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r213494 | qwell | 2009-08-21 11:04:21 -0500 - (Fri, 21 Aug 2009) | 12 lines Merged revisions 213493 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r213493 | qwell | 2009-08-21 11:03:21 -0500 (Fri, 21 Aug 2009) | - 5 lines Clarify queues.conf comments to specify that variables - should be set in the dialplan. (closes issue #15755) Reported by: - trendboy ........ ................ - -2009-08-21 04:25 +0000 [r213475-213481] Moises Silva <moises.silva@gmail.com> - - * channels/chan_dahdi.c, /: Merged revisions 213454 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r213454 | moy | 2009-08-21 00:09:26 -0400 (Fri, 21 Aug 2009) | 1 - line increment the mfcr2 monitor count when clearing the call - request ........ - - * channels/chan_dahdi.c, /: Merged revisions 213216 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r213216 | moy | 2009-08-19 23:26:59 -0400 (Wed, 19 Aug 2009) | 1 - line fixed bug caused by calling ast_request without calling - ast_call on an R2 channel, ie, CHANISAVAIL ........ - -2009-08-21 03:53 +0000 [r213453] Terry Wilson <twilson@digium.com> - - * main/loader.c, /: Merged revisions 213450 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r213450 | - twilson | 2009-08-20 22:48:54 -0500 (Thu, 20 Aug 2009) | 2 lines - Make LOAD_ORDER actually work ........ - -2009-08-20 21:50 +0000 [r213413] Jeff Peeler <jpeeler@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 213404 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r213404 | jpeeler | 2009-08-20 16:33:11 -0500 (Thu, 20 Aug 2009) - | 12 lines Fix greeting retrieval from IMAP Properly check for - the current voicemail state and if it doesn't exist, create it. - (closes issue #14597) Reported by: wtca Patches: 14597_v2.patch - uploaded by mmichelson (license 60) Tested by: jpeeler ........ - -2009-08-20 20:37 +0000 [r213350] Matthew Nicholson <mnicholson@digium.com> - - * /, main/features.c: Merged revisions 213327 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r213327 | - mnicholson | 2009-08-20 15:29:32 -0500 (Thu, 20 Aug 2009) | 7 - lines Fix a crash by checking the proper pointer for validity - before deferencing it. (closes issue #15751) Reported by: atis - Patches: ast_bridge_call_peer_cdr.patch uploaded by atis (license - 242) ........ - -2009-08-19 22:41 +0000 [r213182] Jason Parker <jparker@digium.com> - - * main/alaw.c, main/ulaw.c, /: Merged revisions 213179 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r213179 | qwell | 2009-08-19 17:38:46 -0500 (Wed, 19 Aug 2009) | - 5 lines Fix compile when certain G711 menuselect options are - enabled. (closes issue #15697) Reported by: slavon ........ - -2009-08-19 21:25 +0000 [r213128] David Vossel <dvossel@digium.com> - - * apps/app_mixmonitor.c, /: Merged revisions 213113 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r213113 | dvossel | 2009-08-19 16:21:00 -0500 - (Wed, 19 Aug 2009) | 14 lines Merged revisions 213103 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r213103 | dvossel | 2009-08-19 16:18:37 -0500 (Wed, 19 Aug 2009) - | 8 lines Fixes memory leak caused by incorrectly freeing - mixmonitor (closes issue #15699) Reported by: edantie Patches: - mixmonitor.patch uploaded by edantie (license 862) ........ - ................ - -2009-08-19 21:22 +0000 [r213095-213117] Tilghman Lesher <tlesher@digium.com> - - * configs/sip.conf.sample, /, channels/chan_sip.c: Merged revisions - 213098 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r213098 | - tilghman | 2009-08-19 16:05:17 -0500 (Wed, 19 Aug 2009) | 9 lines - Better parsing for the "register" line Allows characters that are - otherwise used as delimiters to be used within certain fields - (like the secret). (closes issue #15008, closes issue #15672) - Reported by: tilghman Patches: 20090818__issue15008.diff.txt - uploaded by tilghman (license 14) Tested by: lmadsen, tilghman - ........ - - * /, channels/chan_sip.c: Merged revisions 213093 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r213093 | - tilghman | 2009-08-19 15:29:41 -0500 (Wed, 19 Aug 2009) | 7 lines - If we have realtime caching enabled, 'sip reload' must purge - users/peers, even if the config files haven't changed. (closes - issue #12869) Reported by: bcnit Patches: - 20090819__issue12869__2.diff.txt uploaded by tilghman (license - 14) Tested by: lasko ........ - -2009-08-19 15:35 +0000 [r213047] Russell Bryant <russell@digium.com> - - * /, main/features.c: Merged revisions 213046 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r213046 | - russell | 2009-08-19 10:32:18 -0500 (Wed, 19 Aug 2009) | 4 lines - Don't blow up on a NULL cdr. Reported in #asterisk-dev. ........ - -2009-08-18 20:34 +0000 [r212931-212944] Kevin P. Fleming <kpfleming@digium.com> - - * /: Merged revisions 212939 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r212939 | - kpfleming | 2009-08-18 15:33:34 -0500 (Tue, 18 Aug 2009) | 1 line - Remove some accidentally-committed properties. ........ - - * sounds/Makefile, doc/tex/asterisk.tex, CREDITS, /, - UPGRADE-1.4.txt, sounds/sounds.xml, build_tools/prep_tarball: - Merged revisions 212922 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r212922 | - kpfleming | 2009-08-18 15:29:37 -0500 (Tue, 18 Aug 2009) | 6 - lines Convert this branch to Opsound music-on-hold. For more - details: - http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/ - ........ - -2009-08-18 19:28 +0000 [r212866] Tilghman Lesher <tlesher@digium.com> - - * /, configs/extconfig.conf.sample: Merged revisions 212857 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r212857 | tilghman | 2009-08-18 14:25:09 -0500 (Tue, 18 - Aug 2009) | 4 lines Make the default extconfig.conf match entries - with the sample res_mysql.conf. This eliminates a future source - of possible confusion with the configuration of 1.6.1 and higher. - ........ - -2009-08-18 16:56 +0000 [r212769] Richard Mudgett <rmudgett@digium.com> - - * channels/misdn/isdn_lib.c, /: Merged revisions 212758 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r212758 | rmudgett | 2009-08-18 11:29:47 -0500 - (Tue, 18 Aug 2009) | 9 lines Merged revisions 212727 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r212727 | rmudgett | 2009-08-18 11:00:56 -0500 (Tue, 18 - Aug 2009) | 1 line Removed some deadwood and added some doxygen - comments. ........ ................ - -2009-08-18 16:41 +0000 [r212767] Sean Bright <sean@malleable.com> - - * main/manager.c, /: Merged revisions 212764 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r212764 | seanbright | 2009-08-18 12:38:36 -0400 (Tue, 18 Aug - 2009) | 18 lines Merged revisions 212763 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r212763 | seanbright | 2009-08-18 12:36:00 -0400 (Tue, 18 Aug - 2009) | 11 lines Delay the creation of temporary files until we - have a valid manager command to handle. Without this patch, - asterisk creates a temporary file before determining if the - specified command is valid. If invalid, we weren't properly - cleaning up the file. (closes issue #15730) Reported by: zmehmood - Patches: M15730.diff uploaded by junky (license 177) Tested by: - zmehmood ........ ................ - -2009-08-17 20:01 +0000 [r212631] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 212627 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r212627 | tilghman | 2009-08-17 14:57:42 -0500 (Mon, 17 Aug 2009) - | 4 lines Check the return value of opendir(3), or we may crash. - (closes issue #15720) Reported by: tobias_e ........ - -2009-08-17 18:56 +0000 [r212580-212584] Sean Bright <sean@malleable.com> - - * /, channels/chan_agent.c: Merged revisions 212581 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r212581 | seanbright | 2009-08-17 14:50:24 -0400 (Mon, 17 Aug - 2009) | 5 lines Correct spelling of AGENTACCEPTDTMF in - chan_agent. (closes issue #15668) Reported by: davidw ........ - - * main/logger.c: Merged revisions 212574 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r212574 | - seanbright | 2009-08-17 14:18:16 -0400 (Mon, 17 Aug 2009) | 8 - lines Correct the return value check for ast_safe_system. The - logic here was reversed as ast_safe_system returns -1 on error - and not on success. Fix suggested by reporter. (closes issue - #15667) Reported by: loic ........ - -2009-08-17 16:52 +0000 [r212509] Jeff Peeler <jpeeler@digium.com> - - * channels/misdn_config.c, /: Merged revisions 212506 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r212506 | jpeeler | 2009-08-17 11:50:45 -0500 - (Mon, 17 Aug 2009) | 19 lines Merged revisions 212498 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r212498 | jpeeler | 2009-08-17 11:34:56 -0500 (Mon, 17 Aug 2009) - | 12 lines Fix segfault when reloading chan_misdn. If more ports - were specified than configured in misdn.conf a reload would crash - asterisk. The problem was the unconfigured port was using data - from the previously configured port. When the data for an - unconfigured port was freed a crash would result from the double - free. (closes issue #12113) Reported by: agupta Patches: - bug12113.patch uploaded by jpeeler (license 325) ........ - ................ - -2009-08-17 15:51 +0000 [r212434] Richard Mudgett <rmudgett@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 212431 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r212431 | rmudgett | 2009-08-17 10:42:51 -0500 - (Mon, 17 Aug 2009) | 16 lines Merged revisions 212430 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 Fix - uninitialized variable causing random MWI indications. (closes - issue #15727) Reported by: doda Patches: dahdi_changes.patch - uploaded by doda (license 853) ........ r212430 | rmudgett | - 2009-08-17 10:36:28 -0500 (Mon, 17 Aug 2009) | 1 line Fix - uninitialized variable. ........ ................ - -2009-08-14 17:37 +0000 [r212250] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_curl.c, /: Merged revisions 212249 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r212249 | - tilghman | 2009-08-14 12:36:40 -0500 (Fri, 14 Aug 2009) | 2 lines - Add SSL_VERIFYPEER, as requested on the -users list ........ - -2009-08-13 15:47 +0000 [r212116] Kevin P. Fleming <kpfleming@digium.com> - - * /, channels/chan_sip.c: Merged revisions 212113 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r212113 | - kpfleming | 2009-08-13 10:46:25 -0500 (Thu, 13 Aug 2009) | 3 - lines Ensure that T38FaxVersion is put into outgoing SDP in the - proper case. ........ - -2009-08-13 13:56 +0000 [r212070] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 212067 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r212067 | - file | 2009-08-13 10:51:04 -0300 (Thu, 13 Aug 2009) | 2 lines - Check an actual populated variable when seeing if we need to do - video or not. ........ - -2009-08-13 11:47 +0000 [r212030] Gavin Henry <ghenry@suretecsystems.com> - - * contrib/scripts/asterisk.ldap-schema, - contrib/scripts/asterisk.ldif, /: Merged revisions 212027 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r212027 | ghenry | 2009-08-13 12:37:12 +0100 (Thu, 13 - Aug 2009) | 6 lines Fixed typo (closes issue #15710) Reported by: - suretec ........ - -2009-08-12 23:16 +0000 [r211951-211959] Matthew Nicholson <mnicholson@digium.com> - - * apps/app_queue.c, /: Merged revisions 211957 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r211957 | mnicholson | 2009-08-12 18:14:36 -0500 (Wed, 12 Aug - 2009) | 17 lines Merged revisions 211953 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r211953 | mnicholson | 2009-08-12 18:04:02 -0500 (Wed, 12 Aug - 2009) | 10 lines This patch adds additional checking when - generating queue log TRANSFER events. The additional checks - prevent generation of false TRANSFER events in certain - situations. (closes issue #14536) Reported by: aragon Patches: - queue-log-xfer-fix1.diff uploaded by mnicholson (license 96) - Tested by: aragon, mnicholson ........ ................ - - * /, channels/chan_sip.c: Merged revisions 211876 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r211876 | - mnicholson | 2009-08-12 14:53:14 -0500 (Wed, 12 Aug 2009) | 11 - lines Make asterisk handle 423 Interval Too Short messages - better. This change uses separate values for the acceptable - minimum expiry provided by the 423 error and the expiry value - stored in the configuration file. Previously, the value pulled - from the configuration file would be overwritten. (closes issue - #14366) Reported by: Nick_Lewis Patches: sip-expiry-fix1.diff - uploaded by mnicholson (license 96) chan_sip.c-reqexpiry.patch - uploaded by Nick (license 657) Tested by: mnicholson ........ - -2009-08-12 16:21 +0000 [r211785] Gavin Henry <ghenry@suretecsystems.com> - - * res/res_config_ldap.c, contrib/scripts/asterisk.ldap-schema, - contrib/scripts/asterisk.ldif, /: Merged revisions 211767 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r211767 | ghenry | 2009-08-12 17:00:46 +0100 (Wed, 12 - Aug 2009) | 33 lines Added three new attributes and applied a - patch to res_config_ldap.c attributetype ( - AstAccountSubscribeContext NAME 'AstAccountSubscribeContext' DESC - 'Asterisk subscribe context' EQUALITY caseIgnoreMatch SUBSTR - caseIgnoreSubstringsMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.15) - attributetype ( AstAccountIpAddr NAME 'AstAccountIpAddr' DESC - 'Asterisk aaccount IP address' EQUALITY caseIgnoreMatch SUBSTR - caseIgnoreSubstringsMatch SYNTAX 1.3.6.1.4.1.1466.115.121.1.15) - attributetype ( AstAccountUserAgent NAME 'AstAccountUserAgent' - DESC 'Asterisk account user context' EQUALITY caseIgnoreMatch - SUBSTR caseIgnoreSubstringsMatch SYNTAX - 1.3.6.1.4.1.1466.115.121.1.15) and patch - fix_empty_attributes_1.6.1.4_v2.patch (closes issue #13725) - Reported by: macogeek Patches: - fix_empty_attributes_1.6.1.4_v2.patch uploaded by xvisor (license - 863) Tested by: suretec ........ - -2009-08-10 19:51 +0000 [r211580-211585] Tilghman Lesher <tlesher@digium.com> - - * 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 - ........ ................ - - * apps/app_queue.c, apps/app_talkdetect.c, agi/eagi-sphinx-test.c, - res/res_config_curl.c, channels/chan_usbradio.c, - channels/chan_misdn.c, res/snmp/agent.c, apps/app_sms.c, - apps/app_verbose.c, apps/app_stack.c, apps/app_mixmonitor.c, - main/asterisk.c, main/dsp.c, main/timing.c, - doc/CODING-GUIDELINES, funcs/func_speex.c, main/frame.c, - utils/muted.c, apps/app_meetme.c, apps/app_alarmreceiver.c, - cdr/cdr_pgsql.c, res/res_musiconhold.c, channels/chan_iax2.c, - apps/app_followme.c, main/enum.c, main/indications.c, - res/res_config_sqlite.c, channels/misdn_config.c, utils/frame.c, - main/cli.c, pbx/pbx_loopback.c, channels/chan_phone.c, - funcs/func_enum.c, res/res_smdi.c, channels/chan_skinny.c, - funcs/func_odbc.c, apps/app_minivm.c, res/res_agi.c, - res/res_config_ldap.c, apps/app_adsiprog.c, - funcs/func_dialplan.c, main/pbx.c, main/dnsmgr.c, - funcs/func_sprintf.c, funcs/func_timeout.c, channels/chan_sip.c, - apps/app_privacy.c, res/res_limit.c, apps/app_waitforsilence.c, - codecs/codec_speex.c, agi/eagi-test.c, apps/app_morsecode.c, - funcs/func_cut.c, channels/chan_oss.c, main/netsock.c, - apps/app_waitforring.c, funcs/func_channel.c, apps/app_macro.c, - pbx/pbx_dundi.c, utils/extconf.c, pbx/pbx_config.c, - apps/app_chanspy.c, res/res_odbc.c, apps/app_voicemail.c, - apps/app_dahdibarge.c, funcs/func_rand.c, apps/app_readfile.c, /, - apps/app_record.c, main/utils.c, cdr/cdr_adaptive_odbc.c, - res/res_http_post.c, main/config.c, res/ael/pval.c, main/cdr.c, - main/channel.c, channels/chan_dahdi.c, pbx/pbx_spool.c, - main/manager.c, apps/app_setcallerid.c, apps/app_osplookup.c, - main/features.c, main/http.c, channels/xpmr/xpmr.c, - apps/app_rpt.c, channels/chan_mgcp.c, res/res_config_pgsql.c, - channels/chan_agent.c, funcs/func_math.c, apps/app_waituntil.c, - apps/app_disa.c, main/acl.c, apps/app_originate.c, - channels/iax2-provision.c: AST-2009-005 - -2009-08-10 14:15 +0000 [r211350] Joshua Colp <jcolp@digium.com> - - * /, 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 [r211235-211278] Tilghman Lesher <tlesher@digium.com> - - * /, 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 20:18 +0000 [r211122] Russell Bryant <russell@digium.com> - - * 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 18:20 +0000 [r211051] Tilghman Lesher <tlesher@digium.com> - - * 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 13:10 +0000 [r210995] Kevin P. Fleming <kpfleming@digium.com> - - * 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 [r210911-210917] Tilghman Lesher <tlesher@digium.com> - - * 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 17:49 +0000 [r210820] Joshua Colp <jcolp@digium.com> - - * /, 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 20:43 +0000 [r210686] Richard Mudgett <rmudgett@digium.com> - - * 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 18:56 +0000 [r210565-210566] Leif Madsen <lmadsen@digium.com> - - * /: 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 ........ ................ - - * 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 14:55 +0000 [r210191-210241] Kevin P. Fleming <kpfleming@digium.com> - - * 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 ........ ................ - - * configs/sip.conf.sample, configs/skinny.conf.sample, main/rtp.c, - channels/chan_mgcp.c, doc/chan_sip-perf-testing.txt, - contrib/scripts/realtime_pgsql.sql, /, channels/chan_sip.c, - channels/chan_skinny.c, configs/mgcp.conf.sample, - doc/res_config_sqlite.txt, doc/tex/phoneprov.tex, UPGRADE.txt, - configs/res_ldap.conf.sample: Merged revisions 210190 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r210190 | kpfleming | 2009-08-03 15:48:48 -0500 (Mon, 03 - Aug 2009) | 11 lines Rename 'canreinvite' option to - 'directmedia', with backwards compatibility. It is clear from - multiple mailing list, forum, wiki and other sorts of posts that - users don't really understand the effects that the 'canreinvite' - config option actually has, and that in some cases they think - that setting it to 'no' will actually cause various other - features (T.38, MOH, etc.) to not work properly, when in fact - this is not the case. This patch changes the proper name of the - option to what it should have been from the beginning - ('directmedia'), but preserves backwards compatibility for - existing configurations. ........ - -2009-08-01 11:33 +0000 [r209837-209906] Russell Bryant <russell@digium.com> - - * 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 01:34 +0000 [r209816] Kevin P. Fleming <kpfleming@digium.com> - - * pbx/pbx_config.c, channels/misdn/isdn_lib.c, utils/frame.c, - main/pbx.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 22:01 +0000 [r209715] Russell Bryant <russell@digium.com> - - * /, 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 18:51 +0000 [r209594] David Brooks <dbrooks@digium.com> - - * channels/chan_console.c, include/asterisk/abstract_jb.h, - apps/app_forkcdr.c, channels/chan_dahdi.c, - contrib/init.d/rc.debian.asterisk, /, apps/app_sms.c, - codecs/lpc10/pitsyn.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 [r209518] Mark Michelson <mmichelson@digium.com> - - * /, 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 Leif Madsen <lmadsen@digium.com> - - * Release Asterisk 1.6.2.0-beta4 - -2009-07-28 00:19 +0000 [r209328] Tilghman Lesher <tlesher@digium.com> - - * /, 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 [r209265-209282] Kevin P. Fleming <kpfleming@digium.com> - - * /, 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 20:58 +0000 [r209238] Mark Michelson <mmichelson@digium.com> - - * 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 20:33 +0000 [r209234] David Brooks <dbrooks@digium.com> - - * 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 20:23 +0000 [r209135-209222] Mark Michelson <mmichelson@digium.com> - - * 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 16:07 +0000 [r209063] David Brooks <dbrooks@digium.com> - - * apps/app_rpt.c, res/res_smdi.c, pbx/pbx_dundi.c: Just replacing - typos "recieved" with "received". From issue #15360, forgot to - apply to trunk and other branches. - -2009-07-27 15:40 +0000 [r209059] Kevin P. Fleming <kpfleming@digium.com> - - * 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 01:23 +0000 [r208927] Jeff Peeler <jpeeler@digium.com> - - * 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 14:07 +0000 [r208889] Michiel van Baak <michiel@vanbaak.info> - - * 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 12:31 +0000 [r208816-208853] Michiel van Baak <michiel@vanbaak.info> - - * contrib/scripts/install_prereq, /: Merged revisions 208848 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r208848 | mvanbaak | 2009-07-25 14:28:38 +0200 (Sat, 25 - Jul 2009) | 2 lines libxml2-dev is needed as well by default. - ........ - - * main/cli.c, /, configs/cli_aliases.conf.sample: Merged revisions - 208813 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r208813 | - mvanbaak | 2009-07-25 14:03:25 +0200 (Sat, 25 Jul 2009) | 10 - lines add default alias reload to run module reload. Requiring - 'module reload' to reload everything, including core etc makes - russell very unhappy. The default configuration already loads the - 'friendly' aliases template. Added 'reload=module reload' to that - template. Also removed the comment in main/cli.c that reload - should come back. ........ - -2009-07-25 06:26 +0000 [r208755] Jeff Peeler <jpeeler@digium.com> - - * 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 21:13 +0000 [r208695-208710] Russell Bryant <russell@digium.com> - - * /, pbx/pbx_dundi.c: Merged revisions 208709 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r208709 | - russell | 2009-07-24 16:12:43 -0500 (Fri, 24 Jul 2009) | 2 lines - Remove trailing whitespace. ........ - - * main/cli.c, /: Merged revisions 208706 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r208706 | - russell | 2009-07-24 15:54:37 -0500 (Fri, 24 Jul 2009) | 6 lines - Note that "reload" needs to be added back. I keep getting annoyed - at having to type "module reload" to reload everything, so I'm - adding a note that we need to add "reload" back. "module reload" - doesn't really make sense as the command to reload everything, - including the core. ........ - - * main/cli.c, /: Merged revisions 208693 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r208693 | - russell | 2009-07-24 15:25:23 -0500 (Fri, 24 Jul 2009) | 2 lines - Don't log a warning for something that does not affect operation. - ........ - -2009-07-24 19:42 +0000 [r208664] Mark Michelson <mmichelson@digium.com> - - * /: Fixing trunk-blocked property. - -2009-07-24 18:56 +0000 [r208596] Russell Bryant <russell@digium.com> - - * 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 [r208591] Mark Michelson <mmichelson@digium.com> - - * /, 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 15:06 +0000 [r208551] Kevin P. Fleming <kpfleming@digium.com> - - * 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 14:39 +0000 [r208545] Michiel van Baak <michiel@vanbaak.info> - - * 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 22:31 +0000 [r208501] Kevin P. Fleming <kpfleming@digium.com> - - * 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 19:36 +0000 [r208391] Mark Michelson <mmichelson@digium.com> - - * /, 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 19:25 +0000 [r208387] Jeff Peeler <jpeeler@digium.com> - - * 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 [r208266-208320] Mark Michelson <mmichelson@digium.com> - - * /, 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 21:46 +0000 [r208116] Jason Parker <jparker@digium.com> - - * /, 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 ........ - -2009-07-22 16:52 +0000 [r207949-208053] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_realtime.c: Merged revisions 208052 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r208052 | - tilghman | 2009-07-22 11:49:42 -0500 (Wed, 22 Jul 2009) | 7 lines - Clarify documentation on 'realtime update2' to show more than one - condition. (closes issue #15357) Reported by: snuffy Patches: - bug_fix_doc_update2.diff uploaded by snuffy (license 35) - (slightly modified by me) ........ - - * /, 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 22:23 +0000 [r207930] Russell Bryant <russell@digium.com> - - * doc/CODING-GUIDELINES, /: Merged revisions 207925 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r207925 | russell | 2009-07-21 17:22:18 -0500 (Tue, 21 Jul 2009) - | 4 lines Note that we use tabs instead of spaces for - indentation. I'm surprised this was never actually in here... - ........ - -2009-07-21 20:30 +0000 [r207785-207862] Jeff Peeler <jpeeler@digium.com> - - * 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 r207638, this approach could - potentially block for an unacceptable amount of time. - -2009-07-21 14:32 +0000 [r207727] Mark Michelson <mmichelson@digium.com> - - * 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 13:56 +0000 [r207685] Kevin P. Fleming <kpfleming@digium.com> - - * 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 04:51 +0000 [r207638] Jeff Peeler <jpeeler@digium.com> - - * 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 22:14 +0000 [r207523] Mark Michelson <mmichelson@digium.com> - - * /, 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 16:41 +0000 [r207364] Russell Bryant <russell@digium.com> - - * 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 04:19 +0000 [r207327] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c, /: Merged 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 03:50 +0000 [r207315] Richard Mudgett <rmudgett@digium.com> - - * channels/misdn/isdn_lib.c, channels/chan_misdn.c: 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 22:31 +0000 [r207226-207257] Tilghman Lesher <tlesher@digium.com> - - * /, 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: Merged 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 19:40 +0000 [r207104-207159] Jeff Peeler <jpeeler@digium.com> - - * 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 17:52 +0000 [r206869-207030] David Vossel <dvossel@digium.com> - - * /, 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 [r206811] Tilghman Lesher <tlesher@digium.com> - - * 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 22:04 +0000 [r206770] David Vossel <dvossel@digium.com> - - * /, 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 21:50 +0000 [r206765] Richard Mudgett <rmudgett@digium.com> - - * 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 20:20 +0000 [r206703] David Vossel <dvossel@digium.com> - - * /, 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 <sip:123.123.123.123> 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 16:04 +0000 [r206639] Sean Bright <sean@malleable.com> - - * 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 20:26 +0000 [r206598] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_meetme.c, contrib/scripts/meetme.sql: Merged - 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 19:49 +0000 [r206565] Richard Mudgett <rmudgett@digium.com> - - * 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 14:59 +0000 [r206389] Russell Bryant <russell@digium.com> - - * 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 01:59 +0000 [r206373] Richard Mudgett <rmudgett@digium.com> - - * 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 23:27 +0000 [r206281] David Vossel <dvossel@digium.com> - - * /, 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 [r206187] Tilghman Lesher <tlesher@digium.com> - - * 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 21:46 +0000 [r205986] David Vossel <dvossel@digium.com> - - * /, 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 [r205942] Kevin P. Fleming <kpfleming@digium.com> - - * 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 17:54 +0000 [r205882] Mark Michelson <mmichelson@digium.com> - - * /, 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 16:47 +0000 [r205841] David Vossel <dvossel@digium.com> - - * /, 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 16:01 +0000 [r205781] Kevin P. Fleming <kpfleming@digium.com> - - * /, apps/app_fax.c: Merged revisions 205780 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r205780 | - kpfleming | 2009-07-10 11:00:44 -0500 (Fri, 10 Jul 2009) | 11 - lines Eliminate extraneous LOG_DEBUG messages generated by - app_fax. The transmit_audio() and transmit_t38() functions in - app_fax have processing loops that are supposed to wait for - frames to arrive on the channel and then handle them, but they - also have short timeouts so that the loops can have watchdog - timers and do other required processing. This commit changes the - loops to not actually call ast_read() and attempt to process the - returned frame unless a frame actually arrived, eliminating - hundreds of LOG_DEBUG messages and slightly improving - performance. ........ - -2009-07-10 15:58 +0000 [r205779] Mark Michelson <mmichelson@digium.com> - - * /, 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 [r205773] Kevin P. Fleming <kpfleming@digium.com> - - * /, 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 23:56 +0000 [r205731] Richard Mudgett <rmudgett@digium.com> - - * 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 [r205699] Kevin P. Fleming <kpfleming@digium.com> - - * 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 16:19 +0000 [r205595-205603] David Vossel <dvossel@digium.com> - - * 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 08:34 +0000 [r205535] Michiel van Baak <michiel@vanbaak.info> - - * /, 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 22:15 +0000 [r205411-205413] David Vossel <dvossel@digium.com> - - * include/asterisk/pbx.h, include/asterisk/devicestate.h, - main/pbx.c, /, main/devicestate.c: 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. ........ - ................ - - * /, main/devicestate.c: Merged revisions 205410 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r205410 | - dvossel | 2009-07-08 17:02:54 -0500 (Wed, 08 Jul 2009) | 3 lines - missing comma in devstatestring array ........ - -2009-07-08 19:28 +0000 [r205353] Mark Michelson <mmichelson@digium.com> - - * 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 18:22 +0000 [r205302] Jason Parker <jparker@digium.com> - - * 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 18:18 +0000 [r205287] David Brooks <dbrooks@digium.com> - - * /, 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 17:00 +0000 [r205223] Tilghman Lesher <tlesher@digium.com> - - * main/say.c: oops, fixing build - -2009-07-08 16:55 +0000 [r205217] David Vossel <dvossel@digium.com> - - * 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 16:30 +0000 [r205207] Tilghman Lesher <tlesher@digium.com> - - * /, 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 [r205148-205154] Russell Bryant <russell@digium.com> - - * /, 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. ........ - - * include/asterisk/_private.h, res/res_jabber.c, main/asterisk.c, - /, main/Makefile, res/res_crypto.c, main/ssl.c (added): 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 13:41 +0000 [r204951] Kevin P. Fleming <kpfleming@digium.com> - - * 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 22:06 +0000 [r204838] Richard Mudgett <rmudgett@digium.com> - - * 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 16:12 +0000 [r204711] David Vossel <dvossel@digium.com> - - * 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 [r204611] Tilghman Lesher <tlesher@digium.com> - - * /, 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 18:55 +0000 [r204478] Jason Parker <jparker@digium.com> - - * /, 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 [r204473] Tilghman Lesher <tlesher@digium.com> - - * 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 ........ - ................ - -2009-06-30 17:22 +0000 [r204442] Russell Bryant <russell@digium.com> - - * configs/res_config_sqlite.conf (removed), - configs/res_config_sqlite.conf.sample (added), /: Merged - revisions 204440 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r204440 | - russell | 2009-06-30 12:22:16 -0500 (Tue, 30 Jun 2009) | 2 lines - Rename res_config_sqlite.conf to res_config_sqlite.conf.sample - (missing .sample). ........ - -2009-06-29 22:53 +0000 [r204250-204304] Mark Michelson <mmichelson@digium.com> - - * /, 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 09:55 +0000 [r203961] Russell Bryant <russell@digium.com> - - * CHANGES, /: Merged revisions 203960 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r203960 | - russell | 2009-06-27 04:51:45 -0500 (Sat, 27 Jun 2009) | 2 lines - Minor tweaks and spelling fixes for CHANGES and UPGRADE.txt. - ........ - -2009-06-27 01:24 +0000 [r203941] Richard Mudgett <rmudgett@digium.com> - - * 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 22:14 +0000 [r203857] Jeff Peeler <jpeeler@digium.com> - - * 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 21:27 +0000 [r203782-203828] Russell Bryant <russell@digium.com> - - * /, 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 [r203731] David Brooks <dbrooks@digium.com> - - * 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 19:49 +0000 [r203715] Russell Bryant <russell@digium.com> - - * 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 19:48 +0000 [r203712] David Vossel <dvossel@digium.com> - - * 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 19:42 +0000 [r203709] Jeff Peeler <jpeeler@digium.com> - - * 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 ........ - -2009-06-26 19:38 +0000 [r203705] Joshua Colp <jcolp@digium.com> - - * configs/sip.conf.sample, include/asterisk/frame.h, main/rtp.c, - main/channel.c, main/frame.c, /, channels/chan_sip.c, - apps/app_fax.c: 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. ........ - -2009-06-25 21:46 +0000 [r203445] David Vossel <dvossel@digium.com> - - * 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 21:21 +0000 [r203400] Terry Wilson <twilson@digium.com> - - * 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 21:08 +0000 [r203379] Russell Bryant <russell@digium.com> - - * /, 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 [r203276] Jason Parker <jparker@digium.com> - - * 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 16:08 +0000 [r203119] Russell Bryant <russell@digium.com> - - * /, 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 21:27 +0000 [r203077] Richard Mudgett <rmudgett@digium.com> - - * 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 [r202970] Mark Michelson <mmichelson@digium.com> - - * /, 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 18:11 +0000 [r202928] Joshua Colp <jcolp@digium.com> - - * /, 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 23:58 +0000 [r202842] Sean Bright <sean@malleable.com> - - * doc/tex/cdrdriver.tex, /, doc/tex/billing.tex: Merged revisions - 202840-202841 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r202840 | - seanbright | 2009-06-23 19:53:45 -0400 (Tue, 23 Jun 2009) | 1 - line Remove some trailing whitespace before making content - changes. ........ r202841 | seanbright | 2009-06-23 19:57:07 - -0400 (Tue, 23 Jun 2009) | 1 line Change some section names in - the CDR tex documentation. ........ - -2009-06-23 22:47 +0000 [r202805] Russell Bryant <russell@digium.com> - - * doc/tex/cdrdriver.tex, /: Merged revisions 202804 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r202804 | russell | 2009-06-23 17:47:26 -0500 (Tue, 23 Jun 2009) - | 2 lines Clean up section hierarchy for the CDR chapter. - ........ - -2009-06-23 22:12 +0000 [r202765] Matthew Fredrickson <creslin@digium.com> - - * 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 16:33 +0000 [r202673] David Vossel <dvossel@digium.com> - - * /, 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 20:19 +0000 [r202495-202511] Russell Bryant <russell@digium.com> - - * 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 ........ ................ - - * /, 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 16:31 +0000 [r202473] Sean Bright <sean@malleable.com> - - * 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 15:37 +0000 [r202411] David Vossel <dvossel@digium.com> - - * 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 15:17 +0000 [r202340-202346] Mark Michelson <mmichelson@digium.com> - - * /, 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 16:16 +0000 [r202261-202265] Russell Bryant <russell@digium.com> - - * 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 [r202186] Sean Bright <sean@malleable.com> - - * /, 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 [r202007] Matthew Nicholson <mnicholson@digium.com> - - * 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 [r201995] David Vossel <dvossel@digium.com> - - * 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 15:49 +0000 [r201785-201906] Tilghman Lesher <tlesher@digium.com> - - * 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: Merged revisions 201829 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r201829 | tilghman | 2009-06-18 19:43:41 -0500 (Thu, 18 Jun 2009) - | 13 lines Merged revisions 201828 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r201828 | tilghman | 2009-06-18 19:40:41 -0500 (Thu, 18 Jun 2009) - | 6 lines 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 16:44 +0000 [r201679] David Vossel <dvossel@digium.com> - - * 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 15:40 +0000 [r201614] Russell Bryant <russell@digium.com> - - * 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 15:23 +0000 [r201595] David Vossel <dvossel@digium.com> - - * /, 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 21:33 +0000 [r201533] Tilghman Lesher <tlesher@digium.com> - - * 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 20:12 +0000 [r201461-201465] Mark Michelson <mmichelson@digium.com> - - * /, 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 [r201447-201454] David Vossel <dvossel@digium.com> - - * 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 19:49 +0000 [r201446] David Brooks <dbrooks@digium.com> - - * /, 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 15:25 +0000 [r201360] David Vossel <dvossel@digium.com> - - * /, 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 12:06 +0000 [r201265] Kevin P. Fleming <kpfleming@digium.com> - - * /, 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 22:30 +0000 [r201224] David Vossel <dvossel@digium.com> - - * /, 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 19:47 +0000 [r200990-201097] Kevin P. Fleming <kpfleming@digium.com> - - * 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 16:28 +0000 [r200984] David Vossel <dvossel@digium.com> - - * /, 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 16:05 +0000 [r200948] Michiel van Baak <michiel@vanbaak.info> - - * 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 12:55 +0000 [r200842] Eliel C. Sardanons <eliels@gmail.com> - - * res/res_smdi.c, /: Merged revisions 200841 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r200841 | - eliel | 2009-06-16 08:32:00 -0400 (Tue, 16 Jun 2009) | 6 lines - Show the interface name on error, if it is not found. If the - smdiport specified is not found, show the interface name instead - of '(null)'. ........ - -2009-06-16 02:41 +0000 [r200807] Moises Silva <moises.silva@gmail.com> - - * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged - revisions 200799 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r200799 | - moy | 2009-06-15 21:24:30 -0500 (Mon, 15 Jun 2009) | 2 lines keep - backwards compatible chan_dahdi with older openr2 versions by not - using the new skip category feature unless supported ........ - -2009-06-16 01:30 +0000 [r200690-200765] Kevin P. Fleming <kpfleming@digium.com> - - * /, 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: Merged revisions 200689 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r200689 | - kpfleming | 2009-06-15 15:42:38 -0500 (Mon, 15 Jun 2009) | 11 - 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 [r200517] Mark Michelson <mmichelson@digium.com> - - * /, 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 ........ ................ - -2009-06-14 06:33 +0000 [r200512] Moises Silva <moises.silva@gmail.com> - - * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /, - build_tools/menuselect-deps.in: Merged revisions 200477 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r200477 | moy | 2009-06-14 01:13:48 -0500 (Sun, 14 Jun - 2009) | 3 lines added openr2 to menuselect-deps.in, recent commit - in menuselect made me realize this was never done but was working - anyways also added support for skip category request feature of - openr2 and updated chan_dahdi.conf.sample ........ - -2009-06-12 19:08 +0000 [r200364] Mark Michelson <mmichelson@digium.com> - - * 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 ........ ................ - -2009-06-12 02:20 +0000 [r200198-200255] Sean Bright <sean@malleable.com> - - * contrib/init.d/rc.debian.asterisk, /: Merged revisions 200254 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r200254 | seanbright | 2009-06-11 22:20:19 -0400 (Thu, - 11 Jun 2009) | 5 lines Call chgrp instead of chown when setting - run directory group ownership. (issue #13153) Reported by: - pabelanger ........ - - * 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#. ........ - - * /: Undo block of revision 199782 (will be merging it momentarily) - -2009-06-11 21:35 +0000 [r200172] Terry Wilson <twilson@digium.com> - - * main/rtp.c: Don't access rtp->rtcp->* if rtp->rtcp is null - -2009-06-11 21:18 +0000 [r200154] Mark Michelson <mmichelson@digium.com> - - * /, 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 Leif Madsen <lmadsen@digium.com> - - * Release Asterisk 1.6.2.0-beta3 - -2009-06-11 12:19 +0000 [r200051] Leif Madsen <lmadsen@digium.com> - - * 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 20:37 +0000 [r199998] David Brooks <dbrooks@digium.com> - - * 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 [r199966] Mark Michelson <mmichelson@digium.com> - - * /, 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 [r199860] Sean Bright <sean.bright@gmail.com> - - * 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 20:48 +0000 [r199744-199819] David Vossel <dvossel@digium.com> - - * /, 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, - res/res_timing_timerfd.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 [r199634] Sean Bright <sean.bright@gmail.com> - - * 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 17:42 +0000 [r199591] Mark Michelson <mmichelson@digium.com> - - * /, 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 ........ - -2009-06-06 21:39 +0000 [r199369] Russell Bryant <russell@digium.com> - - * Makefile, /: Merged revisions 199368 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r199368 | - russell | 2009-06-06 16:38:54 -0500 (Sat, 06 Jun 2009) | 2 lines - Switch from "echo -n" to printf. On my mac, the -n was just - getting printed out. ........ - -2009-06-05 21:25 +0000 [r199299] David Vossel <dvossel@digium.com> - - * 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 13:52 +0000 [r199230] Mark Michelson <mmichelson@digium.com> - - * 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 19:15 +0000 [r199140] David Vossel <dvossel@digium.com> - - * 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 [r199054] Sean Bright <sean.bright@gmail.com> - - * include/asterisk/_private.h, main/asterisk.c, main/loader.c, /: - 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 15:24 +0000 [r198827-198886] David Vossel <dvossel@digium.com> - - * 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 13:51 +0000 [r198794] Joshua Colp <jcolp@digium.com> - - * configs/sip.conf.sample, /, channels/chan_sip.c: 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 21:04 +0000 [r198730] Russell Bryant <russell@digium.com> - - * channels/iax2-parser.c, /: Merged revisions 198729 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r198729 | russell | 2009-06-01 16:03:18 -0500 (Mon, 01 Jun 2009) - | 2 lines Tell the IAX2 parser about more control frame types. - ........ - -2009-06-01 18:44 +0000 [r198629] Tilghman Lesher <tlesher@digium.com> - - * /, 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 17:53 +0000 [r198471] Tilghman Lesher <tlesher@digium.com> - - * /, funcs/func_strings.c: Merged revisions 198470 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r198470 | tilghman | 2009-05-31 12:52:28 -0500 (Sun, 31 May 2009) - | 2 lines Fix documentation for FIELDQTY. ........ - -2009-05-31 01:48 +0000 [r198440] Eliel C. Sardanons <eliels@gmail.com> - - * /, 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) ........ - -2009-05-31 01:21 +0000 [r198436] Russell Bryant <russell@digium.com> - - * 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 20:22 +0000 [r198297-198397] Sean Bright <sean.bright@gmail.com> - - * 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) ........ - ................ - - * 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 02:35 +0000 [r198250] Joshua Colp <jcolp@digium.com> - - * /, 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 [r198148-198188] Russell Bryant <russell@digium.com> - - * /, 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 19:26 +0000 [r198111] Eliel C. Sardanons <eliels@gmail.com> - - * CREDITS, /: Merged revisions 198083 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r198083 | - eliel | 2009-05-29 15:18:35 -0400 (Fri, 29 May 2009) | 3 lines - Apply anti-spam obfuscation to an email address. ........ - -2009-05-29 19:14 +0000 [r198075] Matthew Nicholson <mnicholson@digium.com> - - * 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 18:40 +0000 [r198066] Joshua Colp <jcolp@digium.com> - - * /, 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 18:18 +0000 [r198008] Sean Bright <sean.bright@gmail.com> - - * 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 16:29 +0000 [r197994] Russell Bryant <russell@digium.com> - - * /, 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 23:54 +0000 [r197894] Leif Madsen <lmadsen@digium.com> - - * apps/app_mixmonitor.c, /: Merged revisions 197828 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r197828 | lmadsen | 2009-05-28 18:04:00 -0400 (Thu, 28 May 2009) - | 8 lines Update documentation in MixMonitor. Updated the - MixMonitor documentation for the 'b' option so that it is more - obvious that you must not optimize away the Local channel when - using this option. (closes issue #14829) Reported by: licedey - Tested by: mmichelson, licedey, lmadsen ........ - -2009-05-28 18:50 +0000 [r197703] Joshua Colp <jcolp@digium.com> - - * 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 16:15 +0000 [r197625] Eliel C. Sardanons <eliels@gmail.com> - - * /, 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 15:44 +0000 [r197548-197619] Mark Michelson <mmichelson@digium.com> - - * 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 ........ ................ - - * main/audiohook.c, apps/app_chanspy.c, /, - include/asterisk/audiohook.h: 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 14:56 +0000 [r197471-197542] Joshua Colp <jcolp@digium.com> - - * /, 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 [r197441] Gavin Henry <ghenry@suretecsystems.com> - - * 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 23:49 +0000 [r197375] Tilghman Lesher <tlesher@digium.com> - - * /, main/xml.c: Merged revisions 197374 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r197374 | - tilghman | 2009-05-27 18:48:15 -0500 (Wed, 27 May 2009) | 2 lines - Revert commit 192032. This define is needed on Mac OS X. ........ - -2009-05-27 22:23 +0000 [r197336] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/agi.h, /: Merged revisions 197335 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r197335 | kpfleming | 2009-05-27 17:21:53 -0500 (Wed, 27 May - 2009) | 3 lines Ensure that this header includes xmldoc.h, since - it depends on it. ........ - -2009-05-27 20:11 +0000 [r197263] Sean Bright <sean.bright@gmail.com> - - * 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 19:30 +0000 [r197247] Tilghman Lesher <tlesher@digium.com> - - * /, 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 17:28 +0000 [r197176] Jeff Peeler <jpeeler@digium.com> - - * 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 [r196950-197092] Sean Bright <sean.bright@gmail.com> - - * 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 ........ - - * /, 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) - ........ - - * main/xmldoc.c, /: Merged revisions 196948 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r196948 | - seanbright | 2009-05-26 18:43:21 -0400 (Tue, 26 May 2009) | 8 - lines Reset the terminal to the correct fg/bg after XML - documenation is rendered. (closes issue #15200) Reported by: - ajohnson Patches: 05262009_xmldoc.patch uploaded by seanbright - (license 71) Tested by: ajohnson ........ - - * main/manager.c, /: Merged revisions 196945 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r196945 | - seanbright | 2009-05-26 18:38:05 -0400 (Tue, 26 May 2009) | 13 - lines Add ActionID to CoreShowChannel event. There is - inconsistency in how we handle manager responses that are lists - of items and, unfortunately, third parties have come to rely on - ActionID being on every event within those lists instead of just - keeping track of the ActionID for the current response. This - change makes CoreShowChannels include the ActionID with each - CoreShowChannel event generated as a result of it being called. - (closes issue #15001) Reported by: sum Patches: - patchactionid2.patch uploaded by sum (license 766) ........ - -2009-05-26 22:44 +0000 [r196870-196949] Russell Bryant <russell@digium.com> - - * /, 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 16:39 +0000 [r196793] Sean Bright <sean.bright@gmail.com> - - * apps/app_queue.c, /: Merged revisions 196792 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r196792 | - seanbright | 2009-05-26 12:38:54 -0400 (Tue, 26 May 2009) | 2 - lines Add a missing unref for queues in handle_statechange. - ........ - -2009-05-26 13:47 +0000 [r196661-196724] Joshua Colp <jcolp@digium.com> - - * /, 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) - ........ ................ - -2009-05-23 05:29 +0000 [r196487] Moises Silva <moises.silva@gmail.com> - - * channels/chan_dahdi.c, /: Merged revisions 196456 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r196456 | moy | 2009-05-22 23:27:47 -0500 (Fri, 22 May 2009) | 1 - line set MFCR2_CATEGORY just when starting the pbx ........ - -2009-05-22 21:59 +0000 [r196452] David Vossel <dvossel@digium.com> - - * configs/sip.conf.sample, /, channels/chan_sip.c: 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 19:48 +0000 [r196378] Eliel C. Sardanons <eliels@gmail.com> - - * /, apps/app_minivm.c: Merged revisions 196377 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r196377 | - eliel | 2009-05-22 15:38:33 -0400 (Fri, 22 May 2009) | 11 lines - Unregister every registered application by MiniVM. The MinivmMWI - application was not being unregistered on unload and we were not - able to load again the module or reload it. (closes issue #15174) - Reported by: junky Patches: unregister_minivm_mwi.diff uploaded - by junky (license 177) ........ - -2009-05-22 13:59 +0000 [r196120] Joshua Colp <jcolp@digium.com> - - * 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 19:15 +0000 [r196000] David Vossel <dvossel@digium.com> - - * 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 15:57 +0000 [r195883] Matthew Nicholson <mnicholson@digium.com> - - * 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 [r195842] Tilghman Lesher <tlesher@digium.com> - - * 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 17:35 +0000 [r195639-195707] Joshua Colp <jcolp@digium.com> - - * /, 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 20:19 +0000 [r195531] Tilghman Lesher <tlesher@digium.com> - - * 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 14:49 +0000 [r195452] Joshua Colp <jcolp@digium.com> - - * /, 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 21:25 +0000 [r195405] Eliel C. Sardanons <eliels@gmail.com> - - * 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 20:55 +0000 [r195359-195373] Tilghman Lesher <tlesher@digium.com> - - * apps/app_queue.c, include/asterisk/smdi.h, res/res_monitor.c, - apps/app_voicemail.c, res/res_smdi.c, /, - include/asterisk/monitor.h: Merged 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 19:01 +0000 [r195319] Mark Michelson <mmichelson@digium.com> - - * 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 15:57 +0000 [r195212] Joshua Colp <jcolp@digium.com> - - * 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 14:54 +0000 [r195164] Eliel C. Sardanons <eliels@gmail.com> - - * 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 14:00 +0000 [r195099] Joshua Colp <jcolp@digium.com> - - * 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 ........ ................ - -2009-05-18 13:50 +0000 [r195093-195094] Eliel C. Sardanons <eliels@gmail.com> - - * /, main/xml.c: Merged revisions 195075 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r195075 | - eliel | 2009-05-18 09:30:34 -0400 (Mon, 18 May 2009) | 3 lines Do - not avoid loading the XML documentation if not XInclude - substitution is done. ........ - - * doc/appdocsxml.dtd, Makefile, /, main/xml.c: Merged revisions - 194982 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r194982 | - eliel | 2009-05-16 16:01:22 -0400 (Sat, 16 May 2009) | 20 lines - Allow to include sections of other parts of the xml - documentation. Avoid duplicating xml documentation by allowing to - include other parts of the xml documentation using XInclude. - Example: <xi:include - xpointer="xpointer(/docs/function[@name='CHANNEL']/synopsis)" /> - (Insert this line to include the synopsis of the CHANNEL function - xml documentation). It is also possible to include documentation - from other files in the 'documentation/' directory using the - href="" attribute inside a xinclude element. (closes issue - #15107) Reported by: lmadsen (issue #14444) Reported by: ewieling - ........ - -2009-05-18 13:39 +0000 [r195092] Joshua Colp <jcolp@digium.com> - - * /, 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 13:14 +0000 [r195024] Russell Bryant <russell@digium.com> - - * 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 ........ ................ - -2009-05-16 18:43 +0000 [r194946] Eliel C. Sardanons <eliels@gmail.com> - - * main/pbx.c, /: Merged revisions 194945 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r194945 | - eliel | 2009-05-16 14:32:11 -0400 (Sat, 16 May 2009) | 8 lines - Fix a missing unlock in case of error, and a missing free(). - Always free the allocated memory for a string field, because we - are always using it (not only when xmldocs are enabled). Also if - there is an error allocating memory for the string field remember - to unlock the list of registered applications, before returning. - ........ - -2009-05-15 22:48 +0000 [r194836-194877] David Vossel <dvossel@digium.com> - - * 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 [r194717-194768] Russell Bryant <russell@digium.com> - - * 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. ........ - -2009-05-15 13:47 +0000 [r194650] Michiel van Baak <michiel@vanbaak.info> - - * CREDITS, /: Merged revisions 194649 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r194649 | - mvanbaak | 2009-05-15 15:43:24 +0200 (Fri, 15 May 2009) | 2 lines - add eliel ........ - -2009-05-15 13:42 +0000 [r194648] Eliel C. Sardanons <eliels@gmail.com> - - * doc/appdocsxml.dtd, main/xmldoc.c, /: Merged revisions 194635 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ........ r194635 | eliel | 2009-05-15 09:23:37 -0400 (Fri, 15 May - 2009) | 16 lines Allow to specify an enumlist inside an enum. It - was not possible to use an enumlist inside an enum: <enumlist> - <enum name="aa"> <enumlist> ... </enumlist> </enum> </enumlist> - Now we will be able to insert as many levels as we want. (closes - issue #15112) Reported by: lmadsen ........ - -2009-05-14 22:31 +0000 [r194545] Kevin P. Fleming <kpfleming@digium.com> - - * /: 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 [r194510] Mark Michelson <mmichelson@digium.com> - - * /, 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 [r194437] Joshua Colp <jcolp@digium.com> - - * /, 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 ........ - -2009-05-14 16:23 +0000 [r194431] Tilghman Lesher <tlesher@digium.com> - - * main/pbx.c, /: Merged revisions 194430 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r194430 | - tilghman | 2009-05-14 11:22:14 -0500 (Thu, 14 May 2009) | 7 lines - If the timing ended on a zero, then we would loop forever. - (closes issue #14983) Reported by: teox Patches: - 20090513__issue14983.diff.txt uploaded by tilghman (license 14) - Tested by: teox ........ - -2009-05-13 13:42 +0000 [r194213] Joshua Colp <jcolp@digium.com> - - * 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 [r194141] Tilghman Lesher <tlesher@digium.com> - - * 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 22:48 +0000 [r194059] Matthew Nicholson <mnicholson@digium.com> - - * 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 [r193962] Mark Michelson <mmichelson@digium.com> - - * /, 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 [r193823-193959] Tilghman Lesher <tlesher@digium.com> - - * 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 22:12 +0000 [r193719] Russell Bryant <russell@digium.com> - - * /, res/res_timing_timerfd.c: Merged revisions 193718 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r193718 | russell | 2009-05-11 17:04:40 -0500 (Mon, 11 May 2009) - | 12 lines Fix some timer state corruption. In res_timer_timerfd, - handle the case that set_rate gets called while a timer is still - in continuous mode. In this case, we want to remember the - configured rate, but not actually set it until continuous mode - has been disabled. Thanks to dvossel for finding and helping to - debug the problem. (closes issue #15080) Reported by: dvossel - Tested by: dvossel ........ - -2009-05-11 19:17 +0000 [r193617] Richard Mudgett <rmudgett@digium.com> - - * 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 18:59 +0000 [r193612] Leif Madsen <lmadsen@digium.com> - - * /, funcs/func_channel.c: Update CHANNEL(transfercapabilities) - documentation. (closes issue #15073) Reported by: pkempgen - Patches: 20090511__issue15073__trunk.diff.txt uploaded by - tilghman (license 14) - -2009-05-10 17:08 +0000 [r193503] Joshua Colp <jcolp@digium.com> - - * main/bridging.c, /: Merged revisions 193502 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r193502 | - file | 2009-05-10 14:07:46 -0300 (Sun, 10 May 2009) | 2 lines Fix - a bug where receiving a control frame of subclass -1 would cause - certain channels to get hung up. ........ - -2009-05-09 11:33 +0000 [r193462] Russell Bryant <russell@digium.com> - - * include/asterisk/event.h, /: Merged revisions 193461 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r193461 | russell | 2009-05-09 06:33:09 -0500 (Sat, 09 May 2009) - | 2 lines Minor documentation update for ast_event_queue(). - ........ - -2009-05-08 20:52 +0000 [r193390] David Vossel <dvossel@digium.com> - - * /, 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 19:51 +0000 [r193350] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c, /: Merged revisions 193349 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r193349 | - mmichelson | 2009-05-08 14:50:44 -0500 (Fri, 08 May 2009) | 12 - lines Reset the members' call counts when resetting queue - statistics. This helps to prevent odd scenarios where a queue - will claim to have taken 0 calls, but the members appear to have - taken a non-zero amount. (closes issue #15068) Reported by: sum - Patches: patchreset.patch uploaded by sum (license 766) Tested - by: sum ........ - -2009-05-08 15:36 +0000 [r193336] Sean Bright <sean.bright@gmail.com> - - * 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 14:55 +0000 [r193266] David Vossel <dvossel@digium.com> - - * 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 14:12 +0000 [r193197] Kevin P. Fleming <kpfleming@digium.com> - - * 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 [r193123] Tilghman Lesher <tlesher@digium.com> - - * 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 22:51 +0000 [r193080] Richard Mudgett <rmudgett@digium.com> - - * 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 17:53 +0000 [r192936-193008] Tilghman Lesher <tlesher@digium.com> - - * /, 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 ........ ................ - -2009-05-07 Leif Madsen <lmadsen@digium.com> - - * Release Asterisk 1.6.2.0-beta2 - -2009-05-06 22:20 +0000 [r192874] Jeff Peeler <jpeeler@digium.com> - - * /, 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 17:57 +0000 [r192813] Matthew Fredrickson <creslin@digium.com> - - * 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 17:41 +0000 [r192637-192810] Joshua Colp <jcolp@digium.com> - - * 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 ........ - - * res/res_clialiases.c, /: Merged revisions 192736 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r192736 | file | 2009-05-06 13:09:27 -0300 (Wed, 06 May 2009) | 4 - lines Make the code that prevents an infinite loop from happening - into a case insensitive check. (thanks eliel) ........ - - * res/res_clialiases.c, /: Merged revisions 192700 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r192700 | file | 2009-05-06 11:35:47 -0300 (Wed, 06 May 2009) | 5 - lines Fix an infinite loop with tab completion of CLI aliases - that reference themselves. (closes issue #15020) Reported by: - junky ........ - - * /, 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 [r192528] Sean Bright <sean.bright@gmail.com> - - * /, 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 18:27 +0000 [r192402-192480] Joshua Colp <jcolp@digium.com> - - * /, 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 13:43 +0000 [r192298-192360] Kevin P. Fleming <kpfleming@digium.com> - - * 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 [r192217] David Vossel <dvossel@digium.com> - - * 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 19:34 +0000 [r192175] Kevin P. Fleming <kpfleming@digium.com> - - * 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 19:31 +0000 [r192135-192173] Tilghman Lesher <tlesher@digium.com> - - * /, configure, res/res_agi.c: Merged 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 ........ - - * 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 17:45 +0000 [r192097] Leif Madsen <lmadsen@digium.com> - - * apps/app_forkcdr.c, /: Merged revisions 192096 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r192096 | - lmadsen | 2009-05-04 13:42:56 -0400 (Mon, 04 May 2009) | 4 lines - Commit documentation changes related to issue #14801. (issue - #14801) ........ - -2009-05-04 15:54 +0000 [r192033] Eliel C. Sardanons <eliels@gmail.com> - - * /, main/xml.c: Merged revisions 192032 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r192032 | - eliel | 2009-05-04 11:35:35 -0400 (Mon, 04 May 2009) | 3 lines Do - not re-define _POSIX_C_SOURCE if it was already defined. ........ - -2009-05-04 10:01 +0000 [r191958] Kevin P. Fleming <kpfleming@digium.com> - - * /, 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. ........ - -2009-05-03 14:06 +0000 [r191885] Russell Bryant <russell@digium.com> - - * Makefile, /: Merged revisions 191884 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r191884 | - russell | 2009-05-03 09:05:10 -0500 (Sun, 03 May 2009) | 2 lines - Remove unnecessary compiler flag ........ - -2009-05-02 18:48 +0000 [r191779] Kevin P. Fleming <kpfleming@digium.com> - - * /, 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 [r191703] Sean Bright <sean.bright@gmail.com> - - * 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 [r191554-191563] Tilghman Lesher <tlesher@digium.com> - - * /, 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 [r191508] Jeff Peeler <jpeeler@digium.com> - - * 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. ........ ................ - -2009-04-30 17:46 +0000 [r191224-191370] Tilghman Lesher <tlesher@digium.com> - - * 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 [r191139] David Brooks <dbrooks@digium.com> - - * 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 08:59 +0000 [r190994] Russell Bryant <russell@digium.com> - - * main/indications.c, /: Merged revisions 190993 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r190993 | - russell | 2009-04-29 03:58:39 -0500 (Wed, 29 Apr 2009) | 7 lines - Log an error message if indications.conf is not found. (closes - issue #14990) Reported by: tzafrir Patches: indications_err.diff - uploaded by tzafrir (license 46) ........ - -2009-04-29 06:38 +0000 [r190985] TransNexus OSP Development <support@transnexus.com> - - * 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 [r190907] Tilghman Lesher <tlesher@digium.com> - - * 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 14:17 +0000 [r190732-190869] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile, /: Merged revisions 190865 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r190865 | - kpfleming | 2009-04-28 09:15:47 -0500 (Tue, 28 Apr 2009) | 5 - lines Build XML documention from *only* the source files that - have docs in them Change the build process so that - doc/core-en_US.xml is dependent solely on the source files that - have documentation in them, not on all source files. ........ - - * /, 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 [r190729] Tilghman Lesher <tlesher@digium.com> - - * 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 ........ - -2009-04-27 19:15 +0000 [r190666] Russell Bryant <russell@digium.com> - - * res/res_smdi.c, /: Merged revisions 190663 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ................ - r190663 | russell | 2009-04-27 14:08:12 -0500 (Mon, 27 Apr 2009) - | 22 lines Merged revisions 190661-190662 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r190661 | russell | 2009-04-27 14:00:54 -0500 (Mon, 27 Apr 2009) - | 9 lines Resolve a crash in res_smdi when used with chan_dahdi. - When chan_dahdi goes to get an SMDI message, it provides no - search criteria. It just grabs the next message that arrives. - This code was written with the SMDI dialplan functions in mind, - since that is now the preferred method of using SMDI. However, - this broke support of it being used from chan_dahdi. (closes - AST-212) ........ r190662 | russell | 2009-04-27 14:03:59 -0500 - (Mon, 27 Apr 2009) | 2 lines Fix a typo from 190661. ........ - ................ - -2009-04-27 16:28 +0000 [r190625] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c, /: Merged revisions 190622 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r190622 | - mmichelson | 2009-04-27 11:26:14 -0500 (Mon, 27 Apr 2009) | 3 - lines Update warning message to not have pipes and contain all - options. ........ - -2009-04-23 21:23 +0000 [r190383] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 190371 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ ........ - -2009-04-23 20:44 +0000 [r190355] Tilghman Lesher <tlesher@digium.com> - - * main/pbx.c, /: Merged revisions 190352 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r190352 | - tilghman | 2009-04-23 15:42:11 -0500 (Thu, 23 Apr 2009) | 7 lines - Labels are sometimes (most of the time?) NULL for extensions. - (closes issue #14895) Reported by: chris-mac Patches: - 20090423__bug14895__2.diff.txt uploaded by tilghman (license 14) - Tested by: lmadsen ........ - -2009-04-23 19:18 +0000 [r190297] Joshua Colp <jcolp@digium.com> - - * channels/chan_local.c, /: Merged revisions 190287 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r190287 | file | 2009-04-23 16:15:30 -0300 (Thu, - 23 Apr 2009) | 13 lines Merged revisions 190286 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r190286 | file | 2009-04-23 16:13:18 -0300 (Thu, 23 Apr 2009) | 6 - lines Fix a bug in chan_local glare hangup detection. If both - sides of a Local channel were hung up at around the same time it - was possible for one thread to destroy the local private - structure and have the other thread immediately try to remove the - already freed structure from the local channel list. ........ - ................ - -2009-04-23 17:47 +0000 [r190253] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c, /: Merged revisions 190250 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r190250 | - mmichelson | 2009-04-23 12:45:35 -0500 (Thu, 23 Apr 2009) | 9 - lines Fix reversed behavior of leavewhenempty option in - queues.conf. (closes issue #14650) Reported by: alecdavis - Patches: 14650.patch uploaded by mmichelson (license 60) Tested - by: mmichelson, lmadsen ........ - -2009-04-22 21:43 +0000 [r190096] Tilghman Lesher <tlesher@digium.com> - - * /, configure, include/asterisk/autoconfig.h.in, configure.ac, - include/asterisk/lock.h: Merged revisions 190093 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r190093 | tilghman | 2009-04-22 16:38:15 -0500 - (Wed, 22 Apr 2009) | 14 lines Merged revisions 190092 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r190092 | tilghman | 2009-04-22 16:35:03 -0500 (Wed, 22 Apr 2009) - | 7 lines Detect availability of pthread_rwlock_timedwrlock() - before using it. (closes issue #14930) Reported by: tilghman - Patches: 20090420__bug14930.diff.txt uploaded by tilghman - (license 14) Tested by: mvanbaak, tilghman ........ - ................ - -2009-04-22 21:18 +0000 [r189997-190066] Jeff Peeler <jpeeler@digium.com> - - * main/cli.c, funcs/func_groupcount.c, /, main/app.c, - include/asterisk/channel.h: Merged revisions 190057 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r190057 | jpeeler | 2009-04-22 16:15:55 -0500 (Wed, 22 Apr 2009) - | 9 lines Fix building of chan_h323 with gcc-3.3 There seems to - be a bug with old versions of g++ that doesn't allow a structure - member to use the name list. Rename list member to group_list in - ast_group_info and change the few places it is used. (closes - issue #14790) Reported by: stuarth ........ - - * channels/h323/chan_h323.h, /, channels/h323/ast_h323.cxx, - channels/chan_h323.c: Merged revisions 189993 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r189993 | - jpeeler | 2009-04-22 14:23:49 -0500 (Wed, 22 Apr 2009) | 18 lines - Make chan_h323 respect packetization settings and fix small - reload issue. Previously, packetization settings were ignored and - now they are not. A new config option 'autoframing' has been - added to mirror the way chan_sip handles it. Turning on the - autoframing option (available both as a global option or per - peer) overrides the local settings with the remote packetization - settings. Testing was performed with varying packetization levels - with the following codecs: ulaw, alaw, gsm, and g729. Also, an - unrelated config reload issue has been fixed in the case of the - config file not changing. (closes issue #12415) Reported by: pj - Patches: 2009012200_h323packetization.diff.txt uploaded by - mvanbaak (license 7), modified by me ........ - -2009-04-22 18:01 +0000 [r189986] Russell Bryant <russell@digium.com> - - * /, main/features.c: Merged revisions 189951 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r189951 | - russell | 2009-04-22 11:56:43 -0500 (Wed, 22 Apr 2009) | 2 lines - Fix call parking callback. Pipes -> Commas. ........ - -2009-04-22 16:04 +0000 [r189816-189914] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_unistim.c, /: Merged revisions 189911 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r189911 | tilghman | 2009-04-22 11:01:30 -0500 (Wed, 22 Apr 2009) - | 7 lines Do not continue to receive DTMF, when the channel is - hungup and about to be destroyed. (closes issue #14858) Reported - by: barryf Patches: 20090421__bug14858.diff.txt uploaded by - tilghman (license 14) Tested by: barryf ........ - - * /, configure, configure.ac: Merged revisions 189813 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r189813 | tilghman | 2009-04-22 01:33:08 -0500 (Wed, 22 Apr 2009) - | 3 lines Detect liblua on SuSE, and add libm for linking for - Fedora. (Reported via the -dev list, Subject: Compiling Asterisk - with LUA) ........ - -2009-04-21 20:45 +0000 [r189775] David Vossel <dvossel@digium.com> - - * /, channels/chan_sip.c: Merged revisions 189771 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r189771 | - dvossel | 2009-04-21 15:28:37 -0500 (Tue, 21 Apr 2009) | 11 lines - Fixes segfault when switching UDP to TCP in sip.conf after - reload. If transport in sip.conf is switched from UDP to TCP, - Asterisk segfaults right after issuing a sip reload. The problem - is the socket type is changed to TCP but the fd may still be - present for UDP. Later, when the TCP session should be created or - set using an existing one, it isn't because the old file - descriptor is still present. Now every time transport is changed - during a sip.conf reload, the file descriptor is set to -1, - signifying it must be created or found. (closes issue #14727) - Reported by: pj Tested by: dvossel Review: - http://reviewboard.digium.com/r/229/ ........ - -2009-04-20 22:11 +0000 [r189540] Tilghman Lesher <tlesher@digium.com> - - * main/stdtime/localtime.c, /: Merged revisions 189539 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r189539 | tilghman | 2009-04-20 17:10:25 -0500 (Mon, 20 Apr 2009) - | 3 lines Use nanosleep instead of poll. This is not just because - mmichelson suggested it, but also because Mac OS X puked on my - poll(). ........ - -2009-04-20 21:41 +0000 [r189536] Terry Wilson <twilson@digium.com> - - * apps/app_dial.c, /: Merged revisions 189495,189516 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk - ................ r189495 | twilson | 2009-04-20 16:24:34 -0500 - (Mon, 20 Apr 2009) | 9 lines Merged revisions 189463 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r189463 | twilson | 2009-04-20 16:00:52 -0500 (Mon, 20 - Apr 2009) | 2 lines Don't treat a NOANSWER like a CHANUNAVAIL - ........ ................ r189516 | twilson | 2009-04-20 16:29:29 - -0500 (Mon, 20 Apr 2009) | 9 lines Merged revisions 189465 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r189465 | twilson | 2009-04-20 16:10:27 -0500 (Mon, 20 Apr 2009) - | 2 lines Update CDR appropriately when AST_CAUSE_NO_ANSWER is - set ........ ................ - -2009-04-20 21:36 +0000 [r189533] Sean Bright <sean.bright@gmail.com> - - * /, res/ael/ael.tab.c, res/ael/ael.y: Merged revisions 189464 via - svnmerge from https://origsvn.digium.com/svn/asterisk/trunk - ................ r189464 | seanbright | 2009-04-20 17:09:59 -0400 - (Mon, 20 Apr 2009) | 20 lines Merged revisions 189462 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r189462 | seanbright | 2009-04-20 16:58:39 -0400 (Mon, 20 Apr - 2009) | 13 lines Properly handle @s within hints in AEL. AEL was - not handling the case of a device hint containing an @ symbol, - which caused parking hints (e.g. hint(park:exten@context)) to - error out the parser. This patch makes AEL treat the @ the same - way it treats colon and ampersand now, meaning the characters are - included in verbatim. (closes issue #14941) Reported by: bpgoldsb - Patches: bug14941.patch uploaded by seanbright (license 71) - Tested by: bpgoldsb ........ ................ - -2009-04-20 17:11 +0000 [r189353] Joshua Colp <jcolp@digium.com> - - * /, 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 14:07 +0000 [r189281] Mark Michelson <mmichelson@digium.com> - - * 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 01:42 +0000 [r189207-189208] David Vossel <dvossel@digium.com> - - * 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/ ........ ................ - - * /, 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 21:56 +0000 [r189140] Richard Mudgett <rmudgett@digium.com> - - * 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 [r189105] Mark Michelson <mmichelson@digium.com> - - * /, 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 19:47 +0000 [r189081] Sean Bright <sean.bright@gmail.com> - - * 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 17:31 +0000 [r189068] Matthew Nicholson <mnicholson@digium.com> - - * 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 14:50 +0000 [r188941-188950] Joshua Colp <jcolp@digium.com> - - * /, 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 [r188777-188839] Tilghman Lesher <tlesher@digium.com> - - * /, 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 20:20 +0000 [r188474-188598] Mark Michelson <mmichelson@digium.com> - - * /, 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 17:46 +0000 [r188259-188416] Joshua Colp <jcolp@digium.com> - - * 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 05:47 +0000 [r188209-188213] Tilghman Lesher <tlesher@digium.com> - - * 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 [r188105] Mark Michelson <mmichelson@digium.com> - - * 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 16:34 +0000 [r188070] Joshua Colp <jcolp@digium.com> - - * /, 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 [r188039] Mark Michelson <mmichelson@digium.com> - - * 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 [r187916] Jeff Peeler <jpeeler@digium.com> - - * 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 17:31 +0000 [r187769] Tilghman Lesher <tlesher@digium.com> - - * 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 [r187724] Kevin P. Fleming <kpfleming@digium.com> - - * /, 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 16:05 +0000 [r187679] Tilghman Lesher <tlesher@digium.com> - - * /, 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 16:01 +0000 [r187677] Russell Bryant <russell@digium.com> - - * tests/test_sched.c, 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 03:57 +0000 [r187601] Tilghman Lesher <tlesher@digium.com> - - * main/audiohook.c, main/bridging.c, 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: 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 21:09 +0000 [r187564] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merge revision 187488 from trunk. - -2009-04-09 19:29 +0000 [r187531-187546] David Vossel <dvossel@digium.com> - - * 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) | 6 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) ........ - - * /, 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/ ........ - ................ - -2009-04-09 19:15 +0000 [r187496] Mark Michelson <mmichelson@digium.com> - - * 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 18:55 +0000 [r187051-187487] Tilghman Lesher <tlesher@digium.com> - - * 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 ........ ................ - - * /, 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. ........ ................ - - * /, funcs/func_odbc.c: Merged revisions 187050 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r187050 | - tilghman | 2009-04-08 12:08:43 -0500 (Wed, 08 Apr 2009) | 7 lines - If the first column is empty, output a delimiter anyway. (closes - issue #14848) Reported by: john8675309 Patches: - 20090408__bug14848.diff.txt uploaded by tilghman (license 14) - Tested by: john8675309 ........ - -2009-04-08 16:54 +0000 [r186988-187049] Mark Michelson <mmichelson@digium.com> - - * 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 12:39 +0000 [r186929] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 186928 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r186928 | - russell | 2009-04-08 07:35:57 -0500 (Wed, 08 Apr 2009) | 13 lines - Update some comments and resolve potential memory corruption in - chan_sip. While browsing chan_sip the other day, I noticed this - dangerous code in dialog_needdestroy(). This function is an - ao2_callback. It is absolutely _not_ okay to unlock the container - from within this function. It's also not clear why it was useful. - Given that it could cause memory corruption, I have removed it. - There was also a TODO comment left describing a potential - implementation of an improvement to the needdestroy handling. I'm - not convinced that what was described is the best choice here, so - I have briefly described the way that this function is used today - that could be improved. ........ - -2009-04-08 05:08 +0000 [r186901] Tilghman Lesher <tlesher@digium.com> - - * /, 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 [r186836-186845] Mark Michelson <mmichelson@digium.com> - - * formats/format_wav_gsm.c, /, formats/format_wav.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 [r186807] Tilghman Lesher <tlesher@digium.com> - - * /, 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 20:59 +0000 [r186723] Mark Michelson <mmichelson@digium.com> - - * 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 [r186469] Kevin P. Fleming <kpfleming@digium.com> - - * 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 20:05 +0000 [r186449] Tilghman Lesher <tlesher@digium.com> - - * 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 15:56 +0000 [r186324] Joshua Colp <jcolp@digium.com> - - * 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 15:19 +0000 [r186302] Tilghman Lesher <tlesher@digium.com> - - * main/stdtime/localtime.c, /: Merged revisions 186297 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r186297 | tilghman | 2009-04-03 10:18:28 -0500 (Fri, 03 Apr 2009) - | 4 lines Compatibility fix for glibc 2.4 (Closes issue #14820) - Reported by: phsultan ........ - -2009-04-03 14:34 +0000 [r186289] Mark Michelson <mmichelson@digium.com> - - * 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 02:11 +0000 [r186233] Russell Bryant <russell@digium.com> - - * 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 22:00 +0000 [r186178] Mark Michelson <mmichelson@digium.com> - - * 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 17:28 +0000 [r186111] Kevin P. Fleming <kpfleming@digium.com> - - * 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 [r186022-186063] Tilghman Lesher <tlesher@digium.com> - - * configs/sip.conf.sample, /, channels/chan_sip.c: 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 ........ - ................ ................ - - * main/strings.c, /: Merged revisions 186021 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r186021 | - tilghman | 2009-04-02 10:14:22 -0500 (Thu, 02 Apr 2009) | 7 lines - Missed a common case for needing to extend the buffer. (closes - issue #14716) Reported by: sum Patches: - 20090402__bug14716.diff.txt uploaded by tilghman (license 14) - Tested by: sum ........ - -2009-04-02 13:54 +0000 [r185957] Kevin P. Fleming <kpfleming@digium.com> - - * 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 22:44 +0000 [r185947] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/pbx.h, include/asterisk/strings.h, - main/taskprocessor.c, res/res_odbc.c, - include/asterisk/res_odbc.h, include/asterisk.h, main/strings.c, - main/manager.c, /, main/tdd.c, include/asterisk/astobj2.h, - main/ast_expr2f.c: Merged revisions 185912 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r185912 | - tilghman | 2009-04-01 15:13:28 -0500 (Wed, 01 Apr 2009) | 4 lines - Merge changes from str_substitution that are unrelated to that - branch. Included is a small bugfix to an ast_str helper, but most - of these changes are simply doxygen fixes. ........ - -2009-04-01 13:51 +0000 [r185775] Russell Bryant <russell@digium.com> - - * 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 [r185667] Kevin P. Fleming <kpfleming@digium.com> - - * 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 22:13 +0000 [r185472-185605] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c, /: Merged revisions 185604 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r185604 | - mmichelson | 2009-03-31 17:12:52 -0500 (Tue, 31 Mar 2009) | 3 - lines Fix trunk's compilation. ........ - - * 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 17:51 +0000 [r185428] David Brooks <dbrooks@digium.com> - - * 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: - <iq from='dbrooksjab@235-22-24-10/Telepathy' - to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'> - <session xmlns='http://www.google.com/session' - initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate' - id='1837267342'> <description - xmlns='http://www.google.com/session/phone'> <payload-type - clockrate='16000' name='speex' id='96'/> <payload-type - clockrate='8000' name='PCMA' id='8'/> <payload-type - clockrate='8000' name='PCMU' id='0'/> <payload-type - clockrate='90000' name='MPA' id='97'/> <payload-type - clockrate='16000' name='SIREN' id='98'/> <payload-type - clockrate='8000' name='telephone-event' id='99'/> </description> - </session> </iq> Review: http://reviewboard.digium.com/r/181/ - ........ ................ - -2009-03-31 14:59 +0000 [r185264] Russell Bryant <russell@digium.com> - - * 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 14:11 +0000 [r185200] Joshua Colp <jcolp@digium.com> - - * 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 20:52 +0000 [r185128-185129] Richard Mudgett <rmudgett@digium.com> - - * 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 16:52 +0000 [r185089] Mark Michelson <mmichelson@digium.com> - - * 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 14:43 +0000 [r184951] Joshua Colp <jcolp@digium.com> - - * /, 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 [r184913] Russell Bryant <russell@digium.com> - - * 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 05:56 +0000 [r184839-184846] Russell Bryant <russell@digium.com> - - * 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 19:21 +0000 [r184779] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_iax2.c, main/timing.c, main/channel.c, /, - bridges/bridge_softmix.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 18:12 +0000 [r184707-184729] Russell Bryant <russell@digium.com> - - * 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. ........ - - * apps/app_queue.c, apps/app_voicemail.c, main/cli.c, - include/asterisk/app.h, /, apps/app_dumpchan.c, main/app.c: - Merged revisions 184693 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r184693 | - russell | 2009-03-27 11:21:10 -0500 (Fri, 27 Mar 2009) | 2 lines - Change global_app_buf to ast_str_thread_global_buf. ........ - -2009-03-27 15:58 +0000 [r184650-184678] Joshua Colp <jcolp@digium.com> - - * /, bridges/bridge_softmix.c: Merged revisions 184677 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r184677 | file | 2009-03-27 12:57:28 -0300 (Fri, 27 Mar 2009) | 7 - lines Fix a potential timer leak in bridge_softmix. It is - possible for a bridge to be created without actually being used. - In that scenario a timing file descriptor would be opened and not - closed. To fix this the timing file descriptor is now closed in - the destroy callback, not the thread function. ........ - - * /, 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) ........ - - * /, bridges/bridge_softmix.c: Merged revisions 184639 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r184639 | file | 2009-03-27 11:18:40 -0300 (Fri, 27 Mar 2009) | 2 - lines Remove a cast that is not needed. ........ - -2009-03-27 14:09 +0000 [r184632] Russell Bryant <russell@digium.com> - - * 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 13:59 +0000 [r184612-184629] Joshua Colp <jcolp@digium.com> - - * /, bridges/bridge_softmix.c: Merged revisions 184628 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r184628 | file | 2009-03-27 10:57:29 -0300 (Fri, 27 Mar 2009) | 6 - lines Fix a potential race condition when creating a software - based mixing bridge. It was possible for no timer to become - available between creating the bridge and starting it. We now - open a timer when creating it and keep it open until the bridge - is destroyed. ........ - - * /, 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 02:35 +0000 [r184514-184552] Russell Bryant <russell@digium.com> - - * /, 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 [r184454] Kevin P. Fleming <kpfleming@digium.com> - - * 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 ........ - ................ - -2009-03-25 22:15 +0000 [r184343-184346] Russell Bryant <russell@digium.com> - - * /, 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. ........ - - * include/asterisk/_private.h, channels/chan_iax2.c, - channels/chan_dahdi.c, include/asterisk/event.h, - apps/app_minivm.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: 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/ ........ - -2009-03-25 19:27 +0000 [r184266-184283] Joshua Colp <jcolp@digium.com> - - * /, 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. - ........ - - * main/bridging.c, /: Merged revisions 183652 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r183652 | - file | 2009-03-22 18:00:28 -0300 (Sun, 22 Mar 2009) | 4 lines Fix - a minor logic flaw with the bridge generic thread. We only want - to move the channel pointers that are actually present. ........ - -2009-03-25 15:33 +0000 [r184256] Eliel C. Sardanons <eliels@gmail.com> - - * 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 14:40 +0000 [r184150-184221] Russell Bryant <russell@digium.com> - - * main/timing.c, /: Merged revisions 184219 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r184219 | - russell | 2009-03-25 09:33:32 -0500 (Wed, 25 Mar 2009) | 2 lines - Include poll-compat.h ........ - - * main/timing.c, /: Merged revisions 184151 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r184151 | - russell | 2009-03-24 21:03:13 -0500 (Tue, 24 Mar 2009) | 2 lines - Change poll() to ast_poll(). ........ - - * 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 [r184082] Mark Michelson <mmichelson@digium.com> - - * 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 22:02 +0000 [r184041-184044] Russell Bryant <russell@digium.com> - - * main/channel.c, /: Merged revisions 184043 via svnmerge from - https://origsvn.digium.com/svn/asterisk/trunk ........ r184043 | - russell | 2009-03-24 17:00:58 -0500 (Tue, 24 Mar 2009) | 2 lines - Put siren7 and siren14 in ast_best_codec() just so they're in - there somewhere. ........ - - * 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 15:29 +0000 [r183868-183917] Tilghman Lesher <tlesher@digium.com> - - * /, 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 19:00 +0000 [r183769] Mark Michelson <mmichelson@digium.com> - - * 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 [r183704] Leif Madsen <lmadsen@digium.com> - - * 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 17:09 +0000 [r183564] Russell Bryant <russell@digium.com> - - * 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. ........ ................ - -2009-03-20 12:19 +0000 [r183519] Eliel C. Sardanons <eliels@gmail.com> - - * channels/chan_dahdi.c, /: Merged revisions 183511 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r183511 | eliel | 2009-03-20 08:12:49 -0400 (Fri, 20 Mar 2009) | - 2 lines Remove duplicate <description> inside the xml - documentation. ........ - -2009-03-19 19:20 +0000 [r183337] Tilghman Lesher <tlesher@digium.com> - - * 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 18:20 +0000 [r183263] Russell Bryant <russell@digium.com> - - * 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 18:12 +0000 [r183247] Mark Michelson <mmichelson@digium.com> - - * 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 Leif Madsen <lmadsen@digium.com> - - * Release Asterisk 1.6.2.0-beta1 - -2009-03-19 16:11 +0000 [r183122] Mark Michelson <mmichelson@digium.com> - - * /, 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 15:45 +0000 [r183068-183111] Joshua Colp <jcolp@digium.com> - - * /, 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 [r183031] Jeff Peeler <jpeeler@digium.com> - - * /, 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 14:39 +0000 [r182947] Russell Bryant <russell@digium.com> - - * 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 20:53 +0000 [r182725] Jeff Peeler <jpeeler@digium.com> - - * 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 16:46 +0000 [r182592] Russell Bryant <russell@digium.com> - - * 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 15:01 +0000 [r182528-182534] Kevin P. Fleming <kpfleming@digium.com> - - * 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 [r182453] Tilghman Lesher <tlesher@digium.com> - - * 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 ........ ................ - -2009-03-17 02:02 +0000 [r182409] Richard Mudgett <rmudgett@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 182408 via svnmerge - from https://origsvn.digium.com/svn/asterisk/trunk ........ - r182408 | rmudgett | 2009-03-16 20:54:53 -0500 (Mon, 16 Mar 2009) - | 8 lines OPENR2 uses an incorrect string value if the extension - delimiter is not present. * Fixed OPENR2 using an incorrect - string value if the extension delimiter is not present in the - Dial() function. This was fixed for SS7 and PRI in trunk - -r172400. * Made OPENR2 stripmsd behavior the same as the SS7, - PRI, and others. * Removed trailing whitespace that appeared with - OPENR2. ........ - -2009-03-16 20:51 +0000 [r182360-182361] Russell Bryant <russell@digium.com> - - * /: svnmerge init - - * / (added): Create a branch for 1.6.2 - -2009-03-16 20:35 +0000 [r182355] Russell Bryant <russell@digium.com> - - * CREDITS, channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, - configure, include/asterisk/autoconfig.h.in, configure.ac, - CHANGES, makeopts.in: Add MFC/R2 support for chan_dahdi. This - commit introduces official support for R2 signaling in - chan_dahdi. The modifications to chan_dahdi, and the supporting - library, LibOpenR2, were both written by Moises Silva. Many users - are using this code, or a variant of it, in Asterisk 1.2, 1.4 and - 1.6 in Brazil, México and Argentina. An unknown number of users - (but at least 1) are using it in each of the following countries: - Colombia, Nepal, Thailand, Venezuela, Perú, and probably others. - To use this code, LibOpenR2 must be installed from - http://www.libopenr2.org/. Information about configuration can be - found in configs/chan_dahdi.conf.sample. The code committed is - the most up to date version, which was being maintained in - svn/asterisk/team/moy/mfcr2/. I would also like to include a - Thank You to the many others that tested this code beyond those - listed in this commit message. These are the names that I could - find in the mantis issue. (closes issue #12509) Reported by: moy - Patches: chan_zap-mfr2.patch uploaded by moy (license 222) Tested - by: moy, korihor, viniciusfontes, Skarmeth, loloski, - asbestoshead, titogarrido, heliocoelhojr, konsultex, ncorrare, - ecarruda, rtorresduque, PTorres, ychen Review: - http://reviewboard.digium.com/r/40/ - -2009-03-16 17:49 +0000 [r182282] David Vossel <dvossel@digium.com> - - * /, channels/chan_iax2.c: 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 17:33 +0000 [r182211-182278] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_env.c: 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 - - * channels/chan_local.c, /: 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 ........ - -2009-03-16 13:58 +0000 [r182171] Joshua Colp <jcolp@digium.com> - - * main/channel.c: 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 21:26 +0000 [r182029-182121] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Change faulty comparison used when announcing - average hold minutes and seconds (closes issue #14227) Reported - by: caspy - - * main/features.c: Remove ast_ prefix from functions which are not - public. - - * /, main/features.c: 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 17:25 +0000 [r182022] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: 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 16:55 +0000 [r181985] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: improve a bit of suboptimal code - -2009-03-13 01:26 +0000 [r181899] Richard Mudgett <rmudgett@digium.com> - - * /: Merged revisions 181898 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 Just - recording the v1.4 change in trunk since it originally came from - here. ........ r181898 | rmudgett | 2009-03-12 20:19:29 -0500 - (Thu, 12 Mar 2009) | 4 lines Use the correct branch integrated - property when generating the version string. Copied the - make_version file from Asterisk trunk. ........ - -2009-03-12 21:43 +0000 [r181769-181846] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Run the macro on the queue member's channel - when he answers, not the caller's channel. - - * /, channels/chan_sip.c: 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 17:32 +0000 [r181731] Tilghman Lesher <tlesher@digium.com> - - * main/translate.c: 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 16:56 +0000 [r181612-181665] Joshua Colp <jcolp@digium.com> - - * /, res/res_musiconhold.c: 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 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 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: Fix crash when sleep and retries argument was - not given to RetryDial application. (closes issue #14647) - Reported by: sherpya - -2009-03-12 01:33 +0000 [r181542-181577] Richard Mudgett <rmudgett@digium.com> - - * build_tools/make_version: Whitespace chages. - - * build_tools/make_version: Use the correct branch integrated - property when generating the version string - -2009-03-11 23:14 +0000 [r181499] Michiel van Baak <michiel@vanbaak.info> - - * configs/sip.conf.sample: Provide correct hint to debug SIP - trouble in the default config (closes issue #14646) Reported by: - strk - -2009-03-11 22:25 +0000 [r181465] Russell Bryant <russell@digium.com> - - * main/channel.c: Make handling of the BRIDGE_PLAY_SOUND variable - thread-safe. - -2009-03-11 22:20 +0000 [r181444] Jason Parker <jparker@digium.com> - - * /, configure, configure.ac: 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 22:14 +0000 [r181424-181428] Russell Bryant <russell@digium.com> - - * main/channel.c: Make handling of the BRIDGEPVTCALLID variable - thread-safe. - - * main/channel.c, /: 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 17:34 +0000 [r181371] David Vossel <dvossel@digium.com> - - * channels/iax2-parser.h, /, channels/chan_iax2.c: 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 17:26 +0000 [r181345] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: 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) ........ - -2009-03-11 17:04 +0000 [r181301] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/astobj2.h: Turn off malloc debugging of astobj2, - since it apparently doesn't work too well during startup. - -2009-03-11 16:40 +0000 [r181296] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: 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 16:19 +0000 [r181292] Russell Bryant <russell@digium.com> - - * doc/google-soc2009-ideas.txt: Replace contents of this doc with a - pointer to its new home - -2009-03-11 14:28 +0000 [r181244] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Fix segfault when dialing a typo'd queue If - trying to dial a non-existent queue, there would be a segfault - when attempting to access q->weight, even though q was NULL. This - problem was introduced during the queue-reset merge and thus only - affects trunk. (closes issue #14643) Reported by: alecdavis - -2009-03-11 13:44 +0000 [r181210] Joshua Colp <jcolp@digium.com> - - * apps/app_confbridge.c: Don't play the "you are about to be placed - into the conference" and "the leader has left the conference" - sounds if the quiet option is enabled. (reported by Vadim Lebedev - on the asterisk-dev list) - -2009-03-11 04:06 +0000 [r181135] Jeff Peeler <jpeeler@digium.com> - - * 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, - pbx/pbx_config.c: 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 02:25 +0000 [r181099] Russell Bryant <russell@digium.com> - - * doc/google-soc2009-ideas.txt: tabs to spaces - -2009-03-11 00:49 +0000 [r181032-181033] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: Add missing comment that quotes RFC 3891 - - * /, channels/chan_sip.c: 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. ........ - -2009-03-11 00:29 +0000 [r181027-181028] Tilghman Lesher <tlesher@digium.com> - - * main/strings.c, main/hashtab.c, include/asterisk/astobj2.h, - main/heap.c, include/asterisk/strings.h, - include/asterisk/hashtab.h, main/astobj2.c, - include/asterisk/heap.h: Add MALLOC_DEBUG to various utility - APIs, so that memory leaks can be tracked back to their source. - (related to issue #14636) - - * main/pbx.c: Spacing changes only - -2009-03-10 22:03 +0000 [r180944] Jason Parker <jparker@digium.com> - - * /, configure, configure.ac, autoconf/ast_prog_sed.m4, - autoconf/ast_check_gnu_make.m4: 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 22:03 +0000 [r180935-180942] Russell Bryant <russell@digium.com> - - * doc/google-soc2009-ideas.txt: Add some notes on getting in - contact with the dev community - - * doc/google-soc2009-ideas.txt: Remove difficulty and language - specifiers - - * doc/google-soc2009-ideas.txt: Expand upon documentation of - manager event project - -2009-03-10 21:15 +0000 [r180898] Michiel van Baak <michiel@vanbaak.info> - - * CHANGES: list the move of the astvarrundir from /var/run to - /var/run/asterisk (actually its $(localstatedir)/run/asterisk - Makes setups with asterisk as non-root easier to manage because - you can setup permissions on this dir instead of touching a file - and setting permissions on that. Files that come to mind are - asterisk.pid and asterisk.ctl socket. Prodded by and ok @russell - -2009-03-10 19:36 +0000 [r180859-180862] Russell Bryant <russell@digium.com> - - * doc/google-soc2009-ideas.txt: add more projects - - * doc/google-soc2009-ideas.txt: add more project ideas - -2009-03-10 14:40 +0000 [r180800] Joshua Colp <jcolp@digium.com> - - * main/manager.c: Reset the thread local string buffer when - handling the UserEvent action. (closes issue #14593) Reported by: - JimDickenson - -2009-03-09 22:00 +0000 [r180750] Russell Bryant <russell@digium.com> - - * doc/google-soc2009-ideas.txt: Add current mentors list, and first - pass on a project list broken out of "PineMango" I will work on - adding projects that have been sent to be via email tomorrow. - -2009-03-09 20:58 +0000 [r180719] Jeff Peeler <jpeeler@digium.com> - - * include/asterisk/rtp.h, include/asterisk/extconf.h, - main/devicestate.c, include/asterisk/tcptls.h, main/enum.c, - include/asterisk/callerid.h, include/asterisk/doxyref.h, - include/asterisk/event.h, include/asterisk/audiohook.h, - include/asterisk/dsp.h, include/asterisk/timing.h, - include/asterisk/udptl.h, include/asterisk/dlinkedlists.h, - include/asterisk/utils.h, include/asterisk/devicestate.h, - include/asterisk/taskprocessor.h, include/asterisk/enum.h, - include/asterisk/astobj2.h, include/asterisk/config.h, - include/asterisk/channel.h, include/asterisk/manager.h, - include/asterisk/heap.h, include/asterisk/logger.h, - include/asterisk/http.h, include/asterisk/res_odbc.h, - include/asterisk/app.h, main/tcptls.c, - include/asterisk/linkedlists.h, include/asterisk/sched.h, - include/asterisk/datastore.h, include/asterisk/lock.h, - include/asterisk/pbx.h, include/asterisk/dnsmgr.h: 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 <ver number> - <description of changes> 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/ - -2009-03-09 14:14 +0000 [r180684] Russell Bryant <russell@digium.com> - - * doc/google-soc2009-ideas.txt (added): Add skeleton for GSoC ideas - list - -2009-03-07 15:36 +0000 [r180641] Russell Bryant <russell@digium.com> - - * contrib/asterisk-ng-doxygen: Make some minor updates to the - doxygen configuration - add bridges directory to be processed - - add some res/ subdirs - alphabetize subdirs - use consistent - indentation - -2009-03-06 18:25 +0000 [r180579] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_voicemail.c: 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 17:26 +0000 [r180534] David Vossel <dvossel@digium.com> - - * /, main/enum.c: 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 23:26 +0000 [r180383-180465] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_voicemail.c: 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 ........ - - * /, configs/voicemail.conf.sample, apps/app_voicemail.c: 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 19:05 +0000 [r180382] Michiel van Baak <michiel@vanbaak.info> - - * Makefile: Make sure we terminate the first s| command so we can - actually produce correct files. - -2009-03-05 18:29 +0000 [r180373] Kevin P. Fleming <kpfleming@digium.com> - - * main/frame.c, /, include/asterisk/frame.h, main/rtp.c: 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/ - ........ - -2009-03-05 18:18 +0000 [r180369] Joshua Colp <jcolp@digium.com> - - * channels/chan_bridge.c (added), main/Makefile, - bridges/bridge_simple.c, bridges/bridge_softmix.c, - include/asterisk/channel.h, bridges/bridge_multiplexed.c, - CHANGES, Makefile, include/asterisk/bridging_technology.h - (added), bridges (added), bridges/bridge_builtin_features.c, - include/asterisk/bridging_features.h (added), - include/asterisk/bridging.h (added), apps/app_confbridge.c - (added), main/bridging.c (added), bridges/Makefile: Merge phase 1 - support for the new bridging architecture. This commit brings in - the bridging core, bridging technologies, and the ConfBridge - application. For usage information on the ConfBridge application - please see the output of "core show application ConfBridge" from - the CLI. For API documentation please see the doxygen page - describing the architecture and the documentation for each API - call. Review: http://reviewboard.digium.com/r/93/ - -2009-03-05 06:21 +0000 [r180304-180334] Tilghman Lesher <tlesher@digium.com> - - * contrib/editors/asterisk.vim: Also highlight the preamble and - postamble - - * contrib/editors/ael.vim (added), contrib/editors/asterisk.vim - (added), contrib/editors (added), contrib/editors/asteriskvm.vim - (added): Add syntax coloring files for Vim, including a new one - for AEL - -2009-03-04 21:01 +0000 [r180261] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: 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 20:48 +0000 [r180259] Tilghman Lesher <tlesher@digium.com> - - * main/aescrypt.c, main/abstract_jb.c, main/acl.c, main/app.c, - main/alaw.c: Spacing changes only - -2009-03-04 19:24 +0000 [r180195] Joshua Colp <jcolp@digium.com> - - * /, main/callerid.c: 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) ........ - -2009-03-04 17:03 +0000 [r180155] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c, configs/sip.conf.sample: Allow for "magic" - pickups to work when we wish to ignore the context When the - subscription context for a call pickup subscription differs from - the context of the call pickup target, there's not an easy way to - divine what context should be used for the pickup. The way to - work around this is to use PICKUPMARK as the context for the - pickup. This has been documented in the sip.conf.sample file - (ABE-1708) closes issue #14567 submitted by: alecdavis - -2009-03-04 14:39 +0000 [r180120] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c: 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 23:35 +0000 [r180079] Steve Murphy <murf@digium.com> - - * utils/Makefile: My bad! left check_expr2 in the ALL_UTILS list by - mistake. Already done to 1.6.x - -2009-03-03 23:21 +0000 [r180032] David Vossel <dvossel@digium.com> - - * main/channel.c, include/asterisk/app.h, apps/app_read.c, - main/app.c: 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 22:49 +0000 [r180007] Mark Michelson <mmichelson@digium.com> - - * /, configs/queues.conf.sample, apps/app_queue.c: 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 ........ - -2009-03-03 22:12 +0000 [r179973] Steve Murphy <murf@digium.com> - - * utils/Makefile, utils/expr2.testinput, /, main/ast_expr2.h, - main/ast_expr2.y, main/ast_expr2f.c, main/ast_expr2.fl, - main/ast_expr2.c: 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 22:01 +0000 [r179972] David Vossel <dvossel@digium.com> - - * apps/app_meetme.c: app_meetme not setting filename and fileformat - correctly for realtime When app_meetme finds a realtime - conference, it doesn't get the filename and fileformat correctly - when 'r' is set. Now app_meetme first checks to see if fileformat - and filename are declared in the db, if they're not it checks the - .conf file, if its not declared there either it then uses - defaults. (closes issue #14545) Reported by: dalbaech Patches: - app_meetme-realtime5.patch uploaded by dvossel (license 671) - Realtime_Conference_Record_workaround.txt uploaded by dalbaech - (license 705) Tested by: dvossel, dalbaech Review: - http://reviewboard.digium.com/r/180/ - -2009-03-03 20:59 +0000 [r179937] Mark Michelson <mmichelson@digium.com> - - * res/res_timing_dahdi.c, doc/timing.txt (added): 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 20:02 +0000 [r179903] Brian Degenhardt <bmd@digium.com> - - * apps/app_directed_pickup.c: fix a leaked channel lock (and future - deadlock) when we try to pick up our own channel - -2009-03-03 18:28 +0000 [r179841] Joshua Colp <jcolp@digium.com> - - * /, main/features.c: 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 17:03 +0000 [r179745] Mark Michelson <mmichelson@digium.com> - - * pbx/pbx_spool.c: Convert pbx_spool to use string fields instead - of statically-sized buffers. In tests run after making this - conversion, I noticed an approximate 85% reduction in memory - usage for call file processing. Review: - http://reviewboard.digium.com/r/168/ - -2009-03-03 16:47 +0000 [r179742] Russell Bryant <russell@digium.com> - - * main/channel.c, /: 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 15:13 +0000 [r179675] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Please prefix default values with DEFAULT - -2009-03-03 14:40 +0000 [r179672] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /: 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 13:54 +0000 [r179609] Russell Bryant <russell@digium.com> - - * main/channel.c, /: 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 00:01 +0000 [r179537] Jeff Peeler <jpeeler@digium.com> - - * main/channel.c, /: 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 23:36 +0000 [r179533] Russell Bryant <russell@digium.com> - - * /, apps/app_meetme.c: 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 23:10 +0000 [r179469] Tilghman Lesher <tlesher@digium.com> - - * /, main/app.c: 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 23:06 +0000 [r179462-179465] Russell Bryant <russell@digium.com> - - * res/res_timing_timerfd.c: Fix a reference leak in - timerfd_set_rate(). (found during a debugging session with - dvossel and mmichelson.) - - * main/channel.c, /: 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 20:16 +0000 [r179396] Jason Parker <jparker@digium.com> - - * /, main/editline/configure, main/editline/np/unvis.c, - main/editline/sys.h, main/editline/configure.in: 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 17:18 +0000 [r179361] Tilghman Lesher <tlesher@digium.com> - - * cdr/cdr_sqlite3_custom.c: Backport 1.6.0 fix to trunk (failsafe - if db is not loaded) - -2009-03-02 14:28 +0000 [r179291-179323] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Do not try to remove a registration - scheduled item if the scheduler context has already been - destroyed. (closes issue #14580) Reported by: alecdavis - - * main/audiohook.c: 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 23:25 +0000 [r179219-179254] Mark Michelson <mmichelson@digium.com> - - * apps/app_speech_utils.c: Swap reversed timevals. This was pointed - out by ScribbleJ in #asterisk-dev. Thanks very much, ScribbleJ! - - * channels/chan_sip.c: 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 21:47 +0000 [r179164] Russell Bryant <russell@digium.com> - - * res/res_ais.c, doc/distributed_devstate.txt, - configs/ais.conf.sample: Mark res_ais as experimental, as the - binary event format is subject to change. - -2009-02-27 21:32 +0000 [r179161] Tilghman Lesher <tlesher@digium.com> - - * cdr/cdr_sqlite3_custom.c: If config file is blank, don't load - module. (Closes issue #14563) - -2009-02-27 21:23 +0000 [r179154] Russell Bryant <russell@digium.com> - - * UPGRADE.txt: Add a note about the ordering of entries in sip.conf - in 1.6.1. - -2009-02-27 20:34 +0000 [r179122] Michiel van Baak <michiel@vanbaak.info> - - * channels/chan_skinny.c: Add reload support to chan_skinny. - Special thanks goes to DEA who had to redo this patch twice - because we first put unload/load support in and later redid the - way we configure devices and lines. (closes issue #10297) - Reported by: DEA Patches: skinny-reload-trunkv2.diff uploaded by - wedhorn (license 30) skinny-reload-trunk-v4.txt uploaded by DEA - (license 3) With mods by me based on feedback from wedhorn and - Russell and seanbright Tested by: DEA, mvanbaak, pj Review: - http://reviewboard.digium.com/r/130/ - -2009-02-27 19:04 +0000 [r179057] Jason Parker <jparker@digium.com> - - * doc/tex/channelvariables.tex: 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 15:51 +0000 [r179021] Russell Bryant <russell@digium.com> - - * sounds/Makefile: Fix downloading SIREN7 and SIREN14 sound - packages. In passing, also fix downloading SLIN16 extra sound - packages. (closes issue #14565) Reported by: jtodd - -2009-02-27 03:45 +0000 [r178986] Steve Murphy <murf@digium.com> - - * /, main/features.c, configs/features.conf.sample: 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 18:41 +0000 [r178919] Tilghman Lesher <tlesher@digium.com> - - * main/features.c, CHANGES, configs/features.conf.sample: Sound - confirmation of call pickup success. (closes issue #13826) - Reported by: azielke Patches: pickupsound2-trunk.patch uploaded - by azielke (license 548) __20081124_bug_13826_updated.patch - uploaded by lmadsen (license 10) Tested by: lmadsen - -2009-02-26 17:46 +0000 [r178871] David Vossel <dvossel@digium.com> - - * channels/chan_iax2.c: IAX2 prune realtime, minor tweak to last - fix A return statement was missing which caused unexpected cli - output. issue #14479 - -2009-02-26 17:45 +0000 [r178828-178870] Steve Murphy <murf@digium.com> - - * apps/app_osplookup.c, apps/app_rpt.c: These small fixes prevent - compiler warnings with ubuntu 8.10's gcc-4.3.2, which tend to - break my dev-mode build. Not a problem in 1.6.x. - - * /, main/features.c: 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 16:42 +0000 [r178801] Joshua Colp <jcolp@digium.com> - - * main/file.c: 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 15:50 +0000 [r178767] David Vossel <dvossel@digium.com> - - * channels/chan_iax2.c: 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 <friend> 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/ - -2009-02-26 15:40 +0000 [r178764] Joshua Colp <jcolp@digium.com> - - * main/indications.c: Ensure there is a valid tone part before - trying to play tones. (closes issue #14558) Reported by: - alecdavis - -2009-02-26 15:02 +0000 [r178733] Olle Johansson <oej@edvina.net> - - * configs/res_snmp.conf.sample: Clarifications on the different - models and reference to further docs. - -2009-02-26 13:39 +0000 [r178703-178704] Kevin P. Fleming <kpfleming@digium.com> - - * README: another minor commit to test post-commit script changes - (now testing post-revprop-change as well, third try) - - * README: minor commit to test post-commit script changes - -2009-02-25 19:49 +0000 [r178573-178607] Tilghman Lesher <tlesher@digium.com> - - * main/stdtime/localtime.c: Picky, picky buildbots - - * configure, include/asterisk/autoconfig.h.in, configure.ac, - main/stdtime/localtime.c: Use notification when timezone files - change and re-scan then. (closes issue #14300) Reported by: - jamessan Patches: 20090127__bug14300.diff.txt uploaded by - tilghman (license 14) 20090224__bug14300.diff uploaded by - jamessan (license 246) Tested by: jamessan Review: - http://reviewboard.digium.com/r/136/ - - * res/res_odbc.c: Oops, wrong direction of command - -2009-02-25 12:45 +0000 [r178509] Russell Bryant <russell@digium.com> - - * /, main/asterisk.c: 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 23:27 +0000 [r178375-178446] Tilghman Lesher <tlesher@digium.com> - - * /, configs/extensions.conf.sample: 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: Apparently, a void cast doesn't override - warn_unused_result. - - * main/asterisk.c: The 3 possible errors with pipe(2) are all - impossible in this situation. - -2009-02-24 20:39 +0000 [r178374] Russell Bryant <russell@digium.com> - - * /, main/rtp.c: 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 20:06 +0000 [r178303-178342] Tilghman Lesher <tlesher@digium.com> - - * utils/astcanary.c, main/asterisk.c: Use a SIGPIPE to kill the - process, instead of depending upon the astcanary process being - inherited by init. - - * utils/astcanary.c: 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 17:42 +0000 [r178300] David Vossel <dvossel@digium.com> - - * doc/manager_1_1.txt, CHANGES, channels/chan_iax2.c: Allows - manager command to see if IAX link is trunked and encrypted. - Displays what kind of encryption is enabled as well. Manager - command "iaxpeers" now shows if a link is trunked and encrypted. - Instead of encryption saying simply "yes" or "no", it now - displays what type of encryption is enabled and if keyrotation is - on or not. (closes issue #14427) Reported by: snuffy Patches: - iax_show_trunks.diff uploaded by snuffy (license 35) - 2009022200_iax2_show_trunkencryption.diff.txt uploaded by - mvanbaak (license 7) Tested by: mvanbaak, dvossel, snuffy Review: - http://reviewboard.digium.com/r/173/ - -2009-02-24 15:18 +0000 [r178213] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: 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 23:11 +0000 [r178142] Russell Bryant <russell@digium.com> - - * /, main/rtp.c: 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 ........ - -2009-02-23 21:02 +0000 [r178107] Tilghman Lesher <tlesher@digium.com> - - * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: - Permit emailsubject and emailbody to be set per mailbox. (closes - issue #14372) Reported by: fhackenberger Patches: - voicemail_individual_subject_and_body_1.6.1 uploaded by - fhackenberger (license 592) with additional fixes by Corydon76 - (license 14) - -2009-02-23 18:23 +0000 [r178061] Michiel van Baak <michiel@vanbaak.info> - - * channels/chan_skinny.c: update the new manager commands in - chan_skinny to match chan_sip's headers. requested by oej. - -2009-02-23 17:59 +0000 [r178030] David Vossel <dvossel@digium.com> - - * channels/chan_iax2.c: Changes the way keyrotation is enabled by - default Key rotation was enabled by default by setting the global - encryption method to IAX_ENCRYPT_KEYROTATE. the problem with this - is that if encryption is not enabled, and the encryption method - is set to anything except 0, the peer appears to have encryption - enabled when issuing a "iax2 show peers". Rather than have the - key rotation bit always set by default, it is now only set when - an encryption method is enabled. (closes issue #14523) Reported - by: mvanbaak - -2009-02-23 17:48 +0000 [r178027] Michiel van Baak <michiel@vanbaak.info> - - * CHANGES: list the addition of the SKINNY manager actions in the - CHANGES file. - -2009-02-23 17:29 +0000 [r178022] Russell Bryant <russell@digium.com> - - * tests/test_sched.c, main/sched.c: Fix a regression in scheduler - entry ordering, and add a regression test for it. (closes issue - #14522) Reported by: pj Tested by: russell - -2009-02-22 23:04 +0000 [r177988] Michiel van Baak <michiel@vanbaak.info> - - * doc/manager_1_1.txt, channels/chan_skinny.c: Add a couple of - manager commands to chan_skinny Added: SKINNYdevices - SKINNYshowdevice SKINNYlines SKINNYshowline (closes issue #14521) - Reported by: mvanbaak Review: - http://reviewboard.digium.com/r/170/ - -2009-02-21 15:59 +0000 [r177944] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_sip.c: On update, test against the existence of - sipregs. - -2009-02-21 14:37 +0000 [r177913] Michiel van Baak <michiel@vanbaak.info> - - * main/asterisk.c: add extra check for sysinfo/sysctl (closes issue - #14513) Reported by: snuffy Patches: bug14513_fixsysinfo.diff - uploaded by snuffy (license 35) - -2009-02-21 14:16 +0000 [r177884] Sean Bright <sean.bright@gmail.com> - - * main/hashtab.c, include/asterisk/hashtab.h: Trailing whitespace, - minor coding guideline fixes, and start beefing up the hashtab - documentation a bit. - -2009-02-21 13:17 +0000 [r177855] Russell Bryant <russell@digium.com> - - * include/asterisk/indications.h: Fix build issues on Solaris and - OpenBSD. (closes issue #14512) Reported by: snuffy - -2009-02-21 13:13 +0000 [r177849-177852] Michiel van Baak <michiel@vanbaak.info> - - * Makefile, contrib/init.d/rc.debian.asterisk, - contrib/init.d/rc.archlinux.asterisk, - contrib/scripts/safe_asterisk: set - ASTVARRUNDIR=$(localstatedir)/run/asterisk as default path When - running asterisk as non-root and without this patch the pidfile - wants to go into /var/run/asterisk.pid. This directory is not - writable for the non-root user and changing permissions is not an - option. Putting it in /var/run/asterisk/asterisk.pid makes it - possible to set permissions on the /var/run/asterisk dir so - everything works as it should be. Patched committed is based on - pabelanger's patch. (closes issue #13153) Reported by: pabelanger - Patches: 2009012900_bug13153-nonrootscripts.diff.txt uploaded by - mvanbaak (license 7) Review: http://reviewboard.digium.com/r/139/ - - * channels/chan_sip.c: make chan_sip.c compile on OpenBSD again. - -2009-02-20 23:02 +0000 [r177732-177787] Tilghman Lesher <tlesher@digium.com> - - * main/pbx.c, /: 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 ........ - - * /, include/asterisk/threadstorage.h: 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 20:29 +0000 [r177699] Dwayne M. Hubbard <dwayne.hubbard@gmail.com> - - * apps/app_fax.c: 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 17:29 +0000 [r177661-177664] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/app.h, main/app.c, apps/app_system.c: 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 - - * apps/app_voicemail.c: Oops, merge broke trunk - -2009-02-20 00:35 +0000 [r177624] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_sip.c: Set sip_request ast_str data to NULL so - ast_str_copy allocates space properly in copy_request (issue - #14478) Reported by: erik_dedecker - -2009-02-19 23:56 +0000 [r177595] Steve Murphy <murf@digium.com> - - * /, main/Makefile, main/ast_expr2f.c: 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 22:33 +0000 [r177506-177537] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_voicemail.c: 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 ........ - - * doc/database_transactions.txt (added): Document how to use - database transactions - -2009-02-19 16:45 +0000 [r177387] Jeff Peeler <jpeeler@digium.com> - - * include/asterisk/channel.h: Fix another merge error from 176708 - -2009-02-19 16:38 +0000 [r177384] Joshua Colp <jcolp@digium.com> - - * /, apps/app_speech_utils.c: 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 15:56 +0000 [r177356] Jeff Peeler <jpeeler@digium.com> - - * main/features.c: Fix mismerge from revision 176708 pointed out by - Kaloyan Kovachev on the asterisk-dev mailing list. Thanks! - -2009-02-19 00:26 +0000 [r177320] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/res_odbc.h, funcs/func_odbc.c, CHANGES, - res/res_odbc.c, configs/res_odbc.conf.sample: ODBC transaction - support - -2009-02-19 00:08 +0000 [r177291] Joshua Colp <jcolp@digium.com> - - * CHANGES: Update CHANGES file to include MWI subscription support - that was added some time ago. - -2009-02-18 23:51 +0000 [r177287] Tilghman Lesher <tlesher@digium.com> - - * main/strings.c: Handle negative length and eliminate a condition - that is always true. - -2009-02-18 23:50 +0000 [r177286] Steve Murphy <murf@digium.com> - - * /, res/ael/ael.tab.c, res/ael/ael.y: 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 23:09 +0000 [r177229] Kevin P. Fleming <kpfleming@digium.com> - - * main/frame.c: 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 22:51 +0000 [r177226] David Vossel <dvossel@digium.com> - - * main/features.c: 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 20:11 +0000 [r177162] Jeff Peeler <jpeeler@digium.com> - - * channels/h323/compat_h323.cxx, autoconf/ast_check_pwlib.m4, - channels/h323/cisco-h225.h, channels/h323/caps_h323.cxx, - channels/h323/ast_h323.cxx, channels/h323/ast_ptlib.h (added), - configure, channels/h323/compat_h323.h, configure.ac, - channels/h323/caps_h323.h, autoconf/ast_prog_sed.m4, - channels/h323/ast_h323.h, channels/h323/chan_h323.h, - channels/h323/cisco-h225.cxx: 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 19:12 +0000 [r177101] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c: 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 19:05 +0000 [r177098] Tilghman Lesher <tlesher@digium.com> - - * /, include/asterisk/config.h: 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 17:24 +0000 [r177035] Doug Bailey <dbailey@digium.com> - - * main/utils.c: Fixed error where a check for an zero length, - terminated string was needed. - -2009-02-18 17:11 +0000 [r177005] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: 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 16:09 +0000 [r176948] Doug Bailey <dbailey@digium.com> - - * main/utils.c: Need to take into account the \0 terminator of the - old string to determine the amount available. - -2009-02-18 15:35 +0000 [r176943] Steve Murphy <murf@digium.com> - - * main/pbx.c: 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 06:14 +0000 [r176901-176904] Russell Bryant <russell@digium.com> - - * include/asterisk/heap.h: Add example code for a heap traversal. - - * main/pbx.c: 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 02:55 +0000 [r176869] Dwayne M. Hubbard <dwayne.hubbard@gmail.com> - - * channels/chan_sip.c: T38 faxdetect should jump to the 'fax' - extension for incoming calls only The previous implementation of - T38 faxdetect resulted in both sides of the call jumping to a fax - extension when both sides had 't38pt_udptl=yes' and - 'faxdetect=yes' in sip.conf and a 'fax' extension in the current - context. This revision will jump to a 'fax' extension on incoming - calls only. - -2009-02-18 02:02 +0000 [r176841] Kevin P. Fleming <kpfleming@digium.com> - - * main/rtp.c: suppress smoothers for Siren codecs as well as Speex - and G.723.1 - -2009-02-17 22:52 +0000 [r176771] Russell Bryant <russell@digium.com> - - * apps/app_milliwatt.c: Remove a dependency that no longer exists. - -2009-02-17 22:28 +0000 [r176760] Shaun Ruffell <sruffell@digium.com> - - * codecs/codec_dahdi.c: 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 22:08 +0000 [r176708] Jeff Peeler <jpeeler@digium.com> - - * main/channel.c, /, main/features.c, include/asterisk/channel.h: - 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 22:02 +0000 [r176706] Mark Michelson <mmichelson@digium.com> - - * tests/test_sched.c: Use constants from inttypes.h to clear up - 32-bit compilation errors - -2009-02-17 21:59 +0000 [r176705] Dwayne M. Hubbard <dwayne.hubbard@gmail.com> - - * channels/chan_sip.c: 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 21:40 +0000 [r176697] Mark Michelson <mmichelson@digium.com> - - * include/asterisk/frame.h: Clear up documentation of - AST_FRIENDLY_OFFSET in frame.h - -2009-02-17 21:23 +0000 [r176669] Tilghman Lesher <tlesher@digium.com> - - * /: Recorded merge of revisions 176661 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r176661 | tilghman | 2009-02-17 15:21:41 -0600 (Tue, 17 Feb 2009) - | 9 lines Backport change to 1.4: 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 21:22 +0000 [r176666] Russell Bryant <russell@digium.com> - - * main/channel.c, res/res_timing_pthread.c, res/res_timing_dahdi.c, - res/res_timing_timerfd.c, include/asterisk/timing.h, - main/timing.c: 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 21:14 +0000 [r176642] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_local.c: 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 21:04 +0000 [r176632-176639] Russell Bryant <russell@digium.com> - - * tests/test_sched.c (added), main/sched.c: Significantly improve - scheduler performance under high load. This patch changes the - scheduler to use a max-heap to store pending scheduler entries - instead of a fully sorted doubly linked list. When the number of - entries in the scheduler gets large, this will perform much - better. For much more detailed information on this change, see - the review request. Review: http://reviewboard.digium.com/r/160/ - - * tests/test_heap.c (added): Add a test module for the heap - implementation. Review: http://reviewboard.digium.com/r/160/ - - * main/Makefile, main/heap.c (added), include/asterisk/heap.h - (added): 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/ - -2009-02-17 20:50 +0000 [r176631] Olle Johansson <oej@edvina.net> - - * include/asterisk/config.h: Typo - -2009-02-17 20:41 +0000 [r176627] Russell Bryant <russell@digium.com> - - * channels/chan_unistim.c, main/pbx.c, apps/app_read.c, - configs/indications.conf.sample, apps/app_playtones.c (added), - include/asterisk/indications.h, apps/app_readexten.c, - apps/app_disa.c, UPGRADE.txt, include/asterisk/channel.h, - include/asterisk/_private.h, main/indications.c, main/loader.c, - main/channel.c, channels/chan_misdn.c, channels/chan_skinny.c, - funcs/func_channel.c, res/snmp/agent.c, main/app.c, - res/res_indications.c (removed), main/asterisk.c: Merge a large - set of updates to the Asterisk indications API. This patch - includes a number of changes to the indications API. The primary - motivation for this work was to improve stability. The object - management in this API was significantly flawed, and a number of - trivial situations could cause crashes. The changes included are: - 1) Remove the module res_indications. This included the critical - functionality that actually loaded the indications configuration. - I have seen many people have Asterisk problems because they - accidentally did not have an indications.conf present and loaded. - Now, this code is in the core, and Asterisk will fail to start - without indications configuration. There was one part of - res_indications, the dialplan applications, which did belong in a - module, and have been moved to a new module, app_playtones. 2) - Object management has been significantly changed. Tone zones are - now managed using astobj2, and it is no longer possible to crash - Asterisk by issuing a reload that destroys tone zones while they - are in use. 3) The API documentation has been filled out. 4) The - API has been updated to follow our naming conventions. 5) Various - bits of code throughout the tree have been updated to account for - the API update. 6) Configuration parsing has been mostly - re-written. 7) "Code cleanup" The code is from - svn/asterisk/team/russell/indications/. Review: - http://reviewboard.digium.com/r/149/ - -2009-02-17 18:49 +0000 [r176592] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_odbc.c, res/res_odbc.c: Add assertions in the quest to - track down a refcount leak. (closes issue #14485) Reported by: - davevg - -2009-02-17 17:33 +0000 [r176557] Russell Bryant <russell@digium.com> - - * main/pbx.c, apps/app_queue.c: 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 17:28 +0000 [r176513-176556] Olle Johansson <oej@edvina.net> - - * configs/extconfig.conf.sample: Typo - - * main/config.c: If there are no realtime engines, there's no - reason to check for realtime families - -2009-02-17 14:39 +0000 [r176360-176501] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_sip.c: In this version, we can combine the queries, - because we support dropping nonexistent columns. - - * /, channels/chan_sip.c: 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 ........ - - * main/strings.c: Might want to update the buffer pointer after a - realloc (or we crash) (closes issue #14485) Reported by: davevg - -2009-02-16 23:37 +0000 [r176356] Kevin P. Fleming <kpfleming@digium.com> - - * sounds/sounds.xml: add support for Siren7 and Siren14 flavors of - prompts and music on hold - -2009-02-16 23:33 +0000 [r176355] David Vossel <dvossel@digium.com> - - * /, channels/chan_iax2.c: 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 23:14 +0000 [r176320] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_skinny.c: 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 21:45 +0000 [r176255] Kevin P. Fleming <kpfleming@digium.com> - - * /, main/utils.c, include/asterisk/stringfields.h: 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 21:40 +0000 [r176253] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_meetme.c: 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 21:30 +0000 [r176248] David Vossel <dvossel@digium.com> - - * channels/chan_iax2.c: 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 18:25 +0000 [r176174] Mark Michelson <mmichelson@digium.com> - - * main/logger.c: 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 17:44 +0000 [r176138] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_dahdi.c: Can't set debug level 2 (intense - debugging) unless the syntax matches - -2009-02-16 17:09 +0000 [r176100] Russell Bryant <russell@digium.com> - - * channels/chan_features.c (removed): Remove chan_features. Review: - http://reviewboard.digium.com/r/161/ - -2009-02-16 15:36 +0000 [r176030] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: 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 02:54 +0000 [r175983] Russell Bryant <russell@digium.com> - - * main/channel.c: Make the causes array static, and remove the type - name as it is not needed. - -2009-02-16 00:26 +0000 [r175952] Michiel van Baak <michiel@vanbaak.info> - - * channels/chan_unistim.c, /, channels/chan_sip.c, - include/asterisk/manager.h, doc/unistim.txt: 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 21:27 +0000 [r175829-175882] Russell Bryant <russell@digium.com> - - * include/asterisk/sched.h, main/sched.c: Make ast_sched_report() - and ast_sched_dump() thread safe. - - * channels/chan_sip.c, include/asterisk/sched.h, main/sched.c: 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. - -2009-02-15 20:39 +0000 [r175783-175827] Olle Johansson <oej@edvina.net> - - * formats/format_ilbc.c, /: Merged revisions 175825 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r175825 | oej | 2009-02-15 21:33:17 +0100 (Sön, 15 Feb - 2009) | 2 lines format_ilbc does not depend on codec libraries - and can therefore always be made. My mistake. Ursäkta! ........ - - * formats/format_ilbc.c, /: Merged revisions 175792 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r175792 | oej | 2009-02-15 21:20:21 +0100 (Sön, 15 Feb - 2009) | 2 lines Disable format_ilbc.so by default, like - codec_ilbc.so ........ - - * /, channels/chan_sip.c: Merged revisions 175777 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r175777 | oej | 2009-02-15 20:48:38 +0100 (Sön, 15 Feb 2009) | 2 - lines Make sure that the debug line is not printed on debug level - 0 ........ - -2009-02-13 20:57 +0000 [r175655-175663] Mark Michelson <mmichelson@digium.com> - - * doc/manager_1_1.txt, CHANGES, apps/app_queue.c: Merge queue-reset - branch to Asterisk From a user point-of-view, this adds new CLI - commands and Manager Actions to better facilitate the reloading - of queues and the resetting of their statistics. The new CLI - commands are the "queue reload" and "queue reset stats" commands. - The new manager actions are the QueueReload and QueueReset - commands. Review: http://reviewboard.digium.com/r/115 - - * doc/manager_1_1.txt, apps/app_chanspy.c: Add manager events for - chanspy starting or stopping (closes issue #14469) Reported by: - caio1982 Patches: chanspy_events2.diff uploaded by caio1982 - (license 22) - -2009-02-13 20:26 +0000 [r175623-175636] Russell Bryant <russell@digium.com> - - * res/res_jabber.c: fix a few more XML documentation problems - - * main/pbx.c: add missing </para> - -2009-02-13 20:11 +0000 [r175597] David Vossel <dvossel@digium.com> - - * configs/iax.conf.sample, channels/iax2.h, channels/chan_iax2.c: - 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. Review: - http://reviewboard.digium.com/r/159/ - -2009-02-13 19:49 +0000 [r175591] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_voicemail.c: 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 16:41 +0000 [r175549] Joshua Colp <jcolp@digium.com> - - * apps/app_record.c: Add an option to keep the recorded file upon - hangup. (closes issue #14341) Reported by: fnordian - -2009-02-13 13:41 +0000 [r175508-175512] Kevin P. Fleming <kpfleming@digium.com> - - * CHANGES: document G.722.1/.1C support - - * main/frame.c, channels/chan_sip.c, include/asterisk/rtp.h, - channels/chan_h323.c, include/asterisk/frame.h, - formats/format_siren14.c (added), main/rtp.c, - formats/format_siren7.c (added): Add basic (passthrough, - playback, record) support for ITU G.722.1 and G.722.1C (also - known as Siren7 and Siren14) This patch adds passthrough, file - recording and file playback support for the codecs listed above, - with negotiation over SIP/SDP supported. Due to Asterisk's - current limitation of treating a codec/bitrate combination as a - unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are - supported. Along the way, some related work was done: 1) The - rtpPayloadType structure definition, used as a return result for - an API call in rtp.h, was moved from rtp.c to rtp.h so that the - API call was actually usable. The only previous used of the API - all was chan_h323.c, which had a duplicate of the structure - definition instead of doing it the right way. 2) The hardcoded - SDP sample rates for various codecs in chan_sip.c were removed, - in favor of storing these sample rates in rtp.c along with the - codec definitions there. A new API call was added to allow - retrieval of the sample rate for a given codec. 3) Some basic - 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip - *must* decline any media streams offered for these codecs that - are not at the bitrates that we support (otherwise Bad Things - (TM) would result). Review: http://reviewboard.digium.com/r/158/ - -2009-02-13 04:22 +0000 [r175411-175475] Dwayne M. Hubbard <dwayne.hubbard@gmail.com> - - * CHANGES: add 'faxbuffers' configuration option information to - CHANGES - - * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Add - dynamic fax buffer configuration option to chan_dahdi.conf When - the 'faxdetect' configuration option is used, one may also want - to use the 'faxbuffers' configuration option in chan_dahdi.conf. - This option will dynamically use the configured 'faxbuffers' - buffer policy on a channel for the life of the call following the - detection of fax tones. The faxbuffers buffer policy will be - reverted during call teardown. An example use of 'faxbuffers' is - below. This example would switch to using 6 buffers with a full - buffer policy. faxbuffers=>6,full - -2009-02-12 21:41 +0000 [r175368] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Remove useless string copy, and make sscanf - safe again - -2009-02-12 21:27 +0000 [r175344] David Vossel <dvossel@digium.com> - - * configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: Adds - force encryption option to iax.conf This patch adds - forceencryption=yes as an iax.conf option. When force encryption - is enabled, no unencrypted connections are allowed. This insures - all connections are encrypted. This is a new feature, so CHANGES - and iax.conf.sample are updated as well. (closes issue #13285) - Reported by: sgofferj Tested by: russell Review: - http://reviewboard.digium.com/r/150/ - -2009-02-12 21:25 +0000 [r175334] Tilghman Lesher <tlesher@digium.com> - - * main/udptl.c, /: 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 20:48 +0000 [r175298] Jeff Peeler <jpeeler@digium.com> - - * /, main/features.c: 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 20:45 +0000 [r175255-175295] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Avoid using ast_strdupa() in a loop. - - * build_tools/cflags.xml: 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 18:48 +0000 [r175250] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_iax2.c: correct warning message to not refer - specifically to DAHDI - -2009-02-12 18:00 +0000 [r175188] Jeff Peeler <jpeeler@digium.com> - - * /, main/features.c: 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 17:07 +0000 [r175127] David Vossel <dvossel@digium.com> - - * channels/chan_iax2.c: 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 16:57 +0000 [r175125] Russell Bryant <russell@digium.com> - - * /, main/rtp.c: 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 16:28 +0000 [r175121] Mark Michelson <mmichelson@digium.com> - - * include/asterisk/astobj2.h, main/astobj2.c: 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. - -2009-02-12 14:25 +0000 [r175058-175089] Philippe Sultan <philippe.sultan@gmail.com> - - * channels/chan_gtalk.c: Issue a warning message if our candidate's - IP is the loopback address. (closes issue #13985) Reported by: - jcovert Tested by: phsultan - - * /, channels/chan_gtalk.c: 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 ........ - -2009-02-11 23:12 +0000 [r174945-174951] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Fix a bit of odd logic for announcing position. - Sync with 1.6.0's logic - - * apps/app_queue.c: 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: 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 14:44 +0000 [r174844] Joshua Colp <jcolp@digium.com> - - * main/channel.c: 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 23:17 +0000 [r174764-174805] Mark Michelson <mmichelson@digium.com> - - * apps/app_chanspy.c: 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: 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 20:15 +0000 [r174710] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Only decrease inringing count if above zero. - (issue #13238) Reported by: kowalma - -2009-02-10 19:38 +0000 [r174705] Kevin P. Fleming <kpfleming@digium.com> - - * main/slinfactory.c, include/asterisk/slinfactory.h: improve - slinfactory API to remove implicit sample rate and require - explicit sample rate selection by creator of the slinfactory - -2009-02-10 18:16 +0000 [r174584] Matthew Nicholson <mnicholson@digium.com> - - * /, main/jitterbuf.c: 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 17:48 +0000 [r174543-174580] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Set the type for the peer structure to be a - peer as the default. (closes issue #14447) Reported by: triccyx - - * channels/chan_sip.c: 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 07:06 +0000 [r174470-174503] Tilghman Lesher <tlesher@digium.com> - - * apps/app_stack.c, apps/app_voicemail.c: Fix0ring build - - * apps/app_stack.c: Remove the usage of the KeepAlive app, as it no - longer exists. - -2009-02-10 04:49 +0000 [r174370-174435] Steve Murphy <murf@digium.com> - - * apps/app_rpt.c: This patch removes the use of AST_PBX_KEEPALIVE - from app_rpt.c. (closes issue #14435) Reported by: D_McNaul - - * apps/app_rpt.c: More intptr_t work. - - * /, apps/app_rpt.c: 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 17:27 +0000 [r174327] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: Fix something I messed up in the merge I - just did - -2009-02-09 17:26 +0000 [r174325] David Vossel <dvossel@digium.com> - - * apps/app_externalivr.c: 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 17:20 +0000 [r174301] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: 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 14:49 +0000 [r174219] Joshua Colp <jcolp@digium.com> - - * /, res/res_musiconhold.c: 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 16:16 +0000 [r174149] Russell Bryant <russell@digium.com> - - * /, res/snmp/agent.c: 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. ........ - -2009-02-06 23:51 +0000 [r174084] Dwayne M. Hubbard <dwayne.hubbard@gmail.com> - - * /, channels/chan_sip.c: 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 20:12 +0000 [r174046] David Vossel <dvossel@digium.com> - - * configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: Adds - immediate yes/no option to iax.conf This is very similar to the - DAHDI immediate=yes option. When the phone is picked up, instead - of giving a dialtone it connects directly to the "s" extension. - Changes where implemented in chan_iax2.c to directly connect to - the "s" extension in the appropriate context when this option is - enabled. Examples explaining its use are added to - iax2.conf.sample. CHANGES has been updated as well. (closes issue - #14266) Reported by: jcovert Patches: chan_iax2.c.patch-trunk - uploaded by jcovert (license 551) iax.conf.sample.patch uploaded - by jcovert (license 551) Tested by: jcovert, dvossel Review: - http://reviewboard.digium.com/r/143/ - -2009-02-06 19:28 +0000 [r173974-174041] Joshua Colp <jcolp@digium.com> - - * channels/chan_dahdi.c: 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 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 16:28 +0000 [r173952] Matthew Nicholson <mnicholson@digium.com> - - * /, channels/chan_sip.c: 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 15:59 +0000 [r173902] Joshua Colp <jcolp@digium.com> - - * main/audiohook.c, apps/app_chanspy.c: 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 10:55 +0000 [r173848-173858] Russell Bryant <russell@digium.com> - - * include/asterisk/sched.h, channels/chan_iax2.c, main/sched.c: Add - a common implementation of a scheduler context with a dedicated - thread. This commit expands the Asterisk scheduler API to include - a common implementation of a scheduler context being processed by - a dedicated thread. chan_iax2 has been updated to use this new - code. Also, as a result, this resolves some race conditions - related to the previous chan_iax2 scheduler handling. Related to - rev 171452 which resolved the same issues in 1.4. Code from - team/russell/sched_thread2 Review: - http://reviewboard.digium.com/r/129/ - - * main/manager.c: Resolve a memory leak that would occur on an - invalid channel given to Action: Status - -2009-02-05 23:48 +0000 [r173773-173776] Mark Michelson <mmichelson@digium.com> - - * configs/extensions.conf.sample: 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 - - * apps/app_voicemail.c: 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 21:00 +0000 [r173697] Jeff Peeler <jpeeler@digium.com> - - * /, apps/app_voicemail.c: 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 20:30 +0000 [r173693] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: 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 19:36 +0000 [r173657] Tilghman Lesher <tlesher@digium.com> - - * res/res_config_sqlite.c: Change the first field, or we don't get - the necessary field separation. - -2009-02-05 18:48 +0000 [r173507-173593] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_mixmonitor.c: 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 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: 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 21:26 +0000 [r173503] Tilghman Lesher <tlesher@digium.com> - - * res/res_jabber.c: Add XML documentation for the applications and - functions in res_jabber (closes issue #14405) Reported by: snuffy - Patches: xml_jabber.diff uploaded by snuffy (license 35) - -2009-02-04 21:25 +0000 [r173502] David Vossel <dvossel@digium.com> - - * channels/iax2-parser.h, channels/chan_iax2.c: 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 21:17 +0000 [r173500] Jeff Peeler <jpeeler@digium.com> - - * /, main/features.c, include/asterisk/features.h: 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 18:48 +0000 [r173458] Tilghman Lesher <tlesher@digium.com> - - * main/tcptls.c: 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 17:45 +0000 [r173354-173397] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_chanspy.c: 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 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: 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 00:43 +0000 [r173311] Tilghman Lesher <tlesher@digium.com> - - * main/pbx.c, pbx/pbx_config.c: 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 17:35 +0000 [r173169] Richard Mudgett <rmudgett@digium.com> - - * channels/chan_dahdi.c: Broke up the large conditional blocks so - it is easy to see if a function is compiled. - -2009-02-03 00:29 +0000 [r173104-173130] Tilghman Lesher <tlesher@digium.com> - - * configure, include/asterisk/autoconfig.h.in, configure.ac, - main/xml.c, include/asterisk/compiler.h, apps/app_stack.c, - include/asterisk/optional_api.h: 1. Make OS X compile cleanly - with app_stack. 2. Use curl to download sound files, as curl is - installed natively on OS X, whereas wget and fetch are not. - (closes issue #14332) Reported by: oej Tested by: Corydon76 - - * /, configs/extensions.conf.sample: 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 ........ - -2009-02-02 23:57 +0000 [r173067] Terry Wilson <twilson@digium.com> - - * /, main/features.c: 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 23:21 +0000 [r173028-173047] Mark Michelson <mmichelson@digium.com> - - * main/manager.c, CHANGES: Reverting commit number 173028 as there - are some potential issues - - * main/manager.c, CHANGES: Add a CLI command to log out a manager - user (closes issue #13877) Reported by: eliel Patches: - cli_manager_logout.patch.txt uploaded by eliel (license 64) - Tested by: eliel, putnopvut - -2009-02-02 20:40 +0000 [r172963] Richard Mudgett <rmudgett@digium.com> - - * /: Recorded merge of revisions 172962 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r172962 | rmudgett | 2009-02-02 14:28:54 -0600 (Mon, 02 Feb 2009) - | 11 lines channels/chan_dahdi.c * Added doxygen comments to the - major dahdi structures. * Fixed PRI using an incorrect string - value if the extension delimiter is not present in the Dial() - function. * Fixed some uninitialized string variables on FXS - ports. configs/chan_dahdi.conf.sample * Updated some - documentation. These changes are already in trunk -r172400 - ........ - -2009-02-02 19:02 +0000 [r172929] Steve Murphy <murf@digium.com> - - * apps/app_dial.c, main/features.c, CHANGES, - include/asterisk/features.h: This reverts the changes I made for - 11583; will reviewboard this before committing again... reopened - 11583 until all Russell's issues are resolved. - -2009-02-02 18:13 +0000 [r172894] Leif Madsen <lmadsen@digium.com> - - * configs/res_ldap.conf.sample: 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 - -2009-02-02 17:37 +0000 [r172890] Steve Murphy <murf@digium.com> - - * apps/app_dial.c, main/features.c, CHANGES, - include/asterisk/features.h: This change allows the disconnect - feature (as in "one-touch" in features.c) to be used within the - dial app, before a call is bridged. Many thanks to sobomax for - submitting this patch. Quoting from bug 11582: "So the goal of - the patch was to use the user configured feature code during the - call setup phase. The original ast_feature_interpret() function - is not well suited for this purpose as it uses much call bridge - specific data and doesn't separate a detection of feature from a - feature handler call. So a new function ast_feature_detect() has - been extracted off the ast_feature_interpret() function but - keeping the original logic intact except some insignificant - changes to locking. "Having created the ast_feature_detect() - function the possibility to use feature detection in almost any - place of the asterisk code. So a call to this function has been - added to wait_for_answer() function of app_dial.so module. This - code doesn't call the feature handler however and uses old call - leg disconnect logic to make the changes as small and simple as - possible to prevent unexpected problems. A disconnect feature - currently is the only one supported during call setup as other - features as call parking and call transfer don't make much sense - during call setup. However if need in some of the features would - arise it is much easier to implement as the infrastructure - changes are already in place with this patch." I have cleaned up - the patch somewhat, and verified that the existing functionality - is not harmed, and that the new functionality works. Terry has - committed his stuff, and there were no conflicts (see 14274). - (closes issue #11583) Reported by: sobomax Patches: - patch-apps__app_dial.c uploaded by sobomax (license 359) - patch-include__asterisk__features.h uploaded by sobomax (license - 359) patch-res__res_features.c uploaded by sobomax (license 359) - enable-features-during-call-setup.diff uploaded by sobomax - (license 359) 11583.newdiff uploaded by murf (license 17) - enable-features-during-call-setup-1.diff uploaded by sobomax - (license 359) 11583.latest-patch uploaded by murf (license 17) - Tested by: sobomax, murf - -2009-02-02 16:42 +0000 [r172855] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Fix a spelling mistake. - -2009-02-02 10:46 +0000 [r172816-172818] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Add a todo. I do need to really check what's - going on with this kill-the-user business ;-) Why do we suddenly - have two flags to set peer type? - - * channels/chan_sip.c: Small formatting change - - * channels/chan_sip.c: Add some well-needed improvements to the - wishlist in the code, so that we can close some bug reports. - -2009-02-02 01:41 +0000 [r172778] Sean Bright <sean.bright@gmail.com> - - * channels/chan_sip.c: The CID lookup feature wasn't actually - working properly with dialog-info+xml supporting devices. The - devices (snoms, specifically) need to receive a SIP URI instead - of just an extension. This adds that functionality. - -2009-02-01 02:44 +0000 [r172706-172741] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c: Blank argument crashes Asterisk (closes - issue #14377) Reported by: amorsen - - * funcs/func_strings.c: 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) - -2009-01-30 22:22 +0000 [r172598] Mark Michelson <mmichelson@digium.com> - - * include/asterisk/channel.h: Fix redefinition of flag in channel.h - -2009-01-30 21:50 +0000 [r172580-172581] Terry Wilson <twilson@digium.com> - - * configs/features.conf.sample: Remove incorrect line from sample - config - - * apps/app_dial.c, main/global_datastores.c, main/features.c, - include/asterisk/global_datastores.h, CHANGES, - configs/features.conf.sample: 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 18:36 +0000 [r172441-172548] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_aes.c: Parameter position reversed in documentation - - * /, autoconf/ast_func_fork.m4, configure, main/app.c, - apps/app_rpt.c, main/asterisk.c: 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 23:15 +0000 [r172370-172440] Richard Mudgett <rmudgett@digium.com> - - * main/cli.c: Remove tabs from comment - - * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: - 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. - - * include/asterisk/say.h: Fixed some doxygen comments - -2009-01-29 17:10 +0000 [r172318-172319] Olle Johansson <oej@edvina.net> - - * channels/chan_local.c: Revert two lines that was extra, but only - on fridays. - - * apps/app_dial.c, channels/chan_local.c, channels/chan_sip.c, - include/asterisk/causes.h, apps/app_queue.c: Fix "cancel answered - elsewhere" through app_queue with members in chan_local. Also, - implement a private cause code (as suggested by Tilghman). This - works with chan_sip, but doesn't propagate through chan_local. - -2009-01-29 16:48 +0000 [r172315] Tilghman Lesher <tlesher@digium.com> - - * configs/func_odbc.conf.sample: Better document mode=multirow, - based upon a conversation with Jared. - -2009-01-29 13:47 +0000 [r172271] Leif Madsen <lmadsen@digium.com> - - * contrib/scripts/realtime_pgsql.sql: 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 13:24 +0000 [r172173-172270] Olle Johansson <oej@edvina.net> - - * configs/sip.conf.sample, CHANGES: Update documentation - - * include/asterisk/app.h, channels/chan_sip.c, main/app.c: - Make - sure we set setvar= variables on outbound calls too, not only - inbound calls. - Also, change a function in app.c to return a - userful value instead of always returning 0. Patch by fnordian, - changed by Corydon76 and myself. This does not close the bug - report, as fnordian had an additional change we're still - discussing. (related to issue #14059) Reported by: fnordian - Patches: chan_sip_hfield.patch uploaded by fnordian (license 110) - 20090116__bug14059.diff.txt uploaded by Corydon76 (license 14) - Tested by: fnordian, Corydon76, oej - - * channels/chan_sip.c: 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 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 ........ - -2009-01-28 22:52 +0000 [r172132] Steve Murphy <murf@digium.com> - - * channels/chan_misdn.c: A further correction: cast the sizeof to - an int. - -2009-01-28 22:48 +0000 [r172131] Tilghman Lesher <tlesher@digium.com> - - * res/res_config_odbc.c: 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 21:48 +0000 [r172063-172099] Steve Murphy <murf@digium.com> - - * channels/chan_misdn.c: my gcc (Ubuntu 4.3.2-1ubuntu11) 4.3.2 - didn't like the \%ld and issued a warning, breaking my dev-mode - build. This fixes it. - - * apps/app_channelredirect.c, main/pbx.c, main/manager.c, /, - main/features.c, include/asterisk/channel.h: 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 17:27 +0000 [r171964] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_dahdi.c, /: 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 14:39 +0000 [r171838-171925] Olle Johansson <oej@edvina.net> - - * CHANGES: Yep. Documentation is important. - - * apps/app_queue.c: Add final part of previously committed work for - answered elsewhere in queue - the missing piece that started with - app_dial() earlier on. This is to avoid having the list and - counter of missed calls being touched by queue calls. Add the C - option to queue() and nothing will be logged on phones that - support the Reason: header on SIP cancel, like the SNOM phones. - - * configs/sip.conf.sample: Add some more notes about device - matching. - - * /, configs/sip.conf.sample: 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. - ........ - -2009-01-28 00:17 +0000 [r171797] Mark Michelson <mmichelson@digium.com> - - * funcs/func_aes.c: Fix some signedness problems in func_aes.c - -2009-01-27 23:28 +0000 [r171793] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_dahdi.c: Don't complain about lack of D-channels on - PTMP connections - -2009-01-27 22:43 +0000 [r171757] David Vossel <dvossel@digium.com> - - * funcs/func_aes.c (added), CHANGES: Adding AES_ENCRYPT and - AES_DECRYPT dialplan functions. (closes issue #14301) Reported - by: amorsen review: http://reviewboard.digium.com/r/128/ - -2009-01-27 21:58 +0000 [r171618-171691] Mark Michelson <mmichelson@digium.com> - - * channels/chan_agent.c: 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 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: 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 15:23 +0000 [r171558] Doug Bailey <dbailey@digium.com> - - * channels/chan_dahdi.c: Handle new VMWI ioctl structure (Now there - are two VMWI ioctl calls.) (issue #14104) Reported by: alecdavis - Tested by: dbailey - -2009-01-27 15:00 +0000 [r171263-171528] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: 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 - ........ - - * channels/chan_sip.c: Moving generic setting to friends - - * channels/chan_sip.c: Continue to move variables into the sip_cfg - structure to make them easier to handle in the future as a group - of settings for a group of devices. At some point, I want one - sip_cfg per domain handled, so we can have "group" settings. - - * channels/chan_sip.c: Just moving around variable declarations so - that we have all globals in the same place. Default setting is - set before we activate the channel or at reloads, not where we - declare the variable. - - * /, channels/chan_sip.c: 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 ........ - - * main/channel.c: Add extensions and context on manager event when - new channel is created. - -2009-01-25 23:58 +0000 [r171188] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_oss.c: 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 16:50 +0000 [r171043-171081] Michiel van Baak <michiel@vanbaak.info> - - * channels/chan_skinny.c: dont segfault when a MWI event occurs on - a line without a registered device - - * configs/skinny.conf.sample: Make the sample skinny.conf work - (closes issue #14325) Reported by: DEA Patches: - skinny.conf.sample-trunk.txt uploaded by DEA (license 3) - -2009-01-25 13:35 +0000 [r170980] Sean Bright <sean.bright@gmail.com> - - * /, apps/app_page.c: 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 02:49 +0000 [r170902-170943] Russell Bryant <russell@digium.com> - - * include/asterisk/utils.h: 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. - - * res/res_agi.c: Add a todo to finish the XML docs in this module - -2009-01-24 13:55 +0000 [r170837] Tilghman Lesher <tlesher@digium.com> - - * /, configs/res_odbc.conf.sample: 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 23:10 +0000 [r170794] Richard Mudgett <rmudgett@digium.com> - - * doc/tex/Makefile: Fix asterisk.pdf generation if branch name has - an underscore in it. - -2009-01-23 22:58 +0000 [r170790] Russell Bryant <russell@digium.com> - - * doc/tex/Makefile: Don't blow up if a branch name has an - underscore in it - -2009-01-23 20:56 +0000 [r170677-170720] Mark Michelson <mmichelson@digium.com> - - * /, configs/res_odbc.conf.sample: 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 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 20:18 +0000 [r170652] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /: 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 19:25 +0000 [r170608] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_iax2.c: 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 19:09 +0000 [r170505-170569] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c, /: 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 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 17:46 +0000 [r170501] Michiel van Baak <michiel@vanbaak.info> - - * channels/chan_h323.c: let's use SENTINEL where needed - -2009-01-23 17:32 +0000 [r170498] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Reset the ast_str used for escape - substitution. We need to do this since it is a thread local - variable that may contain the value of a previous substitution. - (closes issue #14312) Reported by: pj - -2009-01-23 17:03 +0000 [r170463] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_dahdi.c: We should not do restart messages if we're - in PTMP mode - -2009-01-23 16:57 +0000 [r170460] Michiel van Baak <michiel@vanbaak.info> - - * channels/chan_skinny.c: Dont clear the display of skinny phones - when not needed. (closes issue #13182) Reported by: pj Patches: - 2009011901_dontcleardisplay.diff.txt uploaded by mvanbaak - (license 7) Tested by: mvanbaak, pj - -2009-01-23 16:35 +0000 [r170457] Doug Bailey <dbailey@digium.com> - - * 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 15:44 +0000 [r170393] Mark Michelson <mmichelson@digium.com> - - * main/channel.c, /: 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 23:23 +0000 [r170351] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_dahdi.c: Make sure we don't set the channel to be - inalarm for a D-channel drop on PTMP connections - -2009-01-22 21:25 +0000 [r170307] Tilghman Lesher <tlesher@digium.com> - - * main/abstract_jb.c: Create logfile safely. (closes issue #14160) - Reported by: tzafrir Patches: 20090104__bug14160.diff.txt - uploaded by Corydon76 (license 14) - -2009-01-22 20:04 +0000 [r170240] Joshua Colp <jcolp@digium.com> - - * /, main/rtp.c: 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 17:19 +0000 [r170165] Tilghman Lesher <tlesher@digium.com> - - * /, pbx/pbx_config.c: 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 16:52 +0000 [r170148] Joshua Colp <jcolp@digium.com> - - * /, apps/app_meetme.c: 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 ........ - -2009-01-22 15:49 +0000 [r170112] Doug Bailey <dbailey@digium.com> - - * channels/chan_dahdi.c, configure, - include/asterisk/autoconfig.h.in, configure.ac: change VMWI to - use new DAHDI_VMWI ioctl call. Change configure script to detect - the new ioctl call data structure. (issue #14104) Reported by: - alecdavis Patches: mwiioctl_structure_asterisk.diff4.txt uploaded - by dbailey (license ) Tested by: alecdavis, dbailey - -2009-01-22 15:14 +0000 [r170047-170051] Joshua Colp <jcolp@digium.com> - - * main/pbx.c, /: 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: 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 02:10 +0000 [r170007] Richard Mudgett <rmudgett@digium.com> - - * channels/chan_dahdi.c: * Adjust some conditionals to balance - curly braces. * Other minor changes. - -2009-01-22 00:44 +0000 [r169944] Tilghman Lesher <tlesher@digium.com> - - * /, include/asterisk/linkedlists.h: 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 ........ - -2009-01-22 00:23 +0000 [r169910] Richard Mudgett <rmudgett@digium.com> - - * channels/chan_dahdi.c: Whitespace changes only - -2009-01-21 23:25 +0000 [r169869] Joshua Colp <jcolp@digium.com> - - * main/pbx.c, /: 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 23:20 +0000 [r169794-169866] Mark Michelson <mmichelson@digium.com> - - * channels/chan_dahdi.c: Test commit for test issue #14303 - - * main/say.c: 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 - -2009-01-21 22:04 +0000 [r169793] Michiel van Baak <michiel@vanbaak.info> - - * doc/tex/extensions.tex: remove duplicated sentence. - -2009-01-21 21:53 +0000 [r169791] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: 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 21:03 +0000 [r169723] Tilghman Lesher <tlesher@digium.com> - - * /, main/asterisk.c: 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 17:21 +0000 [r169673] Steve Murphy <murf@digium.com> - - * utils/refcounter.c: 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 [r169620-169625] Russell Bryant <russell@digium.com> - - * /: Remove properties that erroneously got merged into trunk - - * main/tcptls.c: 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 00:33 +0000 [r169557-169611] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: 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: Use the default timeout for a queue instead of - -1 (closes issue #14272) Reported by: timking - - * /, channels/chan_sip.c: 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 19:22 +0000 [r169486-169510] Terry Wilson <twilson@digium.com> - - * main/features.c: 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 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 21:42 +0000 [r169438] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/res_odbc.h, funcs/func_odbc.c, - include/asterisk/strings.h, res/res_odbc.c: ast_str_SQLGetData is - *not* part of the ast_str API, it's part of the ast_odbc API and - just happens to use an ast_str as the buffer; move all of it to - res_odbc.c and res_odbc.h, renaming appropriately along the way - fix some minor coding style issues in strings.h and add some - attribute_pure annotations to functions in the ast_str API - -2009-01-19 20:14 +0000 [r169367-169369] Michiel van Baak <michiel@vanbaak.info> - - * main/asterisk.c: fix assignment in swapmode plug. Spotted and fix - provided by ys (closes issue #14129) Reported by: ys Tested by: - ys - - * channels/chan_skinny.c: Redo the event-based MWI in chan_skinny. - Dan saw regular segfaults with the old implementation and rewrote - it to make it really eventbased. I altered it to be trunk - compatible and wedhorn gave some feedback and ideas how to make - it even better. (closes issue #13821) Reported by: DEA Patches: - chan_skinny-mwi-events.txt uploaded by DEA (license 3) Tested by: - mvanbaak, DEA "no probs by me" from wedhorn - -2009-01-19 20:05 +0000 [r169365] Tilghman Lesher <tlesher@digium.com> - - * main/manager.c, /, apps/app_userevent.c: 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 18:36 +0000 [r169327] Michiel van Baak <michiel@vanbaak.info> - - * main/asterisk.c: Make asterisk compile on non-amd64 versions of - OpenBSD. The HW_PHYSMEM64 is only available in latest OpenBSD - and/or amd64 versions of OpenBSD. Use HW_PHYSMEM when - HW_PHYSMEM64 is not available. (closes issue #14129) Reported by: - ys Patches: 2009011600_physmem64.diff.txt uploaded by mvanbaak - (license 7) Tested by: mvanbaak, jtodd - -2009-01-19 18:22 +0000 [r169277-169325] Doug Bailey <dbailey@digium.com> - - * channels/chan_dahdi.c: Get rid of magic number and replace with - DAHDI_VMWI_NUMBER_MASK when determining the number of messages - pending for MWI call - - * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Add - enhanced MWI generation to take advantage of new dahdi line - reversal MWI ability. (closes issue #14104) Reported by: - alecdavis Patches: asttrunk-14104.diff2.txt uploaded by dbailey - (license ) chan_dahdi.rpas_and_fsk.diff.txt uploaded by alecdavis - (license 585) Tested by: alecdavis, dbailey - -2009-01-19 15:54 +0000 [r169211] Mark Michelson <mmichelson@digium.com> - - * channels/chan_local.c, /: 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 18:26 +0000 [r169153] Doug Bailey <dbailey@digium.com> - - * 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 02:52 +0000 [r169116] Sean Bright <sean.bright@gmail.com> - - * pbx/pbx_dundi.c: Change intializer types. Found while working on - asterisk-cpp. I have a new favorite error message from g++: - pbx_dundi.c:4580: sorry, unimplemented: non-trivial designated - initializers not supported I like it when compilers are - apologetic. - -2009-01-17 01:56 +0000 [r169044-169080] Terry Wilson <twilson@digium.com> - - * main/tcptls.c, main/http.c, include/asterisk/tcptls.h: 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: 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 - -2009-01-16 22:43 +0000 [r168976] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: 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 22:16 +0000 [r168941] Terry Wilson <twilson@digium.com> - - * /, main/features.c: 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 19:54 +0000 [r168898] Mark Michelson <mmichelson@digium.com> - - * res/res_timing_timerfd.c: Fix a logic error that occur when using - the timerfd interface This sequence of events posed a problem - timerfd_timer_open timerfd_timer_enable_continuous - timerfd_timer_set_rate timerfd_timer_disable_continuous The - reason was that the timing module was written under the - assumption that timerfd_timer_set_rate would not be called - between enabling and disabling continuous mode. What happened in - this situation was that timerfd_timer_enable_continuous saved off - our previously set timer (in this situation a 0 timer, meaning it - never runs out). Then timerfd_timer_disable_continuous would - restore this 0 timer, even though it logically should set the - timer to be whatever was set in timerfd_timer_set_rate. Now the - behavior in timerfd_timer_set_rate is to overwrite the saved - timer that may or may not have been set in - timerfd_timer_enable_continuous. Even if - timerfd_timer_enable_continuous has not been previously called, - this will not harm the operation. Thanks to Terry Wilson for - discovering the problem and giving me a really great debug - capture that pointed out the problem clearly - -2009-01-16 18:49 +0000 [r168832] Tilghman Lesher <tlesher@digium.com> - - * /, main/say.c, include/asterisk/say.h, apps/app_voicemail.c: - 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 17:09 +0000 [r168759-168760] Russell Bryant <russell@digium.com> - - * CHANGES: Fix a spelling mistake. - - * channels/chan_misdn.c: build in dev mode - -2009-01-16 00:34 +0000 [r168737-168746] Steve Murphy <murf@digium.com> - - * res/ael/pval.c, /: 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.h, main/ast_expr2.y, main/ast_expr2.c: 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 20:18 +0000 [r168734] Kevin P. Fleming <kpfleming@digium.com> - - * res/res_config_odbc.c, build_tools/menuselect-deps.in, configure, - funcs/func_odbc.c, configure.ac, cdr/cdr_adaptive_odbc.c, - cdr/cdr_odbc.c, makeopts.in, res/res_odbc.c, - apps/app_voicemail.c: remove the PBX_ODBC logic from the - configure script, and add GENERIC_ODCB logic that includes - copying the relevant LIB and INCLUDE data from either UnixODBC or - iODBC, based on which was found; if both were found, prefer - UnixODBC this stops modules from being linked against both sets - of libraries on systems that have both installed - -2009-01-15 20:00 +0000 [r168725-168732] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: Add missing brace - - * channels/chan_sip.c: Fix the compactheaders option in sip.conf - - * channels/chan_sip.c: 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 18:47 +0000 [r168722] Olle Johansson <oej@edvina.net> - - * /, configs/extconfig.conf.sample: 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 18:39 +0000 [r168719] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/strings.h: Resolve issue with negative vs - non-negative length parameters. (closes issue #14245) Reported - by: dveiga - -2009-01-15 18:08 +0000 [r168711-168712] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: 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. - - * configs/sip.conf.sample: Clarify some misunderstandings and make - it even more clear that you can refer to a peer in the register= - line. - -2009-01-15 15:33 +0000 [r168705] Sean Bright <sean.bright@gmail.com> - - * apps/app_meetme.c: 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 13:37 +0000 [r168636-168639] Olle Johansson <oej@edvina.net> - - * CREDITS, CHANGES: Related to issue #14246 Update changes for - SIPRemoveHeader() - - * channels/chan_sip.c: Add capability to remove added SIP headers - *before* INVITE is generated. (closes issue #14246) Reported by: - klaus3000 Patches: 2patch_chan_sip_SIPRemoveHeader_trunk.txt - uploaded by klaus3000 (license 65) - - * apps/app_queue.c: Add support for setting the Reason header when - cancelling a call in the queue because someone else answered. - Previously, only dial() was supported. EDV-102 - -2009-01-15 00:14 +0000 [r168629] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: 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 23:10 +0000 [r168626] Sean Bright <sean.bright@gmail.com> - - * main/cli.c: Don't crash when typing 'core set verbose' or 'core - set debug' by themselves. (closes issue #14219) Reported by: - jamesgolovich Patches: asterisk-setverbosecrash.diff.txt uploaded - by jamesgolovich (license 176) - -2009-01-14 21:51 +0000 [r168623] Richard Mudgett <rmudgett@digium.com> - - * /, channels/misdn/isdn_lib.c: 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 21:19 +0000 [r168619] Doug Bailey <dbailey@digium.com> - - * channels/chan_dahdi.c: This fixes a problem where MWI FSK spills - were being injected onto off hook fxs lines. (closes issue - #14143) Reported by: alecdavis Patches: - chan_dahdi-14143.patch.txt uploaded by dbailey (license ) Tested - by: alecdavis - -2009-01-14 20:58 +0000 [r168615] Sean Bright <sean.bright@gmail.com> - - * /, contrib/scripts/autosupport: 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 20:51 +0000 [r168613] Steve Murphy <murf@digium.com> - - * /, apps/app_page.c: 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 20:13 +0000 [r168610] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: 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 19:36 +0000 [r168609] Michiel van Baak <michiel@vanbaak.info> - - * main/asterisk.c: Fix compilation on FreeBSD and OSX This started - as work to fix the 'core show sysinfo' CLI command but while - working on it oej pointed out that read_credentials did not - compile neither. So while being there, fix that as well. Thanks - for all the testing oej! (closes issue #14129) Reported by: ys - Tested by: oej, mvanbaak - -2009-01-14 19:11 +0000 [r168601-168604] Tilghman Lesher <tlesher@digium.com> - - * main/udptl.c, /: 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 ........ - - * channels/chan_misdn.c: Mostly spacing changes; no functionality - change at all. - -2009-01-14 02:00 +0000 [r168594] Terry Wilson <twilson@digium.com> - - * /, apps/app_page.c: 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 ........ - -2009-01-13 23:57 +0000 [r168591] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_misdn.c: Janitor patch for chan_misdn (make channel - variable access safe) (closes issue #12887) Reported by: pputman - Patches: chan_misdn_threadsafe.patch uploaded by pputman (license - 81) - -2009-01-13 23:05 +0000 [r168585-168588] Terry Wilson <twilson@digium.com> - - * res/res_http_post.c: Fully overwrite a same-named file when - uploading (closes issue #14190) Reported by: timking - - * Makefile, include/asterisk/options.h, main/asterisk.c: Add option - to hide console connect messages (closes issue #14222) Reported - by: jamesgolovich Patches: asterisk-hideconnect.diff.txt uploaded - by jamesgolovich (license 176) Tested by: otherwiseguy - -2009-01-13 22:30 +0000 [r168579] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Clarify a message that app_queue prints and - change to a debug-level message The "No one is answering..." - verbose message contained 3 numbers that were not explained in - any way to whoever was viewing the message. It is more helpful - now since the message explains what the numbers mean. Also, the - message has been downgraded to "DEBUG" level. (closes issue - #14172) Reported by: caio1982 Patches: queue_answering_debug.diff - uploaded by caio1982 (license 22) - -2009-01-13 22:22 +0000 [r168578] Terry Wilson <twilson@digium.com> - - * /, channels/chan_sip.c: 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 21:18 +0000 [r168575] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Allow - specifying a port number in the user portion of a register => - line in sip.conf With this commit, a register => line in sip.conf - may contain a port number in the "user" section of the line. - Please see CHANGES and sip.conf.sample for more details regarding - this. (closes issue #14198) Reported by: Nick_Lewis Patches: - chan_sip.c-domainport2.patch uploaded by Nick (license 657) - Tested by: Nick_Lewis - -2009-01-13 19:22 +0000 [r168562] Russell Bryant <russell@digium.com> - - * 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, 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: 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 17:51 +0000 [r168547] Tilghman Lesher <tlesher@digium.com> - - * /, funcs/func_logic.c: 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) - ........ - -2009-01-13 16:02 +0000 [r168539] Dwayne M. Hubbard <dwayne.hubbard@gmail.com> - - * main/taskprocessor.c: correct a CLI description - -2009-01-12 23:45 +0000 [r168526] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_alsa.c: 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 23:12 +0000 [r168523] Mark Michelson <mmichelson@digium.com> - - * main/srv.c: 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 23:06 +0000 [r168522] Tilghman Lesher <tlesher@digium.com> - - * configure, include/asterisk/autoconfig.h.in, configure.ac, - main/app.c: Some platforms (notably, the BSDs) have a more - efficient implementation called closefrom(3). - -2009-01-12 21:51 +0000 [r168508-168517] Jeff Peeler <jpeeler@digium.com> - - * /, res/res_agi.c: 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 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 16:31 +0000 [r168497] Olle Johansson <oej@edvina.net> - - * apps/app_minivm.c: Better to use the proper app name - -2009-01-12 15:00 +0000 [r168485] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: Merged revisions 168482 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r168482 | mmichelson | 2009-01-12 08:58:25 -0600 (Mon, 12 Jan - 2009) | 5 lines I am reverting the fix made in revision 168128 - (and its upward merges) after being contacted by Olle Johansson - and being shown how this fix is incorrect. Thanks to Olle for - clearing this up for me. ........ - -2009-01-12 14:57 +0000 [r168481] Russell Bryant <russell@digium.com> - - * /, configs/indications.conf.sample: 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 ........ - -2009-01-12 14:35 +0000 [r168479] Olle Johansson <oej@edvina.net> - - * main/asterisk.c: Don't include swap.h unless we have swapctl - -2009-01-10 01:42 +0000 [r168334] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_sip.c: sizeof for a stringfield is 4. Kinda low for - reconstructing a field value. - -2009-01-09 23:16 +0000 [r168270] Kevin P. Fleming <kpfleming@digium.com> - - * /, sounds/Makefile: 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 23:15 +0000 [r168269] Richard Mudgett <rmudgett@digium.com> - - * channels/chan_misdn.c: Spacing change - -2009-01-09 23:04 +0000 [r168265] Michiel van Baak <michiel@vanbaak.info> - - * contrib/scripts/sip_nat_settings (added), CHANGES: Add a script - to find out the correct settings for Asterisk behind NAT (closes - issue #13065) Reported by: tzafrir Patches: sip_nat_settings - uploaded by tzafrir (license 46) sip_nat_settings_6 uploaded by - mvanbaak (license 7) Tested by: tzafrir, pabelanger, Dovid and - moi - -2009-01-09 22:21 +0000 [r168200] Russell Bryant <russell@digium.com> - - * /, res/res_musiconhold.c: 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 21:53 +0000 [r168193] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: 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 21:43 +0000 [r168192] Richard Mudgett <rmudgett@digium.com> - - * channels/chan_misdn.c, /: 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 20:25 +0000 [r168142] Terry Wilson <twilson@digium.com> - - * res/res_phoneprov.c: 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 18:30 +0000 [r168090] Tilghman Lesher <tlesher@digium.com> - - * res/res_agi.c, include/asterisk/strings.h: 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 18:01 +0000 [r168014-168054] Matthew Nicholson <mnicholson@digium.com> - - * main/logger.c: Added a comment to logger.c about where to put - includes - - * main/logger.c: Use ast_safe_system() in logger.c instead of - system() (closes issue #14194) Reported by: pabelanger - -2009-01-09 01:15 +0000 [r167935-167973] Terry Wilson <twilson@digium.com> - - * apps/app_originate.c: Set ORIGINATE_STATUS instead of - OUTGOING_STATUS to match the documentation - - * apps/app_dial.c: Set peer context and exten values so MACRO_EXTEN - and MACRO_CONTEXT will be set - -2009-01-08 22:37 +0000 [r167894] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_agi.c: 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) - ........ - -2009-01-08 22:34 +0000 [r167888] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: Revert chan_sip changes which were - accidentally committed in revision 167792 - -2009-01-08 21:40 +0000 [r167835-167837] Tilghman Lesher <tlesher@digium.com> - - * apps/app_minivm.c: Fix variables to comply with documentation - changes - - * apps/app_minivm.c: 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 19:48 +0000 [r167792] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c, CHANGES, apps/app_queue.c: Add the average - talk time for a queue This patch adds the functionality to - app_queue of calculating the average amount of time that channels - are bridged for a queue. The algorithm used to calculate the - average is the same exponential average currently used to - calculate the average holdtime. See the CHANGES file to see the - methods you may use to view this information. (closes issue - #13960) Reported by: coolmig Patches: - app_queue.c.diff.trunk-r158840 uploaded by coolmig (license 621) - -2009-01-08 19:44 +0000 [r167791] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_dahdi.c, CHANGES: Convert dialplan application - DAHDISendCallreroutingFacility to use commas. (closes issue - #13836) Reported by: eliel Patches: chan_dahdi.c.patch uploaded - by eliel (license 64) - -2009-01-08 17:26 +0000 [r167700-167720] Kevin P. Fleming <kpfleming@digium.com> - - * /, channels/chan_sip.c: 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 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 14:27 +0000 [r167662] Leif Madsen <lmadsen@digium.com> - - * contrib/scripts/sip-friends.sql: Oops... fix the fieldname I - changed yesterday to be right. - -2009-01-07 22:36 +0000 [r167542-167569] Russell Bryant <russell@digium.com> - - * /, main/file.c: 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 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 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 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 18:20 +0000 [r167478] BJ Weschke <bweschke@btwtech.com> - - * apps/app_followme.c: 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 18:18 +0000 [r167477] Leif Madsen <lmadsen@digium.com> - - * configs/queues.conf.sample: 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 17:35 +0000 [r167442] Russell Bryant <russell@digium.com> - - * /, main/indications.c: 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 17:05 +0000 [r167416] Doug Bailey <dbailey@digium.com> - - * channels/chan_dahdi.c: Cleanup fsk spill if off hook is detected - during mwi spill. Correct logic error in handling events when - sending mwi spill (closes issue #14143) Reported by: alecdavis - Patches: chan_dahdi.handle_init_event2.diff.txt uploaded by - dbailey - -2009-01-07 14:26 +0000 [r167373] Leif Madsen <lmadsen@digium.com> - - * contrib/scripts/sip-friends.sql: 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 21:36 +0000 [r167301] Mark Michelson <mmichelson@digium.com> - - * /, main/db.c: 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 21:02 +0000 [r167265] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_iax2.c: 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 16:59 +0000 [r167180] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: 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 16:44 +0000 [r167176] Tilghman Lesher <tlesher@digium.com> - - * UPGRADE-1.6.txt: 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 - -2009-01-03 20:29 +0000 [r167125] Jeff Peeler <jpeeler@digium.com> - - * main/asterisk.c: When parsing environment variable - ASTERISK_PROMPT, make sure to proceed to the next character when - a non format specifier is used (no %). Otherwise, the while loop - looking for the null byte will never exit. - -2008-12-31 23:07 +0000 [r167061] Sean Bright <sean.bright@gmail.com> - - * doc/CODING-GUIDELINES, include/asterisk.h, channels/h323/README: - Mostly just whitespace, but also convert 'CVS' to 'SVN' in a - couple places and fix a few typos I found in the - CODING_GUIDELINES. - -2008-12-31 22:53 +0000 [r167057] Terry Wilson <twilson@digium.com> - - * main/xmldoc.c: Don't forget to free typename - -2008-12-31 21:52 +0000 [r167021] Mark Michelson <mmichelson@digium.com> - - * channels/chan_dahdi.c: Change some incorrect syntax for pri set - debug and correct an off-by-one error in ss7 set debug command - -2008-12-31 19:39 +0000 [r166954-166958] Tilghman Lesher <tlesher@digium.com> - - * main/ast_expr2.h, main/ast_expr2.c: That was weird... - - * channels/chan_local.c, /, main/ast_expr2.h, main/ast_expr2.c: - 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 20:50 +0000 [r166908] Terry Wilson <twilson@digium.com> - - * res/res_phoneprov.c, doc/sip-retransmit.txt, - doc/tex/phoneprov.tex, res/res_http_post.c, - phoneprov/polycom_line.xml, doc/realtimetext.txt: Fix some - svn:keywords - -2008-12-29 18:04 +0000 [r166861] Mark Michelson <mmichelson@digium.com> - - * apps/app_dial.c, apps/app_queue.c: 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. - -2008-12-28 15:36 +0000 [r166823] Eliel C. Sardanons <eliels@gmail.com> - - * funcs/func_audiohookinherit.c: Fix a typo in the XML - documentation of the AUDIOHOOK_INHERIT dialplan function. - -2008-12-28 15:15 +0000 [r166773] Russell Bryant <russell@digium.com> - - * /, channels/misdn_config.c: 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 15:10 +0000 [r166731] Terry Wilson <twilson@digium.com> - - * channels/chan_sip.c: There is no section 22.2.2 in rfc 3261. I - believe 26.2.2 is what was meant: Note that in the SIPS URI - scheme, transport is independent of TLS, and thus - "sips:alice@atlanta.com;transport=tcp" and - "sips:alice@atlanta.com;transport=sctp" are both valid (although - note that UDP is not a valid transport for SIPS). The use of - "transport=tls" has consequently been deprecated, partly because - it was specific to a single hop of the request. This is a change - since RFC 2543. - -2008-12-23 20:47 +0000 [r166696] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_sip.c: 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 18:13 +0000 [r166665] Steve Murphy <murf@digium.com> - - * apps/app_dial.c, main/pbx.c, /, main/features.c, - apps/app_macro.c, include/asterisk/pbx.h, apps/app_queue.c, - include/asterisk/features.h: 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<ZOMBIE>, 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<ZOMBIE> 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 16:04 +0000 [r166625] Russell Bryant <russell@digium.com> - - * CHANGES: Fix spelling error. - -2008-12-23 15:17 +0000 [r166569] Mark Michelson <mmichelson@digium.com> - - * main/channel.c, /: 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 04:32 +0000 [r166533] Tilghman Lesher <tlesher@digium.com> - - * main/channel.c, /: 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 23:25 +0000 [r166470] Mark Michelson <mmichelson@digium.com> - - * res/res_agi.c: 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) - -2008-12-22 21:45 +0000 [r166436] Russell Bryant <russell@digium.com> - - * res/res_musiconhold.c: Cosmetic change - don't mix struct - initializer styles. - -2008-12-22 21:08 +0000 [r166382] Mark Michelson <mmichelson@digium.com> - - * channels/chan_dahdi.c, /: 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 20:26 +0000 [r166273-166377] Russell Bryant <russell@digium.com> - - * res/res_musiconhold.c: Fix a bad typo. - - * main/astobj2.c: Remove some error messages. This is the default - handler that is valid to use. - - * /, main/utils.c: 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: 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: 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 16:08 +0000 [r166268] Joshua Colp <jcolp@digium.com> - - * main/dnsmgr.c: 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 16:07 +0000 [r166267] Mark Michelson <mmichelson@digium.com> - - * funcs/func_timeout.c, main/file.c: 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 14:16 +0000 [r166258] Russell Bryant <russell@digium.com> - - * res/res_agi.c: 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. - -2008-12-20 01:37 +0000 [r166219] Russell Bryant <russell@digium.com> - - * include/asterisk/doxyref.h: Make a note about formatting the - review URL in commit messages - -2008-12-19 23:45 +0000 [r166092-166162] Mark Michelson <mmichelson@digium.com> - - * main/audiohook.c: 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: Remove the verbatim tag from the - author line I could have sworn I already did that before, - though... - - * main/channel.c, funcs/func_audiohookinherit.c (added), - include/asterisk/audiohook.h, main/audiohook.c, CHANGES: 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/ - -2008-12-19 21:44 +0000 [r166058] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_dahdi.c, configure, - include/asterisk/autoconfig.h.in, configure.ac: Add configuration - support for half_full DAHDI buffer policy - -2008-12-19 18:20 +0000 [r165954] Eliel C. Sardanons <eliels@gmail.com> - - * apps/app_record.c: Fix the XML documentation for Record(). - <value> tags inside <variable> elements must have CDATA and no - another XML node. - -2008-12-19 15:05 +0000 [r165801-165890] Russell Bryant <russell@digium.com> - - * /, apps/app_chanspy.c: 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) ........ - - * include/asterisk/doxyref.h: Disable some automatic links - generated by doxygen. - - * include/asterisk/doxyref.h: Introduce commit message formatting - guidelines. This documents the recommended outline to use for - commit message. It also covers information on special tags that - can be used in commit messages, as well as the template - functionality that is available on bugs.digium.com. Review: - http://reviewboard.digium.com/r/96/ - - * /, main/utils.c: 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 21:43 +0000 [r165798] Jeff Peeler <jpeeler@digium.com> - - * main/manager.c: (closes issue #13993) Reported by: mika Add - ActionID response to ping if sent with request. - -2008-12-18 21:41 +0000 [r165797] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_voicemail.c: 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 21:21 +0000 [r165792] Joshua Colp <jcolp@digium.com> - - * channels/chan_dahdi.c, channels/chan_misdn.c, - channels/chan_sip.c, pbx/pbx_ael.c, apps/app_queue.c, - channels/chan_oss.c: Numerous documentation updates. (closes - issue #13970) Reported by: pkempgen Patches: - __20081217_cli_usage_fixes.patch.txt uploaded by blitzrage - (license 10) - -2008-12-18 19:34 +0000 [r165724] Mark Michelson <mmichelson@digium.com> - - * res/res_odbc.c: 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 19:33 +0000 [r165662-165723] Russell Bryant <russell@digium.com> - - * apps/app_dial.c, main/pbx.c, include/asterisk/pbx.h: 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/ - - * /, res/res_musiconhold.c: 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 18:36 +0000 [r165658] Tilghman Lesher <tlesher@digium.com> - - * apps/app_voicemail.c: Fix 2 resource leaks and fix another - pipe-to-comma conversion - -2008-12-18 17:13 +0000 [r165599] Joshua Colp <jcolp@digium.com> - - * /, main/rtp.c: 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 16:36 +0000 [r165541] Tilghman Lesher <tlesher@digium.com> - - * res/res_odbc.c: Fix reference counts of the class and add an - assertion to the end. - -2008-12-18 15:25 +0000 [r165502] Eliel C. Sardanons <eliels@gmail.com> - - * main/strings.c, include/asterisk/strings.h: Remove duplicate code - from the ast_str API. We now use __AST_STR_* to access 'struct - ast_str' members, but this must only be used inside the API - implementation. (closes issue #14098) Reported by: eliel Patches: - ast_str.patch uploaded by eliel (license 64) - -2008-12-18 14:23 +0000 [r165433-165469] Russell Bryant <russell@digium.com> - - * apps/app_originate.c: Add a \todo note for app_originate. Jared - Smith suggested that we add a way to be able to set variables and - functions on the outbound channel. I think that it's a great - idea, so I have added it as a todo so that it gets done at some - point. - - * apps/app_originate.c (added), CHANGES: Add a new application, - Originate. (closes issue #14075) Reported by: rcasas Patches: - app_originate.c uploaded by rcasas (license 641), heavily - modified by me Tested by: russell Review: - http://reviewboard.digium.com/r/95/ - -2008-12-17 23:39 +0000 [r165397] Tilghman Lesher <tlesher@digium.com> - - * apps/app_record.c: Add RECORD_STATUS variable, as requested on - the -users list. Patch by me (license 14) - -2008-12-17 21:46 +0000 [r165326-165330] Mark Michelson <mmichelson@digium.com> - - * res/res_odbc.c: Fix a refcount leak in res_odbc - - * apps/app_meetme.c, res/res_realtime.c: Fix the build - -2008-12-17 21:28 +0000 [r165319-165325] Tilghman Lesher <tlesher@digium.com> - - * apps/app_macro.c: Oops, broke trunk - - * /, apps/app_macro.c: 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 21:17 +0000 [r165318] Mark Michelson <mmichelson@digium.com> - - * apps/app_meetme.c, res/res_realtime.c, apps/app_directory.c, - apps/app_queue.c, apps/app_voicemail.c: 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 20:50 +0000 [r165254] Steve Murphy <murf@digium.com> - - * utils/extconf.c: This patch is here committed to satisfy the - buildbot, who has a problem with the const. - -2008-12-17 19:55 +0000 [r165219] Terry Wilson <twilson@digium.com> - - * res/res_phoneprov.c: 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 19:52 +0000 [r165216] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: 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 18:49 +0000 [r165180] Matthew Nicholson <mnicholson@digium.com> - - * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: 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 - -2008-12-17 17:56 +0000 [r165145] Russell Bryant <russell@digium.com> - - * doc/appdocsxml.dtd: argsep is used as an attribute for an - argument, as well - -2008-12-17 17:53 +0000 [r165142-165143] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: And actually assign the function to a - pointer... - - * apps/app_voicemail.c: 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. - -2008-12-17 16:41 +0000 [r165108] Kevin P. Fleming <kpfleming@digium.com> - - * utils: ignore this copied file - -2008-12-17 05:04 +0000 [r165039-165071] Steve Murphy <murf@digium.com> - - * utils/Makefile, pbx/pbx_ael.c, utils/ael_main.c, utils/extconf.c, - utils/conf2ael.c, utils/check_expr.c: 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. - - * pbx/ael/ael-test/ref.ael-vtest17, - pbx/ael/ael-test/ref.ael-vtest13: fixed the regressions - -2008-12-16 23:06 +0000 [r164978] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: 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 22:57 +0000 [r164976] Tilghman Lesher <tlesher@digium.com> - - * main/pbx.c, doc/api-1.6.2-changes.txt (added), - funcs/func_logic.c, include/asterisk/pbx.h, utils/extconf.c, - CHANGES, configs/extensions.conf.sample: Add timezone to the - possible fields in a timespec. (closes issue #14028) Reported by: - mostyn Patches: timezone-v2.patch uploaded by mostyn (license - 398) (with additional code guideline fixes and a memory leak fix - by me - license 14) - -2008-12-16 22:45 +0000 [r164942] Jeff Peeler <jpeeler@digium.com> - - * apps/app_record.c: (closes issue #13669) Reported by: pj Delete - file recording if recording terminated from a hangup. - -2008-12-16 22:31 +0000 [r164941] Terry Wilson <twilson@digium.com> - - * channels/chan_sip.c: Make a note of the feature request in bug - #11157 as per the reporter and oej, and suspend the bug since no - one seems to be keen on implementing it any time soon. - -2008-12-16 21:39 +0000 [r164821-164882] Russell Bryant <russell@digium.com> - - * /, main/utils.c: 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 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/asterisk.c: Fix build issues on Linux after sysinfo related - changes - -2008-12-16 20:47 +0000 [r164809-164814] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Qualify - trumps poke per lmadsen. - - * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add - configuration options for finer control over how Asterisk handles - having to poke all peers at seemingly the same time. (closes - issue #13217) Reported by: cervajs - -2008-12-16 20:41 +0000 [r164807] Russell Bryant <russell@digium.com> - - * main/manager.c, /: 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 20:08 +0000 [r164802] Michiel van Baak <michiel@vanbaak.info> - - * configure, include/asterisk/autoconfig.h.in, configure.ac, - main/asterisk.c: introduce 'core show sysinfo' for systems that - dont have the Linux-ish sysinfo stuff but do have sysctl. (closes - issue #13433) Reported by: mvanbaak Patches: - 2008121300_sysinfosysctl.diff.txt uploaded by mvanbaak (license - 7) with two free calls replaced with ast_free based on feedback - on reviewboard Review: http://reviewboard.digium.com/r/91/ - -2008-12-16 20:04 +0000 [r164801] Steve Murphy <murf@digium.com> - - * main/pbx.c: (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 [r164798] Tilghman Lesher <tlesher@digium.com> - - * contrib/scripts/safe_asterisk: Set up umask as a possible - configuration option. (closes issue #13753) Reported by: irroot - -2008-12-16 17:14 +0000 [r164737] Russell Bryant <russell@digium.com> - - * /, main/threadstorage.c, include/asterisk/threadstorage.h: 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 ........ - -2008-12-16 16:50 +0000 [r164733] Joshua Colp <jcolp@digium.com> - - * pbx/pbx_config.c: Be more detailed about why the include did not - get included. (closes issue #14071) Reported by: kshumard - Patches: pbx_config.patch.improvederroroutput.txt uploaded by - kshumard (license 92) - -2008-12-16 16:00 +0000 [r164675] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: 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 15:44 +0000 [r164659] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: 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 15:31 +0000 [r164648] Steve Murphy <murf@digium.com> - - * main/pbx.c, /: 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 15:00 +0000 [r164602-164623] Russell Bryant <russell@digium.com> - - * apps/app_minivm.c: 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 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: 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) - -2008-12-16 01:52 +0000 [r164565] Sean Bright <sean.bright@gmail.com> - - * doc/tex/odbcstorage.tex: Use tables instead of ASCII art. Also - change a bit of minor formatting. - -2008-12-15 22:25 +0000 [r164519-164525] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: 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: Fix log message to refer to the generic - timing interface, not DAHDI specifically (inspired by issue - #14082) - - * main/frame.c: 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 21:17 +0000 [r164485] Tilghman Lesher <tlesher@digium.com> - - * configs/extconfig.conf.sample, pbx/pbx_realtime.c, CHANGES: Allow - disabling pattern match searches within the Realtime dialplan - switch. (closes issue #13698) Reported by: fhackenberger Patches: - 20081211__bug13698.diff.txt uploaded by Corydon76 (license 14) - Tested by: fhackenberger - -2008-12-15 20:07 +0000 [r164419-164428] Mark Michelson <mmichelson@digium.com> - - * apps/app_page.c: Add an 'i' option to app_page. This option works - the same as the 'i' options for app_dial and app_queue, in that - they will ignore any attempts by phones to forward the call. - (closes issue #13977) Reported by: putnopvut Patches: - page_ignore_forwards.patch uploaded by putnopvut (license 60) - Tested by: putnopvut, acunningham - - * /, include/asterisk/pbx.h: 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 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 19:48 +0000 [r164417] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_sip.c, include/asterisk/strings.h: Revert ast_str - opacity in chan_sip for now, since something wasn't quite right - in the merge. - -2008-12-15 19:42 +0000 [r164415] Steve Murphy <murf@digium.com> - - * include/asterisk/strings.h: I was getting this warning during a - compile on a 64-bit machine running ubuntu server 8.10, and - gcc-4.3.2: [CXXi] chan_vpb.ii -> chan_vpb.oo cc1plus: warnings - being treated as errors In file included from - /home/murf/asterisk/trunk/include/asterisk/utils.h:671, from - chan_vpb.cc:46: - /home/murf/asterisk/trunk/include/asterisk/strings.h: In function - ‘char* ast_str_truncate(ast_str*, ssize_t)’: - /home/murf/asterisk/trunk/include/asterisk/strings.h:479: error: - comparison between signed and unsigned integer expressions - make[1]: *** [chan_vpb.oo] Error 1 make: *** [channels] Error 2 - which this fix silences - -2008-12-15 18:12 +0000 [r164351] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: 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) ........ - -2008-12-15 18:09 +0000 [r164349] Tilghman Lesher <tlesher@digium.com> - - * cdr/cdr_pgsql.c: When querying for the structure of the CDR - table, remove the schema, if it exists. (Closes issue #14058) - -2008-12-15 17:24 +0000 [r164312] Joshua Colp <jcolp@digium.com> - - * main/file.c: 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 17:21 +0000 [r164272-164309] Russell Bryant <russell@digium.com> - - * channels/h323/ast_h323.cxx, include/asterisk/strings.h: Fix a - couple more build issues related to ast_str_opaque - - * pbx/pbx_dundi.c: 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 16:16 +0000 [r164268-164270] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Fix a compile warning and a logic error that - could have been bad for non-realtime queues - - * apps/app_queue.c: 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 15:41 +0000 [r164208-164257] Joshua Colp <jcolp@digium.com> - - * configure, include/asterisk/autoconfig.h.in, apps/app_fax.c, - configure.ac: 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 - - * main/utils.c: Update to work with new ast_str changes. - -2008-12-15 14:40 +0000 [r164202-164203] Russell Bryant <russell@digium.com> - - * main/channel.c, /, main/features.c: 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/ ........ - - * apps/app_jack.c: Fix build WRT ast_str_opaque - -2008-12-14 18:16 +0000 [r164168] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/strings.h: Don't pass a negative to an unsigned - type and expect things to work correctly. - -2008-12-14 15:26 +0000 [r164054-164137] Sean Bright <sean.bright@gmail.com> - - * doc/tex/cdrdriver.tex: Use a \picture instead of ASCII art. - - * res/snmp/agent.c: Use ast_str_strlen() instead of recalculating - the string length. - -2008-12-13 13:26 +0000 [r164028] Michiel van Baak <michiel@vanbaak.info> - - * res/snmp/agent.c: nuke another use of the ast_str internals. - -2008-12-13 08:36 +0000 [r163991] Tilghman Lesher <tlesher@digium.com> - - * cdr/cdr_sqlite3_custom.c, apps/app_meetme.c, - funcs/func_strings.c, utils/hashtest.c, cdr/cdr_adaptive_odbc.c, - main/utils.c, apps/app_chanisavail.c, include/asterisk/tcptls.h, - cdr/cdr_pgsql.c, res/res_http_post.c, apps/app_followme.c, - res/res_config_sqlite.c, main/config.c, main/cli.c, main/cdr.c, - channels/chan_dahdi.c, res/res_config_odbc.c, main/manager.c, - configure, funcs/func_odbc.c, res/res_agi.c, apps/app_dumpchan.c, - main/logger.c, main/http.c, main/app.c, apps/app_externalivr.c, - res/res_config_ldap.c, include/asterisk/threadstorage.h, - cdr/cdr_manager.c, res/res_clialiases.c, utils/refcounter.c, - res/res_config_pgsql.c, main/strings.c (added), main/pbx.c, - channels/chan_sip.c, main/Makefile, main/translate.c, - include/asterisk/cdr.h, apps/app_queue.c, channels/iax2-parser.c, - funcs/func_realtime.c, utils/Makefile, res/res_config_curl.c, - main/tcptls.c, include/asterisk/app.h, funcs/func_curl.c, - utils/hashtest2.c, include/asterisk/strings.h, - include/asterisk/pbx.h, main/asterisk.c, main/xmldoc.c, - apps/app_voicemail.c, utils/check_expr.c: Merge ast_str_opaque - branch (discontinue usage of ast_str internals) - -2008-12-13 03:03 +0000 [r163951-163952] Sean Bright <sean.bright@gmail.com> - - * doc/tex/asterisk.tex: This shouldn't have gotten commited. We - might want to generate this into a separate file instead of the - version controlled one. - - * doc/tex/qos.tex, doc/tex/asterisk.tex: Use actual tables instead - of ASCII art ones. - -2008-12-13 00:59 +0000 [r163912] Joshua Colp <jcolp@digium.com> - - * apps/app_chanspy.c: Only detach and destroy the whisper - audiohooks if they are actually in use. - -2008-12-12 23:48 +0000 [r163873] Terry Wilson <twilson@digium.com> - - * apps/app_queue.c: 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 23:06 +0000 [r163828] Russell Bryant <russell@digium.com> - - * res/res_clioriginate.c: Add a note to indicate why this only - supports one channel for now. - -2008-12-12 22:04 +0000 [r163762] Tilghman Lesher <tlesher@digium.com> - - * main/editline/read.c, /, main/asterisk.c: 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 20:12 +0000 [r163716] Russell Bryant <russell@digium.com> - - * CHANGES, res/res_clioriginate.c: Add a new CLI command, "channel - redirect", which is similar in operation to AMI Redirect. Review: - http://reviewboard.digium.com/r/89/ - -2008-12-12 19:16 +0000 [r163675] Steve Murphy <murf@digium.com> - - * channels/chan_dahdi.c: demote always-appearing debug message (for - certain boards) to ast_debug lev 3 msg instead - -2008-12-12 18:45 +0000 [r163642-163670] Russell Bryant <russell@digium.com> - - * main/tcptls.c, channels/chan_sip.c: 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: 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: 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 18:17 +0000 [r163629] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: 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 17:22 +0000 [r163582-163612] Michiel van Baak <michiel@vanbaak.info> - - * res/res_monitor.c: Document default Monitor file location. - (closes issue #14065) Reported by: kshumard Patches: - res_monitor.documentation.patch.txt uploaded by kshumard (license - 92) - - * channels/chan_skinny.c: Fix codec capability setup in chan_skinny - Behaviour now is that general codec config flows to default_line - and default_device. [devices] stuff amends default_device and - similar for [lines]. These are copied to individual device and - line as they are created. Added confcapability and confprefs for - the configured stuff which doesn't change as device and so on are - connected. prefs are based on line prefs if they exist, else the - device prefs are used (prefs identifies codec order). (closes - issue #13806) Reported by: pj Patches: codecs.diff uploaded by - wedhorn (license 30) Tested by: pj and me - -2008-12-12 16:55 +0000 [r163579] Joshua Colp <jcolp@digium.com> - - * main/channel.c, channels/chan_sip.c: 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 15:10 +0000 [r163516] Doug Bailey <dbailey@digium.com> - - * configs/phoneprov.conf.sample: Add internationalization to sample - configuration file - -2008-12-12 14:44 +0000 [r163449-163512] Russell Bryant <russell@digium.com> - - * /, pbx/pbx_dundi.c: 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) ........ - - * main/channel.c, main/autoservice.c, /, - include/asterisk/channel.h: 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/ - ........ - -2008-12-11 23:38 +0000 [r163384] Tilghman Lesher <tlesher@digium.com> - - * /, main/asterisk.c: 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 22:49 +0000 [r163317] Matthew Nicholson <mnicholson@digium.com> - - * /, pbx/pbx_dundi.c: 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 21:48 +0000 [r163241-163254] Russell Bryant <russell@digium.com> - - * /, funcs/func_strings.c, funcs/func_cut.c: 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: 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 20:57 +0000 [r163198-163213] Mark Michelson <mmichelson@digium.com> - - * configs/voicemail.conf.sample, apps/app_voicemail.c: 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) - - * main/features.c: Add an appropriate goto if ast_call fails - -2008-12-11 20:07 +0000 [r163171] Russell Bryant <russell@digium.com> - - * main/channel.c: 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 20:02 +0000 [r163168] Tilghman Lesher <tlesher@digium.com> - - * configure, configure.ac: Sometimes even Linux needs -lm to link - libtonezone, such as when libtonezone is compiled statically. - (closes issue #13887) Reported by: tzafrir - -2008-12-11 19:40 +0000 [r163166] Mark Michelson <mmichelson@digium.com> - - * main/features.c: Reduce indentation level of - ast_feature_request_and_dial - -2008-12-11 17:06 +0000 [r163094] Russell Bryant <russell@digium.com> - - * /, main/features.c: 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 16:52 +0000 [r163089] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_agi.c: 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 16:47 +0000 [r163081-163085] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_queue.c: 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 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 15:40 +0000 [r163037] Sean Bright <sean.bright@gmail.com> - - * doc/tex/qos.tex: Fix some of the grammar issues in - doc/tex/qos.tex. (closes issue #14049) Reported by: kshumard - Patches: doc.tex.qos.tex.patch uploaded by kshumard (license 92) - (Slight modifications by seanbright) - -2008-12-11 15:05 +0000 [r162997] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: 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 23:01 +0000 [r162930] Tilghman Lesher <tlesher@digium.com> - - * main/pbx.c: Previously missing line, now the substitution works - correctly - -2008-12-10 22:53 +0000 [r162927] Jeff Peeler <jpeeler@digium.com> - - * /, res/res_musiconhold.c: 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! ........ - -2008-12-10 22:48 +0000 [r162923] Joshua Colp <jcolp@digium.com> - - * res/res_clialiases.c: Fix reloads of aliased CLI commands. Due to - changes done to turn it into a single memory allocation we can't - just use the existing CLI alias structure. We have to destroy all - existing ones and then create new ones. (closes issue #14054) - Reported by: pj - -2008-12-10 22:48 +0000 [r162922] Tilghman Lesher <tlesher@digium.com> - - * main/pbx.c: 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 - -2008-12-10 22:11 +0000 [r162891] Jeff Peeler <jpeeler@digium.com> - - * /, res/res_musiconhold.c: 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 19:02 +0000 [r162739-162805] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: 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 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 17:09 +0000 [r162687] Michiel van Baak <michiel@vanbaak.info> - - * include/asterisk.h, main/asterisk.c, main/cli.c: add tab - completion for 'core set debug X filename.c' (closes issue - #13969) Reported by: jtodd Patches: 20081205__bug13969.diff.txt - uploaded by Corydon76 (license 14) Tested by: mvanbaak, eliel - -2008-12-10 16:39 +0000 [r162664-162667] Mark Michelson <mmichelson@digium.com> - - * doc/tex/misdn.tex, /: 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 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 16:11 +0000 [r162619-162660] Joshua Colp <jcolp@digium.com> - - * res/res_http_post.c: FreeBSD also needs libgen.h (closes issue - #14051) Reported by: ys Patches: res_http_post.c.diff uploaded by - ys (license 281) - - * /, main/rtp.c: 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: 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 11:34 +0000 [r162583] Michiel van Baak <michiel@vanbaak.info> - - * res/snmp/agent.c: 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 - -2008-12-10 01:09 +0000 [r162542] Joshua Colp <jcolp@digium.com> - - * doc/janitor-projects.txt, channels/iax2-parser.c, - apps/app_voicemail.c: Finish conversion to using ARRAY_LEN and - remove it as a janitor project. (closes issue #14032) Reported - by: bkruse Patches: 14032.patch uploaded by bkruse (license 132) - -2008-12-09 23:41 +0000 [r162488] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/stringfields.h: it does help if the compiler - attribute syntax is correct - -2008-12-09 23:10 +0000 [r162466] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_voicemail.c: 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 22:38 +0000 [r162414-162418] Russell Bryant <russell@digium.com> - - * include/asterisk/doxyref.h, contrib/asterisk-ng-doxygen, - main/asterisk.c: Add some additional Asterisk project developer - documentation. After the nightly update of the documentation on - asterisk.org, I'll post an update to asterisk-dev with a pointer - to the changes. This covers some release branch and commit policy - information. None of this should be a surprise, since it's just - documenting what we have already been doing. - - * include/asterisk/utils.h, /, main/utils.c, main/asterisk.c: - 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 21:57 +0000 [r162355] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_voicemail.c: 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 21:16 +0000 [r162342] Joshua Colp <jcolp@digium.com> - - * /, apps/app_directed_pickup.c: 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 20:59 +0000 [r162291] Russell Bryant <russell@digium.com> - - * /, apps/app_meetme.c: 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 20:46 +0000 [r162275] Joshua Colp <jcolp@digium.com> - - * /, apps/app_festival.c: 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 20:40 +0000 [r162271] Steve Murphy <murf@digium.com> - - * /, res/ael/ael_lex.c, res/ael/ael.flex: 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 20:30 +0000 [r162266] Mark Michelson <mmichelson@digium.com> - - * main/pbx.c, /: 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 19:48 +0000 [r162197-162205] Joshua Colp <jcolp@digium.com> - - * /, main/rtp.c: 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 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 18:35 +0000 [r162079-162140] Steve Murphy <murf@digium.com> - - * /, res/ael/ael_lex.c, res/ael/ael.flex: 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/pval.c, /, include/asterisk/pval.h, res/ael/ael_lex.c, - res/ael/ael.flex: 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 16:47 +0000 [r161951-162016] Russell Bryant <russell@digium.com> - - * /, apps/app_disa.c: 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 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) ........ - -2008-12-09 14:49 +0000 [r161947] Eliel C. Sardanons <eliels@gmail.com> - - * funcs/func_odbc.c: Avoid allocating memory for a thread that - don't need it. Also, this memory was not being freed until the - main thread ends. (That is never). (closes issue #14040) Reported - by: eliel Patches: func_odbc.c.patch uploaded by eliel (license - 64) - -2008-12-08 23:04 +0000 [r161911] Brandon Kruse <bkruse@digium.com> - - * main/pbx.c: Note that the recently changed waittime parameter is - in milliseconds. - -2008-12-08 21:41 +0000 [r161830-161869] Joshua Colp <jcolp@digium.com> - - * formats/format_pcm.c: Add alw as a valid file extension for alaw - and ulw as a valid file extension for ulaw. (closes issue #14001) - Reported by: henrikw Patches: alw.diff uploaded by henrikw - (license 627) - - * contrib/scripts/autosupport.8, contrib/scripts/autosupport: - Update autosupport script with a few changes. - -2008-12-08 18:49 +0000 [r161790] Tilghman Lesher <tlesher@digium.com> - - * main/manager.c: Allocate enough space initially for the message. - (closes issue #14027) Reported by: junky Patches: M14027.diff - uploaded by junky (license 177) - -2008-12-08 18:47 +0000 [r161726-161787] Joshua Colp <jcolp@digium.com> - - * main/pbx.c: 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 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 17:23 +0000 [r161721] Matthew Nicholson <mnicholson@digium.com> - - * channels/chan_sip.c: 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 - -2008-12-08 16:02 +0000 [r161679] Terry Wilson <twilson@digium.com> - - * channels/chan_sip.c, CHANGES: Add the ability to play a courtesy - tone to the transfer target in a native SIP attended transfer by - setting the variable ATTENEDED_TRANSFER_COMPLETE_SOUND. - -2008-12-08 04:23 +0000 [r161571-161637] Eliel C. Sardanons <eliels@gmail.com> - - * main/xmldoc.c: - Fix a leak while printing an argument - description. - Avoid printing the name of an argument in the - [Arguments] tag if there is no description for that argument. - - * apps/app_voicemail.c: Add voicemail related applications and - functions XML documentation: applications: - VoiceMail() - - VoiceMailMain() - MailboxExists() - VMAuthenticate() functions: - - MAILBOX_EXISTS() - - * apps/app_sms.c: Introduce SMS() application XML documentation. - -2008-12-06 21:18 +0000 [r161536] Eliel C. Sardanons <eliels@gmail.com> - - * apps/app_speech_utils.c: Move Speech* applications and functions - documentation to XML. - -2008-12-05 23:24 +0000 [r161493] Mark Michelson <mmichelson@digium.com> - - * apps/app_stack.c: 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 21:08 +0000 [r161349-161427] Sean Bright <sean.bright@gmail.com> - - * /, include/asterisk/astobj2.h, main/astobj2.c: 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: Use ast_free() instead of free(), pointed - out by eliel on IRC. - - * apps/app_voicemail.c: 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. - -2008-12-05 14:16 +0000 [r161252-161288] Russell Bryant <russell@digium.com> - - * main/pbx.c, /: 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: Resolve a compiler warning from buildbot about - a NULL format string. - -2008-12-05 10:31 +0000 [r161218] Eliel C. Sardanons <eliels@gmail.com> - - * main/udptl.c, main/frame.c, res/res_musiconhold.c, - channels/chan_iax2.c, res/res_jabber.c, res/res_config_sqlite.c, - main/config.c, main/cli.c, channels/chan_dahdi.c, main/manager.c, - channels/chan_skinny.c, res/res_agi.c, main/features.c, - apps/app_minivm.c, pbx/pbx_ael.c, main/logger.c, main/http.c, - res/res_realtime.c, channels/chan_alsa.c, res/res_config_ldap.c, - apps/app_rpt.c, main/db.c, res/res_config_pgsql.c, main/pbx.c, - channels/chan_sip.c, main/translate.c, channels/chan_agent.c, - res/res_convert.c, res/res_crypto.c, apps/app_queue.c, - channels/chan_oss.c, apps/app_playback.c, - channels/chan_usbradio.c, main/file.c, main/astmm.c, - pbx/pbx_dundi.c, res/res_indications.c, pbx/pbx_config.c, - apps/app_mixmonitor.c, res/res_odbc.c, main/asterisk.c, - apps/app_voicemail.c: Janitor, use ARRAY_LEN() when possible. - (closes issue #13990) Reported by: eliel Patches: array_len.diff - uploaded by eliel (license 64) - -2008-12-05 05:41 +0000 [r161181] Tilghman Lesher <tlesher@digium.com> - - * main/config.c: 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 02:47 +0000 [r161147] Sean Bright <sean.bright@gmail.com> - - * apps/app_voicemail.c: Check the return value of fread/fwrite so - the compiler doesn't complain. Only a problem when IMAP_STORAGE - is enabled. - -2008-12-04 23:00 +0000 [r161115] Dwayne M. Hubbard <dwayne.hubbard@gmail.com> - - * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: If - 'faxdetect=yes' in sip.conf, switch to a 'fax' extension (if it - exists) after T38 is negotiated. Terry Wilson created the - original patch for this functionality, which I slightly modified - and added the faxdetect=yes|no configuration option. This patch - is only for T38 fax detection and does not do anything for G711 - over SIP fax detection. By default, this option is disabled. - Reviewboard: http://reviewboard.digium.com/r/69/ This - functionality is for issue AST-140. - -2008-12-04 19:31 +0000 [r161077] Eliel C. Sardanons <eliels@gmail.com> - - * main/cli.c: Fix minor coding guidelines introduced with CLI - permissions. - -2008-12-04 18:32 +0000 [r161014] Jeff Peeler <jpeeler@digium.com> - - * /, main/rtp.c: 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 16:45 +0000 [r160945] Mark Michelson <mmichelson@digium.com> - - * /, main/callerid.c: 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 <number>" (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 16:37 +0000 [r160938] Michiel van Baak <michiel@vanbaak.info> - - * build_tools/cflags-devmode.xml, channels/chan_skinny.c: Add debug - flag so skinny debug will show information about packets. We dont - want to scare users with this, so we added a devmode compile flag - (closes issue #13952) Reported by: wedhorn Patches: - packetdebug3.diff uploaded by wedhorn (license 30) Tested by: - mvanbaak, wedhorn - -2008-12-04 13:45 +0000 [r160896] Eliel C. Sardanons <eliels@gmail.com> - - * res/res_agi.c: Added XML documentation for the following AGI - commands: - get option - get variable - hangup - noop - -2008-12-04 01:36 +0000 [r160854-160856] Richard Mudgett <rmudgett@digium.com> - - * funcs/func_callerid.c: Jcolp pointed out that num will also match - number - - * funcs/func_callerid.c: * 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 21:58 +0000 [r160791] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_voicemail.c: 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 21:09 +0000 [r160760] Steve Murphy <murf@digium.com> - - * /, funcs/func_callerid.c: 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 20:35 +0000 [r160699-160700] Jason Parker <jparker@digium.com> - - * main/manager.c: Another place this is missing - - * main/manager.c: Fix typo when ListCategories returns none. - (closes issue #13994) Reported by: mika Patches: - ListCategoriesActionPatch.diff uploaded by mika (license 624) - -2008-12-03 19:25 +0000 [r160663] Eliel C. Sardanons <eliels@gmail.com> - - * channels/iax2-provision.c: - 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 18:37 +0000 [r160626] Mark Michelson <mmichelson@digium.com> - - * apps/app_dial.c, apps/app_queue.c, apps/app_stack.c: 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 17:48 +0000 [r160562] Eliel C. Sardanons <eliels@gmail.com> - - * apps/app_minivm.c: - Add <variable /> tags when naming a channel - variable. - Add <filename /> tags when naming a filename. - - Simplify the xml formatting putting some enters. - -2008-12-03 17:38 +0000 [r160559] Tilghman Lesher <tlesher@digium.com> - - * pbx/pbx_spool.c, /: 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 17:07 +0000 [r160555] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: 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 17:01 +0000 [r160481-160552] Tilghman Lesher <tlesher@digium.com> - - * pbx/pbx_spool.c, /: 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 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. ........ - -2008-12-03 11:01 +0000 [r160447] Eliel C. Sardanons <eliels@gmail.com> - - * apps/app_stack.c: - Avoid setting .synopsis and .syntax if we are - using XML documentation (or the xml documentation wont be - loaded). - Use <variable></variable> to refer to a dialplan - variable. - -2008-12-02 18:48 +0000 [r160344-160346] Tilghman Lesher <tlesher@digium.com> - - * CHANGES: Info on LOCAL_PEEK function. - - * apps/app_stack.c: Add LOCAL_PEEK function, as requested by - lmadsen. - -2008-12-02 18:04 +0000 [r160319-160333] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_dahdi.c: remove duplicate comment that I - accidentally merged - - * channels/chan_dahdi.c: (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 17:56 +0000 [r160208-160308] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_sip.c: 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) ........ - - * main/pbx.c, main/frame.c, /, channels/chan_features.c, - include/asterisk/stringfields.h, apps/app_voicemail.c, - main/cli.c: 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 23:37 +0000 [r160170-160172] Sean Bright <sean.bright@gmail.com> - - * main/manager.c, /: 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 ........ - - * channels/chan_phone.c: Silence a build warning. - (chan_phone.c:810: warning: value computed is not used) - - * utils/smsq.c: Pay attention to the return value of system(), even - if we basically ignore it. - -2008-12-01 21:23 +0000 [r160097] Tilghman Lesher <tlesher@digium.com> - - * configure, configure.ac: Use AST_EXT_LIB_SETUP before using - AST_EXT_LIB_CHECK or bad things happen. - -2008-12-01 18:52 +0000 [r160062] Eliel C. Sardanons <eliels@gmail.com> - - * configs/cli_permissions.conf.sample (added), configure, - include/asterisk/autoconfig.h.in, configure.ac, - include/asterisk/cli.h, include/asterisk/_private.h, CHANGES, - main/asterisk.c, main/cli.c: Introduce CLI permissions. Based on - cli_permissions.conf configuration file, we are able to permit or - deny cli commands based on some patterns and the local user and - group running rasterisk. (Sorry if I missed some of the testers). - Reviewboard: http://reviewboard.digium.com/r/11/ (closes issue - #11123) Reported by: eliel Tested by: eliel, IgorG, Laureano, - otherwiseguy, mvanbaak - -2008-12-01 17:34 +0000 [r159911-160004] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: 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. - ........ - - * /, .cleancount: 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 ... - ........ - -2008-12-01 14:09 +0000 [r159898] Michiel van Baak <michiel@vanbaak.info> - - * main/manager.c, /: 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. ........ - -2008-11-29 18:33 +0000 [r159853] Tilghman Lesher <tlesher@digium.com> - - * apps/app_readexten.c: Allow the '#' sign to exist within an - extension (inspired by issue #13330) - -2008-11-29 17:57 +0000 [r159774-159818] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_vpb.cc, /, main/utils.c, channels/chan_iax2.c, - utils/frame.c, include/asterisk/astmm.h, configure, - include/asterisk/compat.h, main/features.c, - include/asterisk/module.h, main/logger.c, - include/asterisk/dlinkedlists.h, main/dns.c, - include/asterisk/utils.h, include/asterisk/devicestate.h, - channels/chan_sip.c, include/asterisk/dundi.h, - include/asterisk/enum.h, configure.ac, channels/chan_agent.c, - include/asterisk/config.h, utils/astman.c, - include/asterisk/cli.h, include/asterisk/channel.h, - include/jitterbuf.h, include/asterisk/manager.h, - utils/conf2ael.c, cdr/cdr_tds.c, main/ast_expr2.c, - include/asterisk/logger.h, Makefile, include/asterisk/res_odbc.h, - main/srv.c, channels/chan_misdn.c, - include/asterisk/linkedlists.h, main/event.c, - include/asterisk/lock.h, include/asterisk/strings.h, - utils/extconf.c, makeopts.in, include/asterisk/stringfields.h, - main/xmldoc.c, utils/check_expr.c: 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 - - * Makefile, funcs/func_sprintf.c (added), main/Makefile, - channels/misdn/ie.c, funcs/func_strings.c, UPGRADE.txt, - res/res_config_sqlite.c, channels/misdn_config.c, funcs/Makefile: - we can now build with -Wformat=2, which found a couple of real - bugs because SPRINTF() use non-literal format strings (which - cannot be checked), move it into its own module so the rest of - func_strings can benefit from format string checking - -2008-11-28 14:20 +0000 [r159734] Michiel van Baak <michiel@vanbaak.info> - - * res/Makefile: Make res_config_ldap compile with the official - OpenLDAP 2.3.X versions. They removed the LDAP_DEPRECATED define - from their source and since we are using a couple of deprecated - function calls we should define it with a CFLAG. Tested by me on - OpenBSD 4.4 and snuff-home on Linux to make sure everything keeps - compiling. It shouldn't break, we only define the LDAP_DEPRECATED - with this which is what all 2.2.X and older versions of OpenLDAP - did in their own tree. - -2008-11-27 20:29 +0000 [r159701] Philippe Sultan <philippe.sultan@gmail.com> - - * res/res_jabber.c: Removed duplicate code - -2008-11-26 22:11 +0000 [r159664-159666] Russell Bryant <russell@digium.com> - - * main/pbx.c: Make a formatting change to test a new post-commit - hook for reviewboard. http://reviewboard.digium.com/r/65/ - - * main/pbx.c: Make a formatting change to test a new post-commit - hook for reviewboard. http://reviewboard.digium.com/r/65/ - - * main/pbx.c: Make a formatting change to test a new post-commit - hook for reviewboard. http://reviewboard.digium.com/r/65/ - -2008-11-26 21:20 +0000 [r159629-159631] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/agi.h, configure, - include/asterisk/autoconfig.h.in, contrib/asterisk-ng-doxygen, - autoconf/ast_gcc_attribute.m4, configure.ac, res/res_agi.c, - apps/app_stack.c, include/asterisk/optional_api.h (added): - improve handling of API calls provided by loaded modules through - use of some GCC features; this makes app_stack's usage of AGI - APIs even cleaner, and will allow it to work 'as expected' either - with or without res_agi being loaded reviewed at - http://reviewboard.digium.com/r/62 - - * main/manager.c, CHANGES: add support for event suppression for - AMI-over-HTTP - -2008-11-26 19:57 +0000 [r159554] Mark Michelson <mmichelson@digium.com> - - * apps/app_dial.c: 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 19:08 +0000 [r159534] Kevin P. Fleming <kpfleming@digium.com> - - * agi/Makefile, utils/Makefile, /, Makefile.moddir_rules, - Makefile.rules: 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 18:33 +0000 [r159475] Tilghman Lesher <tlesher@digium.com> - - * main/udptl.c: 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 14:58 +0000 [r159437] Mark Michelson <mmichelson@digium.com> - - * channels/chan_agent.c: 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 03:18 +0000 [r159402] Jeff Peeler <jpeeler@digium.com> - - * main/features.c: 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 23:03 +0000 [r159360] Steve Murphy <murf@digium.com> - - * main/cdr.c, /, channels/chan_iax2.c: 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 22:45 +0000 [r159276-159317] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, - include/asterisk/dsp.h, CHANGES, main/dsp.c: Add an option, - waitfordialtone, for UK analog lines which do not end a call - until the originating line hangs up. (closes issue #12382) - Reported by: one47 Patches: - zap-waitfordialtone-trunk.080901.patch uploaded by one47 (license - 23) zap-waitfordialtone-bra-1.4.21.2.patch uploaded by fleed - (license 463) Tested by: fleed - - * /, channels/chan_iax2.c: 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 ........ - -2008-11-25 21:49 +0000 [r159250] Mark Michelson <mmichelson@digium.com> - - * apps/app_followme.c: Make the options for the general and - profiles more consistent for the "pls_hold_prompt" option. This - does not affect any released version of Asterisk, so there is no - need to update the CHANGES file for this. (closes issue #13893) - Reported by: eliel - -2008-11-25 21:42 +0000 [r159162-159247] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_iax2.c: 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 ........ ................ - - * pbx/pbx_realtime.c: Don't actually do anything with a negative - priority, because we ignore it in the result, anyway. - - * main/pbx.c: Don't limit the length of the hint at the final step - (from ~8100 chars max (or ~500 chars max on LOW_MEMORY) to 80 - chars max). This will allow more channels to be used in a single - hint. - -2008-11-25 16:18 +0000 [r159093] Terry Wilson <twilson@digium.com> - - * apps/app_festival.c: Add missing variable declaration for PPC - code - -2008-11-25 05:19 +0000 [r159050-159054] Tilghman Lesher <tlesher@digium.com> - - * apps/app_readexten.c: Copyright clarification; also, have - variable set to "t" or "i" on timeout or invalid extension, - respectively. (closes issue #13944) Reported by: chappell - - * channels/chan_usbradio.c, /, configure, - include/asterisk/autoconfig.h.in, configure.ac, - channels/xpmr/xpmr.c, apps/app_rpt.c: 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 03:49 +0000 [r158992] Terry Wilson <twilson@digium.com> - - * channels/chan_usbradio.c: Make chan_usbradio compile under dev - mode - -2008-11-25 01:01 +0000 [r158959] Sean Bright <sean.bright@gmail.com> - - * funcs/func_dialgroup.c, channels/chan_sip.c, - include/asterisk/astobj2.h, res/res_phoneprov.c, - main/taskprocessor.c, channels/chan_console.c, - channels/chan_iax2.c, apps/app_queue.c, main/astobj2.c, - main/config.c, main/manager.c, res/res_timing_pthread.c, - main/features.c, res/res_timing_timerfd.c, utils/hashtest2.c, - res/res_clialiases.c: This is basically a complete rollback of - r155401, as it was determined that it would be best to maintain - API compatibility. Instead, this commit introduces - ao2_callback_data() which is functionally identical to - ao2_callback() except that it allows you to pass arbitrary data - to the callback. Reviewed by Mark Michelson via ReviewBoard: - http://reviewboard.digium.com/r/64 - -2008-11-25 00:19 +0000 [r158876-158925] Matthew Nicholson <mnicholson@digium.com> - - * main/file.c: Fix compiling in dev mode. - - * UPGRADE.txt, apps/app_queue.c: 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/manager.c, res/res_agi.c, include/asterisk/manager.h: 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 21:52 +0000 [r158857] Tilghman Lesher <tlesher@digium.com> - - * main/dsp.c: 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 21:27 +0000 [r158851] Matthew Nicholson <mnicholson@digium.com> - - * main/file.c: Make ast_streamfile() check the result of - ast_openstream() before doing anything with it. (closes issue - #13955) Reported by: chris-mac - -2008-11-24 18:11 +0000 [r158808] Terry Wilson <twilson@digium.com> - - * apps/app_minivm.c: This patch adds a new application for sending - MWI to phones via Asterisk's event subsystem. Also, the minivm - documentation is all converted to use xmldocs. (closes issue - #13946) Reported by: Marquis Patches: - minivmmwi_plus_xmldocs.patch uploaded by Marquis (license 32) - Tested by: otherwiseguy, Marquis - -2008-11-23 03:36 +0000 [r158754-158756] Sean Bright <sean.bright@gmail.com> - - * channels/chan_sip.c, configs/sip.conf.sample: If you enabled - 'notifycid' one of the limitations is that the calling channel is - only found if it dialed the extension that was subscribed to. You - can now specify 'ignore-context' for the 'notifycid' option in - sip.conf which will, as it's value implies, ignore the current - context of the caller when doing the lookup. - - * channels/chan_sip.c: No need to use a separate structure for this - since we can just pass our sip_pvt pointer in directly. - -2008-11-22 17:17 +0000 [r158686-158723] Michiel van Baak <michiel@vanbaak.info> - - * funcs/func_realtime.c: last commit worked on OpenBSD but still - generated warning on Ubuntu. Initialise a variable so - --enable-dev-mode does not complain - - * channels/chan_skinny.c: 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: Add Media Source Update to skinny's - control2str (issue #13948) - - * channels/chan_skinny.c: fix a very occasional core dump in - chan_skinny found by wedhorn. (issue #13948) - - * funcs/func_realtime.c: make this compile under devmode - -2008-11-21 23:40 +0000 [r158606] Steve Murphy <murf@digium.com> - - * /, main/features.c: 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 23:33 +0000 [r158602-158605] Tilghman Lesher <tlesher@digium.com> - - * main/pbx.c: Allow space within an extension, when the space is - within a character class. (requested by lmadsen on -dev, patch by - me) - - * main/pbx.c, /: 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 22:12 +0000 [r158540] Russell Bryant <russell@digium.com> - - * /, include/asterisk/astobj2.h, main/astobj2.c: 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 21:47 +0000 [r158484] Steve Murphy <murf@digium.com> - - * /, main/features.c: 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 21:06 +0000 [r158482] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_dahdi.c: Fix for #13963. Make physical channel - mapping unconfigured default - -2008-11-21 20:42 +0000 [r158449] Kevin P. Fleming <kpfleming@digium.com> - - * UPGRADE-1.2.txt, UPGRADE-1.4.txt, UPGRADE.txt, UPGRADE-1.6.txt, - CHANGES: as suggested by jtodd, document the purposes of the - CHANGES and UPGRADE files - -2008-11-21 19:40 +0000 [r158414] Jason Parker <jparker@digium.com> - - * main/manager.c: 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 17:08 +0000 [r158374] Terry Wilson <twilson@digium.com> - - * cdr/cdr_csv.c: 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 15:53 +0000 [r158315] Doug Bailey <dbailey@digium.com> - - * channels/chan_sip.c: Add fix to prevent crash during reload if - there is an outstanding MWI registration message pending. - -2008-11-21 01:22 +0000 [r158230-158266] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: Use a more expressive constant for a 64-bit - scanned int - - * channels/chan_sip.c: Use some magic constants to get the right - size for this sscanf statement. Thanks Richard! - - * channels/chan_sip.c: 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: 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 - -2008-11-20 19:41 +0000 [r158188] Sean Bright <sean.bright@gmail.com> - - * res/ael/pval.c: 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 18:20 +0000 [r158082-158133] Mark Michelson <mmichelson@digium.com> - - * include/asterisk/file.h, main/frame.c, /, channels/chan_sip.c, - main/file.c, include/asterisk/frame.h: 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 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 17:53 +0000 [r158078] Ryan Brindley <rbrindley@digium.com> - - * main/config.c: more formatting corrections :: one line for loops - and if statements still need {} - -2008-11-20 17:48 +0000 [r158072] Terry Wilson <twilson@digium.com> - - * cdr/cdr_sqlite3_custom.c, cdr/cdr_sqlite.c, cdr/Makefile, - cdr/cdr_adaptive_odbc.c, cdr/cdr_pgsql.c, cdr/cdr_odbc.c, - cdr/cdr_radius.c, cdr/cdr_custom.c, cdr/cdr_manager.c, - cdr/cdr_csv.c: Begin on a crusade to end trailing whitespace! - -2008-11-20 17:46 +0000 [r158070] Ryan Brindley <rbrindley@digium.com> - - * main/config.c: formatting changes :: one line for loops and if - statements should have {} - -2008-11-20 17:39 +0000 [r158066] Mark Michelson <mmichelson@digium.com> - - * apps/app_dial.c, /, channels/chan_sip.c: 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 17:37 +0000 [r158062] Jeff Peeler <jpeeler@digium.com> - - * main/file.c: (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 17:37 +0000 [r158061] Jason Parker <jparker@digium.com> - - * README: Whitespace fix - -2008-11-20 00:08 +0000 [r157974] Kevin P. Fleming <kpfleming@digium.com> - - * main/stdtime, /, main/db1-ast/hash, codecs/gsm/Makefile, - Makefile.moddir_rules, main/db1-ast/db, channels/misdn, - main/db1-ast/mpool, res/ais, res/Makefile, pbx/Makefile, - Makefile.rules, res/snmp, main/stdtime/Makefile, codecs/gsm/src, - main/db1-ast/btree, channels/misdn/Makefile, main/db1-ast/recno, - res/ael, pbx/ael, channels, main/db1-ast/Makefile: 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 ........ - -2008-11-20 00:06 +0000 [r157973] Terry Wilson <twilson@digium.com> - - * res/res_timing_timerfd.c: Fix compiling - -2008-11-19 23:30 +0000 [r157906-157940] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Add a space to the output - - * apps/app_queue.c: Add a RES_NOT_DYNAMIC case for the CLI command - 'queue remove member' - - * CHANGES: Commit CHANGES change I promised when submitting - res_timing_timerfd - -2008-11-19 22:01 +0000 [r157893] Tilghman Lesher <tlesher@digium.com> - - * CHANGES: Add info about REALTIME_FIELD and REALTIME_HASH - -2008-11-19 21:55 +0000 [r157874] Mark Michelson <mmichelson@digium.com> - - * res/res_timing_timerfd.c: Cast this value since a uint64_t is not - the same as an unsigned long long on a 64-bit machine. Reported - by kpfleming on IRC - -2008-11-19 21:54 +0000 [r157870] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_realtime.c: Two new functions, REALTIME_FIELD, and - REALTIME_HASH, which should make querying realtime from the - dialplan a little more consistent and easy to use. The original - REALTIME function is preserved, for those who are already - accustomed to that interface. (closes issue #13651) Reported by: - Corydon76 Patches: 20081119__bug13651__2.diff.txt uploaded by - Corydon76 (license 14) Tested by: blitzrage, Corydon76 - -2008-11-19 19:37 +0000 [r157820] Mark Michelson <mmichelson@digium.com> - - * build_tools/menuselect-deps.in, configure, - include/asterisk/autoconfig.h.in, res/res_timing_pthread.c, - configure.ac, res/res_timing_dahdi.c, res/res_timing_timerfd.c - (added), makeopts.in: Merge the changes from the - res_timing_timerfd branch. This provides a new timing interface. - In order to use it, you must be running a Linux with a kernel - version of 2.6.25 or newer and glibc 2.8 or newer. This timing - interface is a good alternative if a timing source is necessary - (e.g. for IAX trunking) but DAHDI is otherwise unnecessary for - the system. For now, this commit contains the actual work done in - the res_timing_timerfd branch. There are no notices in the README - or CHANGES files yet, but they will be added in my next commit. - The timing API of Asterisk also needs to have a bit of work done - with regards to choosing which timing interface to use. This - commit makes the choice a build-time decision, by only allowing - one of the timer interfaces to be chosen in menuselect. It would - be preferable if the choice could be made at run-time, however. - The preferred timing interface could be loaded and tested, and if - it does not work, choice number two may be used instead. That - sort of thing. That is beyond the scope of work in this branch - though. - -2008-11-19 19:25 +0000 [r157818] Terry Wilson <twilson@digium.com> - - * channels/chan_vpb.cc, cdr/cdr_sqlite3_custom.c, - channels/iax2-provision.c, cdr/cdr_adaptive_odbc.c, - cdr/cdr_pgsql.c, cdr/cdr_radius.c, cdr/cdr_tds.c, - channels/misdn_config.c, cdr/cdr_csv.c, channels/chan_usbradio.c, - channels/chan_skinny.c, main/logger.c, res/ais/evt.c, - pbx/pbx_dundi.c, cdr/cdr_odbc.c, cdr/cdr_custom.c, - cdr/cdr_manager.c, main/xmldoc.c, res/res_clialiases.c: Fix - checking for CONFIG_STATUS_FILEINVALID so that modules don't - crash upon trying to parse an invalid config - -2008-11-19 18:28 +0000 [r157784] Tilghman Lesher <tlesher@digium.com> - - * configure, configure.ac: 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 13:45 +0000 [r157706-157743] Kevin P. Fleming <kpfleming@digium.com> - - * res/res_agi.c: correct small bug introduced during API conversion - - * UPGRADE.txt, UPGRADE-1.6.txt: move relevant entries into - UPGRADE.txt and resync UPGRADE-1.6.txt with previous branches - - * include/asterisk/agi.h, res/res_agi.c, UPGRADE.txt, - UPGRADE-1.6.txt (added), apps/app_stack.c: 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 05:37 +0000 [r157675] Terry Wilson <twilson@digium.com> - - * configs/cdr_adaptive_odbc.conf.sample: Comment out config line - that is in a commented out context - -2008-11-19 01:02 +0000 [r157639] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/logger.h, main/logger.c, main/utils.c, - include/asterisk/strings.h: 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 00:59 +0000 [r157632] Mark Michelson <mmichelson@digium.com> - - * main/astmm.c: 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 00:27 +0000 [r157600] Sean Bright <sean.bright@gmail.com> - - * Makefile, build_tools/make_version, configure, configure.ac, - build_tools/make_buildopts_h, makeopts.in: 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 [r157496-157592] Mark Michelson <mmichelson@digium.com> - - * res/res_musiconhold.c: 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) - - * Makefile: Add some missing $(DESTDIR)s to the bininstall target - of the Makefile. (closes issue #13875) Reported by: pabelanger - Patches: Makefile.155928 uploaded by pabelanger (license 224) - - * apps/app_voicemail.c: 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 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: 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. - -2008-11-18 21:15 +0000 [r157460-157463] Jason Parker <jparker@digium.com> - - * Makefile: Remove echo line that is unnecessary (Thanks - seanbright). - - * contrib/init.d/rc.archlinux.asterisk: Make this executable - - * Makefile, contrib/init.d/rc.archlinux.asterisk (added): Add init - script for ArchLinux (closes issue #13667) Reported by: sherif - Patches: archlinux_rc_makefile.patch uploaded by sherif (license - 591) archlinux_rc_makefile-2.patch uploaded by mvanbaak (license - 7) - -2008-11-18 20:23 +0000 [r157427] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: * 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 19:16 +0000 [r157366] Jeff Peeler <jpeeler@digium.com> - - * /, apps/app_meetme.c: 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 18:31 +0000 [r157306] Mark Michelson <mmichelson@digium.com> - - * apps/app_dial.c, channels/chan_local.c, /, main/features.c, - include/asterisk/channel.h, apps/app_followme.c: 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 18:07 +0000 [r157302] Steve Murphy <murf@digium.com> - - * main/config.c: (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 22:25 +0000 [r157253] Tilghman Lesher <tlesher@digium.com> - - * apps/app_dial.c: 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 19:51 +0000 [r157105-157167] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile.rules: ensure that if a .i file (preprocessed source) is - present, the .o file is made from it, not from the .c file (this - only works because GNU makes respects the order the rules are - defined) - - * Makefile, /, Makefile.moddir_rules, Makefile.rules: 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 157104 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r157104 | 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 15:37 +0000 [r157073] Eliel C. Sardanons <eliels@gmail.com> - - * main/xmldoc.c: Avoid a not needed cast, making code more - readable. - -2008-11-15 04:25 +0000 [r157039-157041] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c, main/features.c, main/taskprocessor.c: Fix a - few more places where the case insensitive hash should be used - since the comparison is case insensitive. - - * channels/chan_console.c: Use the new case insensitive hash - function for console interfaces. The comparison function is case - insensitive. - -2008-11-14 22:36 +0000 [r157006] Tilghman Lesher <tlesher@digium.com> - - * cdr/cdr_adaptive_odbc.c, configs/cdr_adaptive_odbc.conf.sample: - Allow setting static values in CDRs - -2008-11-14 21:19 +0000 [r156962] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: 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 17:35 +0000 [r156916-156918] Terry Wilson <twilson@digium.com> - - * res/res_phoneprov.c: Cleanup whitespace issues - - * res/res_phoneprov.c: Use Mark's new ast_str_case_hash function - instead of jumping through hoops to do insensitive case lookups - -2008-11-14 17:02 +0000 [r156911] Tilghman Lesher <tlesher@digium.com> - - * main/manager.c: Ping is missing the standard double-newline after - the event. (closes issue #13903) Reported by: kebl0155 - -2008-11-14 16:53 +0000 [r156883] Mark Michelson <mmichelson@digium.com> - - * UPGRADE.txt, include/asterisk/strings.h, apps/app_queue.c: Fix - some refcounting in app_queue.c and change the hashing used by - app_queue.c to be case-insensitive. This is accomplished by - adding a new case-insensitive hashing function. This was - necessary to prevent bad refcount errors (and potential crashes) - which would occur due to the fact that queues were initially read - from the config file in a case-sensitive manner. Then, when a - user issued a CLI command or manager action, we allowed for - case-insensitive input and used that input to directly try to - find the queue in the hash table. The result was either that we - could not find a queue that was input or worse, we would end up - hashing to a completely bogus value based on the input. This - commit resolves the problem presented in issue #13703. However, - that issue was reported against 1.6.0. Since this fix introduces - a behavior change, I am electing to not place this same fix in to - the 1.6.0 or 1.6.1 branches, and instead will opt for a change - which does not change behavior. - -2008-11-14 16:34 +0000 [r156874] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_dahdi.c: Remove some useless locking and make sure - we hangup channels on a link when we get a GRS. - -2008-11-14 15:20 +0000 [r156817] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_voicemail.c: 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 00:43 +0000 [r156690-156756] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_while.c: 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 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 19:17 +0000 [r156649] Jeff Peeler <jpeeler@digium.com> - - * main/pbx.c: (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 19:10 +0000 [r156647] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_dahdi.c: Command offsets were not changed correctly - when the command syntax for 'pri set debug' was changed from 'pri - debug'. - -2008-11-13 17:07 +0000 [r156612] Mark Michelson <mmichelson@digium.com> - - * configure, autoconf/ast_c_compile_check.m4: Kevin sent a note - indicating that this change is not necessary, so I am reverting - it - -2008-11-13 15:46 +0000 [r156535-156575] Eliel C. Sardanons <eliels@gmail.com> - - * apps/app_meetme.c, doc/appdocsxml.dtd, main/xmldoc.c: Introduce - XML documentation for: - MeetMe() - MeetMeCount() - - MeetMeChannelAdmin() - MeetMeAdmin() - SLAStation() - SLATrunk() - - Add an attribute to optionlist 'hasparams' with the same - functionality as the one we have in <parameter> and <argument> - (the DTD was updated) - Fix a leak when getting an attribute - while parsing an <optionlist>. - - * main/xmldoc.c: Fix a typo introduced when changing - xmldoc_has_arguments() to xmldoc_has_inside() we need to pass the - name of the node that we are looking for. - - * include/asterisk/xml.h, include/asterisk/xmldoc.h, main/xmldoc.c: - Remove trailing whitespaces using ':%s/\s\+$//' pointed by - seanbright on #asterisk-dev - -2008-11-12 23:13 +0000 [r156443] Sean Bright <sean.bright@gmail.com> - - * /: Use the reviewboard:url SVN property so post-review will work - without modification. - -2008-11-12 21:34 +0000 [r156388] Tilghman Lesher <tlesher@digium.com> - - * apps/app_dial.c, /: 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 20:27 +0000 [r156355] Eliel C. Sardanons <eliels@gmail.com> - - * res/res_clialiases.c: - Make alias->real_cmd point to the - allocated space outside alias->alias. - Register the aliased cli - command (or we will not alias anything). - Use ARRAY_LEN() when - possible. - -2008-11-12 19:47 +0000 [r156299] Steve Murphy <murf@digium.com> - - * main/pbx.c, /: 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 19:38 +0000 [r156298] Russell Bryant <russell@digium.com> - - * res/res_clialiases.c: Fix a bug caused by using sizeof(pointer) - instead of sizeof(the struct) - -2008-11-12 19:28 +0000 [r156295] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_meetme.c: 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 [r156290] Jeff Peeler <jpeeler@digium.com> - - * /, apps/app_meetme.c: 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 18:55 +0000 [r156243] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_iax2.c: 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/ - ........ - -2008-11-12 18:32 +0000 [r156228] Jeff Peeler <jpeeler@digium.com> - - * /, apps/app_meetme.c: 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 17:41 +0000 [r156169] Mark Michelson <mmichelson@digium.com> - - * apps/app_dial.c, /: 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 [r156166] Russell Bryant <russell@digium.com> - - * /, main/asterisk.c: 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 17:28 +0000 [r156162] Eliel C. Sardanons <eliels@gmail.com> - - * main/xmldoc.c: - The paramname is a pointer allocated with - strdup() or malloc(), so, we need to free it with ast_free(). - -2008-11-12 15:33 +0000 [r156127] Mark Michelson <mmichelson@digium.com> - - * configure, autoconf/ast_c_compile_check.m4: 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. - -2008-11-12 13:43 +0000 [r156125] Eliel C. Sardanons <eliels@gmail.com> - - * res/res_agi.c: Add XML documentation for AGI commands: - database - deltree - database get - exec - get data - get full variable - -2008-11-12 06:46 +0000 [r156120] Michiel van Baak <michiel@vanbaak.info> - - * main/udptl.c, main/pbx.c, channels/chan_sip.c, - configs/cli_aliases.conf.sample (added), include/asterisk/cli.h, - CHANGES, res/res_jabber.c, main/rtp.c, main/cli.c, main/cdr.c, - channels/chan_skinny.c, res/res_agi.c, pbx/pbx_ael.c, - pbx/pbx_dundi.c, funcs/func_devstate.c, main/asterisk.c, - channels/chan_mgcp.c, res/res_clialiases.c (added): This commit - does two things: - Add CLI aliases module to asterisk. - Remove - all deprecated CLI commands from the code Initial work done by - file. Junk-Y and lmadsen did a lot of work and testing to get the - list of deprecated commands into the configuration file. - Deprecated CLI commands are now handled by this new module, see - cli_aliases.conf for more info about that. ok russellb@ via - reviewboard (closes issue #13735) Reported by: mvanbaak - -2008-11-12 02:20 +0000 [r156051-156087] Eliel C. Sardanons <eliels@gmail.com> - - * res/res_agi.c, doc/appdocsxml.dtd: - Add 'database del', - 'database put' and 'set music' AGI commands XML documentation. - - Add to the DTD the possibility to put a parameter inside an - <enum>. - - * include/asterisk/agi.h, res/res_agi.c, doc/appdocsxml.dtd, - main/xmldoc.c: Implement AGI XML documentation parsing functions. - A new <agi> element is used to describe the XML documentation. We - have the usual synopsis,syntax,description and seealso for AGI - commands. The CLI 'agi show commands' command was changed to show - all the documentation se ctions. - -2008-11-11 23:32 +0000 [r156017-156018] Pari Nannapaneni <paripurnachand@digium.com> - - * main/manager.c: changing comment style to conform coding - guidelines - - * main/manager.c: Patch by Ryan Brindley -- Make sure that manager - refuses any duplicate 'new category' requests in updateconfig - -2008-11-11 17:57 +0000 [r155967] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/strings.h: use some fancy compiler magic (thanks - to Matthew Woehlke on the gcc-help mailing list) to restore - type-safety to S_OR by going back to a macro, but preserve the - side-effect-safe usage of the macro arguments - -2008-11-11 16:46 +0000 [r155934] Doug Bailey <dbailey@digium.com> - - * res/res_phoneprov.c, phoneprov/polycom_line.xml: Add LINEKEYS - variable to allow for a user to set the number of keys assigned - to a line on a polycom phone - -2008-11-11 16:07 +0000 [r155929] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Remove commentary from the issues list for - SIP TCP/TLS - -2008-11-10 21:14 +0000 [r155863] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_agent.c: 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 21:12 +0000 [r155763-155862] Tilghman Lesher <tlesher@digium.com> - - * res/res_realtime.c: Make documentation of update method match - documentation and update update2 method to match. Reported by: - atis, via -dev mailing list. Fixed by: me - - * /, doc/valgrind.txt: Merged revisions 155803 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - 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: Fix memory leak when MALLOC_DEBUG is - enabled. (closes issue #13864) Reported by: eliel Patches: - readline.c.patch uploaded by eliel (license 64) - -2008-11-10 13:53 +0000 [r155711] Eliel C. Sardanons <eliels@gmail.com> - - * main/pbx.c, main/Makefile, include/asterisk/xmldoc.h (added), - include/asterisk/term.h, include/asterisk/_private.h, - main/asterisk.c, main/xmldoc.c (added): Move all the XML - documentation API from pbx.c to xmldoc.c. Export the XML - documentation API: ast_xmldoc_build_synopsis() - ast_xmldoc_build_syntax() ast_xmldoc_build_description() - ast_xmldoc_build_seealso() ast_xmldoc_build_arguments() - ast_xmldoc_printable() ast_xmldoc_load_documentation() - -2008-11-09 16:30 +0000 [r155554-155671] Sean Bright <sean.bright@gmail.com> - - * configs/chan_dahdi.conf.sample: Fix this as well. Pointed out by - tzafrir. - - * configs/chan_dahdi.conf.sample: Fix some spelling errors, and - convert tabs to spaces. - - * main/channel.c, channels/chan_sip.c, apps/app_directed_pickup.c, - main/features.c, include/asterisk/channel.h: In order to move - away from nested function use, some changes to the recently - introduced ast_channel_search_locked need to be made. - Specifically, the caller needs to be able to pass arbitrary data - which in turn is passed to the callback. This patch addresses all - of the nested functions currently in asterisk trunk. - - * apps/app_dial.c, /, main/features.c, include/asterisk/channel.h, - apps/app_followme.c, apps/app_queue.c: 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 21:46 +0000 [r155513-155516] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c, include/asterisk/strings.h: - Check for - failure when putting the packet in the ast_str - fix a spelling - error in a header file - - * channels/chan_sip.c: Remove some code that is basically a no-op. - Code above this already ensures that the buffer is terminated. - -2008-11-07 23:41 +0000 [r155467] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: 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 22:39 +0000 [r155401] Sean Bright <sean.bright@gmail.com> - - * main/manager.c, channels/chan_sip.c, funcs/func_dialgroup.c, - res/res_timing_pthread.c, include/asterisk/astobj2.h, - main/features.c, res/res_phoneprov.c, utils/hashtest2.c, - channels/chan_console.c, main/taskprocessor.c, apps/app_queue.c, - channels/chan_iax2.c, main/astobj2.c, main/config.c: Add ability - to pass arbitrary data to the ao2_callback_fn (called from - ao2_callback and ao2_find). Currently, passing OBJ_POINTER to - either of these mandates that the passed 'arg' is a hashable - object, making searching for an ao2 object based on outside - criteria difficult. Reviewed by Russell and Mark M. via - ReviewBoard: http://reviewboard.digium.com/r/36/ - -2008-11-07 22:28 +0000 [r155395-155399] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_sip.c: 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: Two bugs relating to colnames found by - Marquis42 on #asterisk-dev - -2008-11-07 21:14 +0000 [r155360] Mark Michelson <mmichelson@digium.com> - - * configs/voicemail.conf.sample: 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 20:13 +0000 [r155324] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_dahdi.c: 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 16:18 +0000 [r155284] Sean Bright <sean.bright@gmail.com> - - * include/asterisk/indications.h, res/res_indications.c, - main/indications.c: Convert open-coded linked list in indications - to the AST_LIST_* macros. This cleans the code up some and should - make it more maintainable as time goes on. Reviewed by Russell, - Kevin, Mark M., and Tilghman via ReviewBoard: - http://reviewboard.digium.com/r/34/ - -2008-11-07 15:52 +0000 [r155282] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: stringfields conversion for struct sip_peer, - as requested :-) - -2008-11-07 15:42 +0000 [r155241-155264] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Remove a bogus ast_free() that Kevin - noticed. This was probably just left over from pre-astobj2ified - chan_sip. - - * include/asterisk/astobj2.h: 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.) - - * main/pbx.c, channels/chan_sip.c: 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. - -2008-11-07 03:09 +0000 [r155206] Kevin P. Fleming <kpfleming@digium.com> - - * pbx/pbx_config.c: correct logic error noticed by rmudgett - (thanks!) - -2008-11-07 03:02 +0000 [r155175-155204] Eliel C. Sardanons <eliels@gmail.com> - - * main/pbx.c: If 'asterisk.conf' is not found, instead of giving - up, load documentation for the 'en_US' language (fix my last - commit). - - * main/pbx.c: Fix an asterisk crash if no asterisk.conf - configuration file is present. - -2008-11-06 22:49 +0000 [r155066-155121] Kevin P. Fleming <kpfleming@digium.com> - - * res/ael/ael_lex.c, utils/extconf.c, res/ael/ael.flex: don't - blindly assume that Darwin and Cygwin need GLOB_ABORTED defined; - only define it if it is not already defined - - * pbx/pbx_config.c: coding style/guidelines cleanup, plus use new - side-effect safe S_OR - - * include/asterisk/strings.h: make S_OR and S_COR safe to use even - if the parameters are function calls or have side effects. it - still bothers me that these are called S_OR and not something - like ast_string_or, but that's water over the bridge - - * channels/chan_dahdi.c: put ifdef protection around the rest of - the libpri function calls that were added at the same time as - progress_with_cause move parsing of the qsig channel mapping - configuration option outside ifdef HAVE_PRI_INBANDDISCONNECT and - into a properly ifdef'd block - -2008-11-06 19:46 +0000 [r155012] Mark Michelson <mmichelson@digium.com> - - * /, configs/voicemail.conf.sample: 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. ........ - -2008-11-06 18:19 +0000 [r154967] Eliel C. Sardanons <eliels@gmail.com> - - * main/pbx.c: Simplify the output of [See Also]. Functions are - printed without parenthesis like: FUNTION Applications are - printed with parenthesis like: AppName() Cli commands are printed - like: 'core show application' The other type of references are - printed as they are inside the <ref> tag. - -2008-11-05 22:22 +0000 [r154923-154926] Sean Bright <sean.bright@gmail.com> - - * apps/app_directed_pickup.c: Fix some whitespace. - - * apps/app_directed_pickup.c, main/features.c: Update a couple - places to use the new ast_channel_search_locked API call. - -2008-11-05 22:19 +0000 [r154922] Tilghman Lesher <tlesher@digium.com> - - * main/asterisk.c: Don't read history on -rx commands. (Closes - issue #13571) Reported by: tzafrir Patch - '0001-no-need-for-history-on-asterisk-rx.patch' uploaded by - tzafrir. - -2008-11-05 22:01 +0000 [r154919] Sean Bright <sean.bright@gmail.com> - - * include/asterisk.h: Fix a problem found while building res_snmp. - -2008-11-05 21:58 +0000 [r154915] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/app.h, funcs/func_strings.c, main/app.c, - CHANGES: Add LISTFILTER dialplan function, along with supporting - documentation. See documentation for more information on how to - use it. - -2008-11-05 20:45 +0000 [r154875] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_dahdi.c, configure, - include/asterisk/autoconfig.h.in, configure.ac: Make compilation - of chan_dahdi so that it does not require the new - pri_progress_with_cause function to have libpri support work. - -2008-11-05 20:33 +0000 [r154839] Michiel van Baak <michiel@vanbaak.info> - - * res/res_http_post.c: make this compile on OpenBSD again. - -2008-11-05 20:17 +0000 [r154796-154837] Eliel C. Sardanons <eliels@gmail.com> - - * channels/chan_agent.c: Add AgentLogin(), AgentMonitorOutgoing() - applications and AGENT() function XML documentation. - - * apps/app_test.c: Add TestClient() and TestServer() applications - XML documentation. - - * apps/app_mixmonitor.c: Add more [see also] references based on - TFOT. - - * apps/app_macro.c: Add Macro(), MacroExit(), MacroExclusive() and - MacroIf() applications XML documentation. (closes issue #13699) - Reported by: snuffy Patches: bug13699_20081016.diff uploaded by - snuffy (license 35) - -2008-11-05 16:11 +0000 [r154687] Steve Murphy <murf@digium.com> - - * main/channel.c, /: 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. ........ - -2008-11-05 14:37 +0000 [r154467-154647] Eliel C. Sardanons <eliels@gmail.com> - - * main/pbx.c, apps/app_privacy.c, apps/app_sayunixtime.c, - main/features.c, apps/app_morsecode.c, apps/app_alarmreceiver.c, - apps/app_amd.c: Add more SeeAlso references based on TFOT. - - * doc/appdocsxml.dtd: We now can have a reference to a filename - inside a <see-also> tag. - - * apps/app_parkandannounce.c: - Add ParkAndAnnounce() application - XML documentation. - - * main/pbx.c, apps/app_page.c, apps/app_authenticate.c, - apps/app_dumpchan.c, apps/app_disa.c, apps/app_image.c, - apps/app_chanspy.c, apps/app_stack.c, apps/app_adsiprog.c: - Add - more <see-also> based on TFOT. - Add the 'filename' type to the - see-also ref. To be able to reference a filename. - - * apps/app_readfile.c, funcs/func_db.c, apps/app_sendtext.c, - funcs/func_blacklist.c, apps/app_url.c, apps/app_queue.c, - apps/app_senddtmf.c, apps/app_db.c: - Add some see-also - references based on TFOT. - - * apps/app_read.c: - Add Read() application XML documentation. - - * apps/app_followme.c: - Add FollowMe() application XML - documentation. - - * apps/app_forkcdr.c, res/res_indications.c: - Add PlayTones() and - StopPlayTones() applications XML documentation. - Fix a dot that - was outside of the <para> in the ForkCDR() XML documentation. - -2008-11-04 23:23 +0000 [r154429] Sean Bright <sean.bright@gmail.com> - - * main/channel.c, channels/chan_sip.c, include/asterisk/channel.h: - Introduce a new API call ast_channel_search_locked, which - iterates through the channel list calling a caller-defined - callback. The callback returns non-zero if a match is found. This - should speed up some of the code that I committed earlier today - in chan_sip (which is also updated by this commit). Reviewed by - russellb and kpfleming via ReviewBoard: - http://reviewboard.digium.com/r/28/ - -2008-11-04 23:03 +0000 [r154366-154428] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_iax2.c: 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) - - * /, channels/chan_iax2.c: 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 20:12 +0000 [r154329] Eliel C. Sardanons <eliels@gmail.com> - - * Makefile: We need to pass the DTD to xmlstarlet to validate - against it the XML. (I thought it was being read within the - DOCTYPE definition inside the XML). - -2008-11-04 19:07 +0000 [r154268] Richard Mudgett <rmudgett@digium.com> - - * channels/chan_misdn.c, /: 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 18:59 +0000 [r154260-154264] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_skinny.c, channels/chan_h323.c: 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). ........ - - * include/asterisk/devicestate.h, main/manager.c, apps/app_page.c, - include/asterisk/config.h, main/features.c, main/devicestate.c, - apps/app_queue.c, main/config.c, apps/app_voicemail.c: Slightly - optimize ast_devstate_str and rename global functions - devstate2str and config_text_file_save to have an ast_ prefix - -2008-11-04 18:06 +0000 [r154225] Eliel C. Sardanons <eliels@gmail.com> - - * apps/app_forkcdr.c: Add XML documentation for the ForkCDR() - application. - -2008-11-04 17:23 +0000 [r154186-154191] Sean Bright <sean.bright@gmail.com> - - * main/pbx.c: GLOB_BRACE is already added to MY_GLOB_FLAGS if it is - supported on the platform. This should resolve some build errors - on Solaris. (issue #13704) Reported by: dougm - - * channels/chan_sip.c, configs/sip.conf.sample: Allow devices that - accept dialog-info+xml (like snoms) to get the Caller ID of the - calling party when subscribed to the state of an extension that - is ringing. This has some limitations which are documented in - sip.conf.sample. (closes issue #13827) Reported by: seanbright - Patches: issue13827.patch uploaded by seanbright (license 71) - Reviewed by: russellb - - * main/Makefile: Fix build errors. - -2008-11-04 15:07 +0000 [r154151] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_vpb.cc, res/res_crypto.c, configure.ac, - cdr/cdr_adaptive_odbc.c, channels/chan_oss.c, - channels/chan_usbradio.c, res/res_config_odbc.c, - apps/app_osplookup.c, funcs/func_odbc.c, configure, - build_tools/menuselect-deps.in, channels/chan_alsa.c, - makeopts.in, cdr/cdr_odbc.c, res/res_odbc.c, - apps/app_voicemail.c: improve configure script to remember the - previous value of each dependency in build_tools/menuselect-deps, - so that (once it has been written) menuselect can use this - information to warn the user when a previously met dependency is - no longer met along the way, change tags used in configure - script, menuselect-deps and code for various dependencies to be - consistently named - -2008-11-04 14:38 +0000 [r154149] Eliel C. Sardanons <eliels@gmail.com> - - * channels/chan_dahdi.c: Add XML documentation for: Applications - - DAHDISendKeypadFacility() - DAHDISendCallreroutingFacility() - -2008-11-03 22:28 +0000 [r154023-154072] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_voicemail.c: 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: Should have passed the string pointer, not the - ast_str structure. (closes issue #13830) Reported by: Marquis - -2008-11-03 18:02 +0000 [r153983] Olle Johansson <oej@edvina.net> - - * configs/sip.conf.sample: Updating docs - -2008-11-03 17:11 +0000 [r153947] Eliel C. Sardanons <eliels@gmail.com> - - * apps/app_stack.c: Add LOCAL() function XML documentation. - -2008-11-03 15:25 +0000 [r153904-153905] Olle Johansson <oej@edvina.net> - - * configs/sip.conf.sample: Spaces to replace tabs... - - * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Adding a - separation of remote authentication and our authentication. - remotesecret => our password for a remote service secret => our - authentication when someone calls us Secret => still has both - functions if remotesecret is not used. - -2008-11-03 13:33 +0000 [r153803-153852] Eliel C. Sardanons <eliels@gmail.com> - - * channels/chan_iax2.c: Add XML documentation for: Functions - - IAXPEER() - IAXVAR() - - * channels/chan_sip.c: Add XML documentation for: Applications - - SIPDtmfMode() - SIPAddHeader() Functions - SIP_HEADER() - - SIPPEER() - SIPCHANINFO() - CHECKSIPDOMAIN() - -2008-11-03 12:26 +0000 [r153787] Kevin P. Fleming <kpfleming@digium.com> - - * configure, autoconf/ast_ext_lib.m4: when --without-<foo> is - passed to the configure script, explicitly inform menuselect that - the package was disabled by the user - -2008-11-03 01:01 +0000 [r153747] Eliel C. Sardanons <eliels@gmail.com> - - * apps/app_waitforring.c, apps/app_waitforsilence.c, apps/app_db.c, - apps/app_ivrdemo.c: Add XML documentation for: - WaitForSilence() - - WaitForNoise() - WaitForRing() - IVRDemo() - DBDel() - - DBDeltree() (issue #13699) Reported by: snuffy Patches: - bug13699_20081016.diff uploaded by snuffy (license 35) (With - minor changes) - -2008-11-02 23:34 +0000 [r153709] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/agi.h, configure, - include/asterisk/autoconfig.h.in, autoconf/ast_gcc_attribute.m4, - configure.ac, include/asterisk/compiler.h, apps/app_stack.c: - 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. - -2008-11-02 20:06 +0000 [r153652] Russell Bryant <russell@digium.com> - - * /, include/asterisk/features.h: 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 - ........ - -2008-11-02 19:39 +0000 [r153616-153650] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_dahdi.c: fix one more warning missed because i did - not have new enough libpri installed - - * res/res_musiconhold.c: fix small bug introduced while cleaning up - compiler warnings - - * /: mark this revision as merged manually - - * utils/muted.c, apps/app_authenticate.c, res/res_phoneprov.c, - main/utils.c, formats/format_wav_gsm.c, res/res_http_post.c, - res/res_musiconhold.c, channels/chan_iax2.c, res/res_jabber.c, - res/res_config_sqlite.c, utils/frame.c, utils/stereorize.c, - main/channel.c, channels/chan_dahdi.c, main/manager.c, - res/ael/ael.tab.c, funcs/func_odbc.c, main/ast_expr2f.c, - res/res_agi.c, main/http.c, main/logger.c, formats/format_gsm.c, - apps/app_adsiprog.c, apps/app_dial.c, channels/chan_sip.c, - apps/app_festival.c, formats/format_wav.c, res/ael/ael.y, - main/db1-ast/hash/hash_page.c, agi/eagi-test.c, res/res_crypto.c, - utils/astman.c, pbx/pbx_lua.c, formats/format_ogg_vorbis.c, - utils/astcanary.c, apps/app_queue.c, channels/chan_oss.c, - agi/eagi-sphinx-test.c, res/ael/ael_lex.c, channels/chan_h323.c, - main/file.c, apps/app_sms.c, pbx/pbx_dundi.c, res/ael/ael.flex, - pbx/pbx_config.c, apps/app_chanspy.c, apps/app_stack.c, - utils/streamplayer.c, main/asterisk.c, apps/app_voicemail.c: - bring over all the fixes for the warnings found by gcc 4.3.x from - the 1.4 branch, and add the ones needed for all the new code here - too - -2008-11-02 06:24 +0000 [r153582] Eliel C. Sardanons <eliels@gmail.com> - - * channels/chan_iax2.c: Add IAX2Provision() application XML - documentation. - -2008-11-02 05:56 +0000 [r153577-153580] Russell Bryant <russell@digium.com> - - * Makefile: validate-docs is a PHONY target - - * Makefile, configure, configure.ac, makeopts.in: Add a handy - makefile target so that you can validate the documentation - against the DTD by running "make validate-docs" - - * Makefile: Modify the Makefile logic for extracting documentation. - - Build the documentation when you run "make", as opposed to - "make install" - Only rebuild the documentation when source code - has been changed - -2008-11-02 05:10 +0000 [r153541-153543] Eliel C. Sardanons <eliels@gmail.com> - - * apps/app_flash.c: Add Flash() application XML documentation. - - * apps/app_talkdetect.c: Fix a typo in the name of the application. - -2008-11-02 04:14 +0000 [r153472-153507] Sean Bright <sean.bright@gmail.com> - - * channels/Makefile: There is a troublesome assert() in the - alsa/control.h header that causes GCC 4.3.2 to complain that the - passed argument will always evaluate to true. So to get things to - compile, disable assert when building chan_usbradio.so. - - * apps/app_record.c: Another little one. - -2008-11-02 02:55 +0000 [r153362-153470] Russell Bryant <russell@digium.com> - - * apps/app_page.c: fix a typo (thanks sean) - - * apps/app_dial.c, funcs/func_speex.c, apps/app_page.c, - apps/app_record.c, funcs/func_env.c, apps/app_dahdiras.c, - funcs/func_math.c, funcs/func_strings.c, apps/app_userevent.c, - apps/app_exec.c, apps/app_chanspy.c, apps/app_playback.c: Fix - various spelling and grammatical issues in documentation - - * apps/app_voicemail.c: - Use a for loop instead of a while loop - - Get rid of an unnecessary variable - - * apps/app_directed_pickup.c: Instead of doing a couple of strlen() - calls each iteration of the loop, only do it once at the - beginning of the function - - * channels/chan_sip.c: Don't ignore the result of find_peer() when - looking for a peer by IP in check_peer_ok(). - - * funcs/func_speex.c, apps/app_dahdibarge.c, funcs/func_rand.c, - apps/app_readfile.c, funcs/func_module.c, funcs/func_dialgroup.c, - include/asterisk/autoconfig.h.in, funcs/func_env.c, - apps/app_dahdiscan.c, apps/app_record.c, funcs/func_strings.c, - apps/app_sayunixtime.c, include/asterisk/extconf.h, - apps/app_alarmreceiver.c, apps/app_image.c, - apps/app_chanisavail.c, apps/app_ices.c, apps/app_exec.c, - main/config.c, main/term.c, include/asterisk/compat.h, configure, - funcs/func_shell.c, apps/app_skel.c, apps/app_dumpchan.c, - include/asterisk/module.h, main/features.c, apps/app_amd.c, - apps/app_url.c, apps/app_milliwatt.c, apps/app_dial.c, - main/pbx.c, include/asterisk/xml.h (added), apps/app_page.c, - funcs/func_timeout.c, main/Makefile, apps/app_privacy.c, - apps/app_echo.c, apps/app_softhangup.c, apps/app_fax.c, - funcs/func_math.c, apps/app_dahdiras.c, configure.ac, - apps/app_disa.c, apps/app_morsecode.c, funcs/func_cut.c, - apps/app_talkdetect.c, apps/app_transfer.c, apps/app_playback.c, - doc/tex/asterisk-conf.tex, Makefile, apps/app_sendtext.c, - funcs/func_channel.c, funcs/func_cdr.c, apps/app_zapateller.c, - build_tools/get_documentation (added), funcs/func_iconv.c, - apps/app_mixmonitor.c, apps/app_chanspy.c, main/asterisk.c, - apps/app_cdr.c, funcs/func_base64.c, funcs/func_md5.c, - apps/app_dictate.c, apps/app_authenticate.c, - apps/app_readexten.c, apps/app_userevent.c, funcs/func_vmcount.c, - main/xml.c (added), funcs/func_sha1.c, funcs/func_logic.c, - funcs/func_uri.c, apps/app_controlplayback.c, funcs/func_enum.c, - apps/app_setcallerid.c, funcs/func_groupcount.c, - funcs/func_config.c, funcs/func_volume.c, funcs/func_odbc.c, - apps/app_mp3.c, apps/app_directory.c, apps/app_jack.c, - apps/app_adsiprog.c, apps/app_while.c, apps/app_nbscat.c, - funcs/func_dialplan.c, funcs/func_db.c, funcs/func_version.c, - apps/app_festival.c, funcs/func_lock.c, apps/app_waituntil.c, - doc, include/asterisk/term.h, include/asterisk/_private.h, - apps/app_system.c, apps/app_getcpeid.c, apps/app_queue.c, - funcs/func_global.c, funcs/func_extstate.c, - funcs/func_realtime.c, apps/app_channelredirect.c, - funcs/func_blacklist.c, apps/app_directed_pickup.c, - include/asterisk/pbx.h, include/asterisk/strings.h, makeopts.in, - apps/app_senddtmf.c, funcs/func_devstate.c, - funcs/func_callerid.c, doc/appdocsxml.dtd (added), - apps/app_verbose.c, apps/app_stack.c: Merge changes from - team/group/appdocsxml This commit introduces the first phase of - an effort to manage documentation of the interfaces in Asterisk - in an XML format. Currently, a new format is available for - applications and dialplan functions. A good number of conversions - to the new format are also included. For more information, see - the following message to asterisk-dev: - http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html - - * channels/chan_sip.c: Ensure that the sip_pvt properly has its - refcount incremented when the scheduler holds a reference to it - for session timer processing. - -2008-11-01 01:55 +0000 [r153296] Sean Bright <sean.bright@gmail.com> - - * configs/sip.conf.sample: The default in chan_sip for - notifyringing is yes, so update the sample conf to reflect that. - -2008-10-31 20:05 +0000 [r153223] Mark Michelson <mmichelson@digium.com> - - * main/dial.c, apps/app_page.c, include/asterisk/dial.h, CHANGES: * - Fixed timeout logic in the dialing API as setting timeouts had no - effect * Updated dialing API documentation to indicate that - timeouts are specified in milliseconds * Added a new timeout - argument to the Page application. If time expires, any endpoints - which have not answered will be hung up. - -2008-10-31 18:55 +0000 [r153181] Terry Wilson <twilson@digium.com> - - * apps/app_dial.c, main/features.c, include/asterisk/channel.h, - apps/app_followme.c, apps/app_queue.c: 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 17:18 +0000 [r153122-153124] Tilghman Lesher <tlesher@digium.com> - - * configs/func_odbc.conf.sample, funcs/func_odbc.c, CHANGES: - Failover for func_odbc, allowing an INSERT query to be performed - when the UPDATE query initially affects 0 rows. (closes issue - #13083) Reported by: Corydon76 Patches: - 20081031__bug13083.diff.txt uploaded by Corydon76 (license 14) - - * /, channels/chan_sip.c: Merged revisions 153114 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r153114 | tilghman | 2008-10-31 11:30:32 -0500 (Fri, 31 Oct 2008) - | 3 lines Turn off qualify on uncached realtime peers. (Closes - issue #13383) ........ - -2008-10-31 09:31 +0000 [r153057] Russell Bryant <russell@digium.com> - - * main/channel.c: Use the ast_str API call to reset the string - instead of manually editing its internals (closes issue #13816) - Reported by: eliel Patches: channel.c.patch uploaded by eliel - (license 64) - -2008-10-30 20:59 +0000 [r152993] Sean Bright <sean.bright@gmail.com> - - * /, bootstrap.sh: Merged revisions 152992 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r152992 | seanbright | 2008-10-30 16:58:24 -0400 (Thu, 30 Oct - 2008) | 2 lines The -I argument to aclocal needs a space before - the include directory name. ........ - -2008-10-30 20:46 +0000 [r152990] Russell Bryant <russell@digium.com> - - * include/asterisk/timing.h: Add a todo for a new timing API - implementation that would work for Linux systems as of kernel - 2.6.25 and glibc 2.8 - -2008-10-30 20:35 +0000 [r152923-152969] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_h323.c: Merged revisions 152958 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r152958 | tilghman | 2008-10-30 15:33:28 -0500 (Thu, 30 - Oct 2008) | 3 lines Cannot join detached threads. See - http://www.opengroup.org/onlinepubs/000095399/functions/pthread_join.html - (Closes issue #13400) ........ - - * channels/chan_local.c, /: Merged revisions 152922 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r152922 | tilghman | 2008-10-30 14:43:38 -0500 (Thu, 30 - Oct 2008) | 6 lines 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 19:40 +0000 [r152887-152920] Russell Bryant <russell@digium.com> - - * 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. - -2008-10-30 19:23 +0000 [r152879] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: I just noticed this construct and thought it - was silly to have a bunch of case statements with duplicated code - in each case. Instead, just use the built-in fallthrough - capability of case statements and reduce the code to a single - instance - -2008-10-30 19:21 +0000 [r152875-152877] Russell Bryant <russell@digium.com> - - * 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. - - * funcs/func_db.c: - spaces to tabs - add some braces - remove - unnecessary cast - -2008-10-30 16:54 +0000 [r152809-152812] Kevin P. Fleming <kpfleming@digium.com> - - * main/cdr.c, /: Merged revisions 152811 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r152811 | kpfleming | 2008-10-30 11:53:48 -0500 (Thu, 30 Oct - 2008) | 3 lines instead of comparing the string pointer to 0, - let's compare the value that was actually parsed out of the - string (found by sparse) ........ - - * include/asterisk/buildinfo.h (added): try to get this committed - before the buildbot complains about a broken tree - - * channels/misdn/isdn_lib_intern.h, channels/misdn/isdn_lib.h, - main/dial.c, main/dnsmgr.c, main/buildinfo.c, - codecs/lpc10/chanwr.c, utils/astcanary.c, - channels/misdn/isdn_lib.c, main/asterisk.c, apps/app_adsiprog.c: - fix a few small things found by using sparse - -2008-10-30 16:38 +0000 [r152807] Mark Michelson <mmichelson@digium.com> - - * main/features.c, CHANGES, configs/features.conf.sample: After - seeing another problem in #asterisk stemming from the low default - value of featuredigittimeout, I decided it was high time to - change it. I have changed the default to 2000 ms based on a - suggestion from Leif Madsen. - -2008-10-30 04:26 +0000 [r152689-152765] Tilghman Lesher <tlesher@digium.com> - - * 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 - - * CHANGES, apps/app_directory.c: Pay attention to the - searchcontexts entry in voicemail.conf (related to AST-125) - - * main/pbx.c: Track down and fix annoying lock errors - -2008-10-29 20:53 +0000 [r152646] Mark Michelson <mmichelson@digium.com> - - * 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 05:47 +0000 [r152605] Steve Murphy <murf@digium.com> - - * apps/app_dial.c, /, apps/app_queue.c, - configs/features.conf.sample: Merged revisions 152538 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r152538 | murf | 2008-10-28 23:19:04 -0600 (Tue, 28 Oct 2008) | - 14 lines 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 05:34 +0000 [r152569] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 152539 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r152539 | russell | 2008-10-29 00:23:51 -0500 (Wed, 29 Oct 2008) - | 7 lines 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 05:01 +0000 [r152536] Steve Murphy <murf@digium.com> - - * apps/app_dial.c, /, main/features.c, include/asterisk/pbx.h, - apps/app_queue.c, include/asterisk/features.h: Merged revisions - 152535 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | - 46 lines 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 22:33 +0000 [r152467] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 152463 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r152463 | tilghman | 2008-10-28 17:32:34 -0500 (Tue, 28 - Oct 2008) | 3 lines Quoting in the wrong direction (Fixes - AST-107) ........ - -2008-10-28 22:26 +0000 [r152448] Doug Bailey <dbailey@digium.com> - - * configs/phoneprov.conf.sample: Add more polycom firmware files to - static mapping - -2008-10-28 21:38 +0000 [r152369-152442] Tilghman Lesher <tlesher@digium.com> - - * 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, /: Merged revisions 152368 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r152368 | tilghman | 2008-10-28 12:04:56 -0500 (Tue, 28 Oct 2008) - | 8 lines 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 23:31 +0000 [r152287] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 152286 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r152286 | jpeeler | 2008-10-27 18:28:49 -0500 (Mon, 27 - Oct 2008) | 2 lines Buffer policy setting for half is not needed. - ........ - -2008-10-27 21:34 +0000 [r152134-152216] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_local.c, /: Merged revisions 152215 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r152215 | tilghman | 2008-10-27 16:32:00 -0500 (Mon, 27 - Oct 2008) | 6 lines 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: Set ARGC in subroutines with the number of - arguments passed. - - * 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 16:03 +0000 [r152132] Jason Parker <jparker@digium.com> - - * 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 20:25 +0000 [r152060] Sean Bright <sean.bright@gmail.com> - - * /, funcs/func_strings.c: Merged revisions 152059 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r152059 | seanbright | 2008-10-26 16:23:36 -0400 (Sun, - 26 Oct 2008) | 7 lines 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 ........ - -2008-10-26 10:23 +0000 [r151980-152020] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Trying to fix the user/peer matching - correctly. This will need some testing before getting merged into - 1.6.1 - - * channels/chan_sip.c: Moving more variables to the sip_cfg - structure, as I have some future ideas for the usage of that - structure. - - * channels/chan_sip.c: Doxygen changes and some formatting. - -2008-10-25 11:02 +0000 [r151906] Russell Bryant <russell@digium.com> - - * /, main/asterisk.c: Merged revisions 151905 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r151905 | russell | 2008-10-25 05:59:02 -0500 (Sat, 25 Oct 2008) - | 8 lines 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 - ........ - -2008-10-23 21:27 +0000 [r151830] Terry Wilson <twilson@digium.com> - - * funcs/func_odbc.c: allow to compile under --enable-dev-mode (gcc - didn't actually complain when I was using ccache...) - -2008-10-23 15:54 +0000 [r151762] Tilghman Lesher <tlesher@digium.com> - - * contrib/scripts/vmdb.sql: Clarify documentation, following merge - of realtime_update2 branch - -2008-10-23 15:38 +0000 [r151739-151761] Olle Johansson <oej@edvina.net> - - * CHANGES: Thanks russellb for reminding an old man.... - - * channels/chan_sip.c, doc/tex/channelvariables.tex: Adding a small - new feature. Setting _SIPFROMDOMAIN in a channel will set the - domain we use for the URI in the outbound call leg. - -2008-10-23 15:28 +0000 [r151732] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_odbc.c: Simplify some nested functions, as suggested - by Russell on -dev - -2008-10-23 15:09 +0000 [r151722] Doug Bailey <dbailey@digium.com> - - * res/res_http_post.c: Add patch to handle how IE7 issues POST - requests using Window path spec including backslash delimiters - -2008-10-22 22:11 +0000 [r151682] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_odbc.c, CHANGES: Added debugging CLI functions - -2008-10-22 20:45 +0000 [r151642] BJ Weschke <bweschke@btwtech.com> - - * channels/chan_sip.c: revert the changes in issue #13705 - it's - being re-opened as while the results fixed the complaint in the - issue, it introduced other more undesirable issues than what was - already reported - -2008-10-22 20:05 +0000 [r151601] Tilghman Lesher <tlesher@digium.com> - - * contrib/scripts/live_ast (added): 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 [r151600] Mark Michelson <mmichelson@digium.com> - - * 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) - -2008-10-22 17:45 +0000 [r151554-151555] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Print out the right var in the log message - - * channels/chan_sip.c: Fix this check to use the proper variable - (the result from get_in_brackets) - -2008-10-22 15:08 +0000 [r151420-151512] Mark Michelson <mmichelson@digium.com> - - * 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 15:20 +0000 [r151371] Tilghman Lesher <tlesher@digium.com> - - * 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 11:02 +0000 [r151327] BJ Weschke <bweschke@btwtech.com> - - * 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 05:07 +0000 [r151246] BJ Weschke <bweschke@btwtech.com> - - * pbx/pbx_config.c, main/config.c: Do NOT attempt to do anything - with the ast_config struct when it's been returned as INVALID by - the config file interpreter. (closes issue #13741) - -2008-10-20 05:00 +0000 [r151242-151243] Kevin P. Fleming <kpfleming@digium.com> - - * autoconf/ast_check_pwlib.m4, /, autoconf/ast_check_openh323.m4, - configure.ac: Merged revisions 151241 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r151241 | kpfleming | 2008-10-20 07:57:33 +0300 (Mon, 20 Oct - 2008) | 2 lines rename this macro to properly reflect what it - does ........ - - * autoconf/ast_prog_egrep.m4, autoconf/ast_c_define_check.m4, - autoconf/ast_ext_tool_check.m4 (added), - autoconf/ast_check_mandatory.m4 (added), /, - autoconf/ast_check_openh323.m4, autoconf/ast_prog_ld_gnu.m4, - autoconf/ast_prog_sed.m4, acinclude.m4 (removed), - autoconf/ast_check_pwlib.m4, autoconf (added), - 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: Merged - revisions 151240 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r151240 | kpfleming | 2008-10-20 07:45:56 +0300 (Mon, 20 Oct - 2008) | 3 lines 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 ........ - -2008-10-19 20:30 +0000 [r151188-151190] BJ Weschke <bweschke@btwtech.com> - - * /: Block 151167 from coming forward into the /trunk this is a 1.4 - fix only. - - * /: Block 151100 from coming forward into the /trunk this is a 1.4 - fix only. - -2008-10-19 19:11 +0000 [r151101] Kevin P. Fleming <kpfleming@digium.com> - - * 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 - -2008-10-19 13:10 +0000 [r151060] Michiel van Baak <michiel@vanbaak.info> - - * channels/chan_skinny.c: dont segfault when placing a call to a - line that has no registered device. - -2008-10-19 07:20 +0000 [r151019] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Adding changes from train and flight back - home from SIPit23 in Lannion, France. - Additional comments on - TCP/TLS implementation - Some additions for new drafts/rfcs (no - new functionality really, mostly documentation) - Other random - small fixes - -2008-10-18 10:27 +0000 [r150930-150971] Michiel van Baak <michiel@vanbaak.info> - - * Makefile: Make sure we support nested functions and generation of - trampolines under OpenBSD. (closes issue #13724) Reported by: - mvanbaak - - * contrib/init.d/rc.mandriva.asterisk, - contrib/init.d/rc.debian.asterisk, - contrib/init.d/rc.redhat.asterisk, - contrib/init.d/rc.suse.asterisk: dont use deprecated commands in - the init scripts. (closes issue #13720) Reported by: - decryptus_proformatique Patches: - contrib_initd_module_reload.patch uploaded by decryptus (license - 555) With mods by me to fix stop commands as well - -2008-10-18 03:35 +0000 [r150773-150887] BJ Weschke <bweschke@btwtech.com> - - * apps/app_authenticate.c, CHANGES: Give app_authenticate the - ability to select a prompt other than the default. (closes issue - #13734) reported and patched by: jvandal - - * 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 - - * doc/manager_1_1.txt, CHANGES, apps/app_queue.c: The QueueEntry - event now has the uniqueid of the channel included. (closes issue - #13731) reported and patched by: caio1982 - -2008-10-17 21:48 +0000 [r150731] Matthew Fredrickson <creslin@digium.com> - - * configure, configure.ac: Update configure check to check for new - function in libpri (pri_progress_with_cause) - -2008-10-17 21:35 +0000 [r150729] Jason Parker <jparker@digium.com> - - * codecs/codec_adpcm.c, codecs/ex_g722.h (added), - codecs/codec_gsm.c, codecs/ex_adpcm.h (added), codecs/ex_alaw.h - (added), codecs/ex_g726.h (added), codecs/ex_gsm.h (added), - codecs/slin_ulaw_ex.h (removed), codecs/slin_lpc10_ex.h - (removed), codecs/codec_resample.c, codecs/slin_g722_ex.h - (removed), codecs/g722_slin_ex.h (removed), codecs/ex_ulaw.h - (added), codecs/adpcm_slin_ex.h (removed), codecs/ex_ilbc.h - (added), codecs/slin_adpcm_ex.h (removed), codecs/g726_slin_ex.h - (removed), codecs/slin_g726_ex.h (removed), codecs/codec_lpc10.c, - codecs/gsm_slin_ex.h (removed), codecs/slin_gsm_ex.h (removed), - codecs/codec_a_mu.c, codecs/codec_g722.c, codecs/ex_lpc10.h - (added), codecs/codec_alaw.c, codecs/codec_speex.c, - codecs/codec_g726.c, include/asterisk/slin.h (added), - codecs/ex_speex.h (added), codecs/slin_resample_ex.h (removed), - codecs/ulaw_slin_ex.h (removed), codecs/slin_ilbc_ex.h (removed), - codecs/ilbc_slin_ex.h (removed), codecs/lpc10_slin_ex.h - (removed), codecs/codec_ulaw.c, codecs/codec_ilbc.c, - codecs/speex_slin_ex.h (removed), codecs/slin_speex_ex.h - (removed): Merge codec_consistency branch. This should make - sample usage much happier. - -2008-10-17 17:31 +0000 [r150664] Michiel van Baak <michiel@vanbaak.info> - - * main/cli.c: Fix CLI command 'channel request hangup' Prodded on - IRC by Russell and fixed by eliel (closes issue #13730) Reported - by: eliel Patches: main_cli.patch uploaded by eliel (license 64) - -2008-10-17 17:25 +0000 [r150640] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample: Merge in - patch for #13454. Includes CallRereouting dialplan application, - option for discard of remote hold messages, and using the - alternate logical channel mapping in Q.SIG instead of the default - physical channel mapping. - -2008-10-17 17:09 +0000 [r150580-150635] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_iax2.c: Make helper call a little safer (suggested - by Russell on IRC) - - * include/asterisk/sched.h, channels/chan_iax2.c: Fix the FRACK! - warnings in chan_iax2 when POKE/LAGRQ packets are not answered. - -2008-10-17 08:42 +0000 [r150469-150510] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Adding some additional thoughts on - configuration changes to TCP/TLS - - * Makefile: Make sure we support nested functions with GCC 4.01 - OS/X. This might not be OS/X only, but I'll leave it to kpfleming - to add this to the configure script for testing. - -2008-10-17 06:00 +0000 [r150426] Michiel van Baak <michiel@vanbaak.info> - - * channels/chan_skinny.c, UPGRADE.txt, configs/skinny.conf.sample, - CHANGES: Break up skinny.conf into seperate sections for devices - and lines. (closes issue #13412) Reported by: wedhorn Patches: - config-restruct-v4.diff uploaded by wedhorn (license 30) - -2008-10-17 04:28 +0000 [r150384] Tilghman Lesher <tlesher@digium.com> - - * apps/app_meetme.c: Fix option handling code. (closes issue - #11040) Reported by: DEA Patches: rt-meetme-flag-fixes-v2.txt - uploaded by DEA (license 3) with additional fixes by me - -2008-10-17 00:18 +0000 [r150311] Mark Michelson <mmichelson@digium.com> - - * doc/manager_1_1.txt, CHANGES, channels/chan_iax2.c: Add an - IAXregistry manager command. See doc/manager_1_1.txt for more - details of this command. (closes issue #13326) Reported by: ib2 - Patches: bug13326_trunk_20080822.diff uploaded by snuffy (license - 35) - -2008-10-17 00:14 +0000 [r150309] Jeff Peeler <jpeeler@digium.com> - - * apps/app_meetme.c: Initialize character arrays as they are not - guaranteed to be set. - -2008-10-17 00:13 +0000 [r150207-150307] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: After a long discussion on #asterisk-bugs, - it seems kind of odd that a channel would be named after the - originating port. For endpoints that always include ":5060" as - part of the From: header, it will mean that you have a ton of - channels with names like "SIP/5060-3ea38a8b." I am boldly moving - forward with this change in trunk, but I'm not touching other - branches with this one since this definitely would qualify as a - behavior change. If there is a problem with this commit, and I - haven't seen the obvious reason why you'd want to name the - channel after the port from which the call originated, then - please feel free to revert this - - * main/manager.c, /: Merged revisions 150304 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r150304 | mmichelson | 2008-10-16 18:40:54 -0500 (Thu, 16 Oct - 2008) | 6 lines 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: Merged revisions 150298,150301 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r150298 | mmichelson | 2008-10-16 18:34:37 -0500 (Thu, 16 Oct - 2008) | 10 lines 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 ........ - r150301 | mmichelson | 2008-10-16 18:35:07 -0500 (Thu, 16 Oct - 2008) | 3 lines 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) - - * apps/app_skel.c: Updating app_skel.c to follow coding guidelines - with regards to braces used on if statements. (closes issue - #13696) Reported by: alecdavis Patches: - app_skel.bug13696B.115850.diff.txt uploaded by alecdavis (license - 585) - - * channels/chan_iax2.c: Remove an odd redundant comparison - - * 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 16:04 +0000 [r150125] Richard Mudgett <rmudgett@digium.com> - - * channels/chan_misdn.c, /: Merged revisions 150124 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r150124 | rmudgett | 2008-10-16 10:56:06 -0500 (Thu, 16 - Oct 2008) | 1 line Fix memory leak found by customer ........ - -2008-10-16 15:48 +0000 [r150118-150121] Terry Wilson <twilson@digium.com> - - * configs/modules.conf.sample: This is nolonger needed - - * res/res_phoneprov.c: func_strings isn't a dependency of this - module anymore - -2008-10-16 15:02 +0000 [r150052] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: ensure that type=peer entries are only - matched on IP/port, not on name (after oej audits all the calls - to find_peer() to make sure that forcenamematch is set correctly - in each case) - -2008-10-16 15:00 +0000 [r150008-150051] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Doxygen addition - - * channels/chan_sip.c: Add some notes on problems with the TCP/TLS - implementation - -2008-10-16 13:28 +0000 [r149917-149981] Kevin P. Fleming <kpfleming@digium.com> - - * 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 - - * channels/chan_sip.c: some additional debugging tools added at - SIPit23: - move all setting of 'needdestroy' on dialog structures - into the history - report all tags involved when a pedantic check - fails on a REFER - - * 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 - - * configs/musiconhold.conf.sample, res/res_musiconhold.c, CHANGES: - support relative paths in musiconhold.conf, which makes moh work - by default when Asterisk was configured using --prefix and 'make - samples' is run - -2008-10-15 21:36 +0000 [r149848] BJ Weschke <bweschke@btwtech.com> - - * /: Blocking 149840 from coming forward. - -2008-10-15 20:55 +0000 [r149802] Mark Michelson <mmichelson@digium.com> - - * 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 20:14 +0000 [r149756] BJ Weschke <bweschke@btwtech.com> - - * configs/agents.conf.sample, /: Merged revisions 149683 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r149683 | bweschke | 2008-10-15 14:28:54 -0400 (Wed, 15 Oct 2008) - | 4 lines 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 19:07 +0000 [r149588-149687] Tilghman Lesher <tlesher@digium.com> - - * 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 13:52 +0000 [r149542] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Adding a note about a missing part of - "kill-the-user" - I got lost in the Ao2 world... We're going to - try to get time to fix this and kpfleming believes that there's - code in ao2 so that we can solve it... - -2008-10-15 11:26 +0000 [r149384-149487] Kevin P. Fleming <kpfleming@digium.com> - - * /, channels/chan_sip.c: Merged revisions 149452 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r149452 | kpfleming | 2008-10-15 12:30:40 +0200 (Wed, 15 Oct - 2008) | 3 lines fix some problems when parsing SIP messages that - have the maximum number of headers or body lines that we support - ........ - - * configure, configure.ac: reverting this change... had not read - the commit list yet, didn't realize the code had been upgraded - - * configure, configure.ac: do complete version check for SpanDSP, - since the app_fax code is not compatible with 0.0.6 yet - - * apps/app_stack.c: building this module depends on res_agi being - built as well - -2008-10-15 07:45 +0000 [r149342] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Fixing sytax errors ;-) - -2008-10-14 23:57 +0000 [r149201-149279] Mark Michelson <mmichelson@digium.com> - - * apps/app_dial.c, CHANGES: 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: Merged revisions 149266 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r149266 | mmichelson | 2008-10-14 18:43:58 -0500 (Tue, 14 Oct - 2008) | 4 lines Change this warning to an error message. - Suggestion comes from Sean Bright. Thanks Sean! ........ - - * /, channels/chan_sip.c: Merged revisions 149207 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r149207 | mmichelson | 2008-10-14 18:10:26 -0500 (Tue, 14 Oct - 2008) | 9 lines 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: Merged - revisions 149204 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r149204 | mmichelson | 2008-10-14 18:00:01 -0500 (Tue, 14 Oct - 2008) | 12 lines 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: Merged revisions 149200 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r149200 | mmichelson | 2008-10-14 17:40:42 -0500 (Tue, 14 Oct - 2008) | 12 lines 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 22:38 +0000 [r149199] Tilghman Lesher <tlesher@digium.com> - - * 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, - include/asterisk/hashtab.h, main/chanvars.c, main/config.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 21:08 +0000 [r149131] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 149130 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r149130 | mmichelson | 2008-10-14 15:49:02 -0500 (Tue, 14 Oct - 2008) | 7 lines Don't allow reserved characters to be used in - register lines in sip.conf. (closes issue #13570) Reported by: - putnopvut ........ - -2008-10-14 20:16 +0000 [r149062] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_waitforsilence.c: Merged revisions 149061 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r149061 | tilghman | 2008-10-14 15:09:06 -0500 (Tue, 14 Oct 2008) - | 6 lines 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 19:35 +0000 [r149040] Leif Madsen <lmadsen@digium.com> - - * 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 19:03 +0000 [r148917-148988] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 148987 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r148987 | tilghman | 2008-10-14 14:03:08 -0500 (Tue, 14 - Oct 2008) | 2 lines Some compilers warn, some don't. Fixing. - ........ - - * 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: Merged revisions 148916 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r148916 | tilghman | 2008-10-14 12:41:08 -0500 (Tue, 14 - Oct 2008) | 4 lines 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 17:38 +0000 [r148913] Mark Michelson <mmichelson@digium.com> - - * channels/chan_local.c, /: Merged revisions 148912 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r148912 | mmichelson | 2008-10-14 12:33:38 -0500 (Tue, - 14 Oct 2008) | 9 lines 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 15:15 +0000 [r148868] Tilghman Lesher <tlesher@digium.com> - - * 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 15:00 +0000 [r148867] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Fix reference count issue that Russell - brought up in SIP MWI NOTIFY support. Bump the reference count up - before we add it to the scheduler, duh. - -2008-10-14 14:18 +0000 [r148825] Doug Bailey <dbailey@digium.com> - - * phoneprov/polycom.xml: Allow MWI registration for all configured - lines. - -2008-10-14 11:31 +0000 [r148695-148754] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: fix some references to the owner of a - private structure that may not be present - - * Makefile, /: Merged revisions 148736 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r148736 | kpfleming | 2008-10-14 12:30:54 +0200 (Tue, 14 Oct - 2008) | 3 lines 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: this structure should be static - - * 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 - -2008-10-14 09:16 +0000 [r148679] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Adding some clarifications - -2008-10-14 08:06 +0000 [r148612] Kevin P. Fleming <kpfleming@digium.com> - - * /, main/translate.c: Merged revisions 148611 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r148611 | kpfleming | 2008-10-14 02:54:41 -0500 (Tue, 14 Oct - 2008) | 3 lines it would be nice if this message printing code - had actually been tested before it was committed... ........ - -2008-10-14 00:08 +0000 [r148570] Tilghman Lesher <tlesher@digium.com> - - * res/res_config_curl.c, res/res_config_pgsql.c, - res/res_config_odbc.c, include/asterisk/config.h, - res/res_realtime.c, include/asterisk/strings.h, - res/res_config_ldap.c, res/res_config_sqlite.c, main/config.c, - apps/app_voicemail.c: Merge realtime_update2 branch, which adds a - new realtime API call named 'update2', which permits updates - which match across multiple columns, instead of requiring all - tables to have a single unique identifier. All of the other API - calls with the exception of 'update' already had the ability to - match on multiple fields, so it was a missing and very desireable - feature that an API call implementing an update should have this, - too. This does not change any outward performance of Asterisk, - but it should make life easier for application developers who use - the RealTime framework. - -2008-10-13 17:14 +0000 [r148519] Steve Murphy <murf@digium.com> - - * 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 15:56 +0000 [r148471-148474] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: - Doxygen formatting. (tss tss) - Fixing - language - - * main/tcptls.c, channels/chan_sip.c: Highlightning even more bugs - in the current tcp/tls implementation. - - * channels/chan_sip.c: Sending a 403 after a 200 is considered very - bad. (found at SIPit) - -2008-10-12 09:19 +0000 [r148425] Michiel van Baak <michiel@vanbaak.info> - - * res/res_agi.c: fix the 'agi show commands' CLI function. (closes - issue #13666) Reported by: eliel Patches: res_agi.c.patch - uploaded by eliel (license 64) - -2008-10-10 21:21 +0000 [r148373-148376] Mark Michelson <mmichelson@digium.com> - - * 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 18:59 +0000 [r148268-148329] Tilghman Lesher <tlesher@digium.com> - - * pbx/pbx_config.c: Reset continuation items at the beginning of - each context (suggested by kpfleming). - - * CHANGES, pbx/pbx_config.c: Add keyword "same", which allows you - to create multiple steps in a dialplan, without needing to - respecify an extension pattern multiple times. (closes issue - #13632) Reported by: blitzrage Patches: - 20081006__bug13632.diff.txt uploaded by Corydon76 (license 14) - Tested by: blitzrage, Corydon76 - - * /, apps/app_voicemail.c: Merged revisions 148257 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r148257 | tilghman | 2008-10-10 11:25:31 -0500 (Fri, 10 - Oct 2008) | 7 lines 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 00:42 +0000 [r148200] Sean Bright <sean.bright@gmail.com> - - * include/asterisk.h, main/tdd.c, main/cryptostub.c, - res/res_config_sqlite.c, apps/app_voicemail.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 23:54 +0000 [r148144-148160] Mark Michelson <mmichelson@digium.com> - - * 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 23:25 +0000 [r148120] Tilghman Lesher <tlesher@digium.com> - - * 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 23:15 +0000 [r148112] Mark Michelson <mmichelson@digium.com> - - * /, main/features.c: Merged revisions 146026 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r146026 | murf | 2008-10-03 12:12:54 -0500 (Fri, 03 Oct 2008) | - 18 lines (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 21:47 +0000 [r148000-148071] Tilghman Lesher <tlesher@digium.com> - - * formats/format_wav.c, apps/app_minivm.c, channels/chan_agent.c, - main/file.c, res/res_monitor.c, apps/app_voicemail.c: Reverting - format addition for now - - * apps/app_minivm.c, channels/chan_agent.c, main/file.c, - res/res_monitor.c, apps/app_voicemail.c: Fudges for wav16, just - like wav49 - - * formats/format_wav.c: Add native 16kHz format for wav file - format. (Closes issue #13657) - - * sounds/sounds.xml, sounds/Makefile: Publish MOH files in sln16 - format - - * /, apps/app_voicemail.c: Merged revisions 147997 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r147997 | tilghman | 2008-10-09 14:38:33 -0500 (Thu, 09 - Oct 2008) | 4 lines When blank, callerid name and number should - display "unknown caller" in voicemail emails. (Closes issue - #13643) ........ - -2008-10-09 19:27 +0000 [r147952] Jeff Peeler <jpeeler@digium.com> - - * 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 17:48 +0000 [r147899] Michiel van Baak <michiel@vanbaak.info> - - * 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 17:46 +0000 [r147896] Tilghman Lesher <tlesher@digium.com> - - * 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 17:04 +0000 [r147854] Terry Wilson <twilson@digium.com> - - * phoneprov/000000000000.cfg, res/res_phoneprov.c, - configs/phoneprov.conf.sample: Make phoneprov case-insensitive to - remove the func_strings dependency of the default config - -2008-10-09 17:01 +0000 [r147853] Michiel van Baak <michiel@vanbaak.info> - - * channels/chan_dahdi.c, channels/chan_misdn.c, - channels/chan_h323.c: fix some CLI commands we borked during - devcon2008 Thanks rmudget for letting me know and providing hints - on how to fix it best. - -2008-10-09 14:17 +0000 [r147807] Steve Murphy <murf@digium.com> - - * main/pbx.c, include/asterisk.h, doc/CODING-GUIDELINES, - include/asterisk/autoconfig.h.in, channels/vcodecs.c, - main/translate.c, configure.ac, channels/console_video.c, - channels/chan_iax2.c, main/astobj2.c, channels/chan_oss.c, - main/rtp.c, main/config.c, main/cli.c, channels/chan_usbradio.c, - configure, channels/console_gui.c, utils/extconf.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 - -2008-10-09 01:43 +0000 [r147760-147761] Joshua Colp <jcolp@digium.com> - - * configs/sip.conf.sample: *whistle* - - * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add - support for subscribing to a voice mailbox on a remote SIP server - and making the new/old message count available to local devices. - (issue #AST-77) - -2008-10-08 22:32 +0000 [r147714] Mark Michelson <mmichelson@digium.com> - - * 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 22:26 +0000 [r147689] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 147681 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r147681 | kpfleming | 2008-10-08 17:22:09 -0500 (Wed, 08 - Oct 2008) | 3 lines 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 20:07 +0000 [r147635] Sean Bright <sean.bright@gmail.com> - - * configs/voicemail.conf.sample: Add some examples of IMAP - accounts. - -2008-10-08 19:08 +0000 [r147592] Tilghman Lesher <tlesher@digium.com> - - * 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 14:53 +0000 [r147518] Joshua Colp <jcolp@digium.com> - - * /, apps/app_speech_utils.c: Merged revisions 147517 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r147517 | file | 2008-10-08 11:51:42 -0300 (Wed, 08 Oct - 2008) | 2 lines If we receive DTMF make sure that the state of - the speech structure goes back to being not ready. (issue - #LUMENVOX-8) ........ - -2008-10-08 12:28 +0000 [r147476] Bradley Latus <brad.latus@gmail.com> - - * configs/iax.conf.sample: Adjust commented default trunkmtu value - to match documentation above it - -2008-10-08 12:15 +0000 [r147388-147457] Sean Bright <sean.bright@gmail.com> - - * funcs/func_curl.c, apps/app_meetme.c, cdr/cdr_adaptive_odbc.c, - res/res_odbc.c: Keep up with shadow warnings. One day I'll - actually enable this in the Makefile. - - * utils/Makefile: When echoing our copies, strip off ASTTOPDIR from - the front of the source file. - - * apps/app_dial.c, channels/chan_dahdi.c, channels/chan_iax2.c: - Move the DAHDI-to-DAHDI operator mode check from app_dial into - chan_dahdi so we don't have to hardcode anything. (closes issue - #13636) Reported by: seanbright Patches: 13636.diff uploaded by - seanbright (license 71) Reviewed by: russellb, putnopvut - -2008-10-07 20:15 +0000 [r147266-147347] Michiel van Baak <michiel@vanbaak.info> - - * configure, configure.ac: Make format_vorbis_ogg compile on - OpenBSD (closes issue #13639) Reported by: mvanbaak Patches: - 2008100700_oggsupportOBSD.diff.txt uploaded by mvanbaak (license - 7) 2008100700_oggsupportOBSD-configurescript.diff.txt uploaded by - mvanbaak (license 7) Tested by: mvanbaak - - * channels/Makefile: make this work on OpenBSD - - * configure, configure.ac: Make sure the configs on OpenBSD are in - /etc/asterisk by default (closes issue #13641) Reported by: jtodd - - * contrib/scripts/safe_asterisk_restart, - contrib/scripts/safe_asterisk: use pkill instead of killall to be - more portable - -2008-10-07 18:00 +0000 [r147265] Sean Bright <sean.bright@gmail.com> - - * apps/app_voicemail.c: This was flawed. The issue that I was - trying to address was addressed by adding the imapsecret alias - for imappassword. Will rethink this one and give it another shot - on a rainy day TBD. - -2008-10-07 17:49 +0000 [r147264] Michiel van Baak <michiel@vanbaak.info> - - * CHANGES: fix wording as pointed out by Corydon - -2008-10-07 17:44 +0000 [r147262] Tilghman Lesher <tlesher@digium.com> - - * UPGRADE.txt, include/asterisk/options.h, main/asterisk.c, - main/term.c: Allow people to select the old console behavior of - white text on a black background, by using the startup flag '-B'. - -2008-10-07 16:52 +0000 [r147191-147194] Sean Bright <sean.bright@gmail.com> - - * /, apps/app_voicemail.c: Merged revisions 147193 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r147193 | seanbright | 2008-10-07 12:48:30 -0400 (Tue, - 07 Oct 2008) | 2 lines Make 'imapsecret' an alias to - 'imappassword' in voicemail.conf. ........ - - * apps/app_voicemail.c: Or not. - - * apps/app_voicemail.c: There was a boo-boo in TFOT that is causing - some confusion on the mailing lists so include 'imapsecret' as an - alias to 'imappassword' (and print a little notice nudging users - toward the right option name). - -2008-10-07 16:04 +0000 [r147146] Jeff Peeler <jpeeler@digium.com> - - * 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 14:59 +0000 [r147050-147099] Sean Bright <sean.bright@gmail.com> - - * apps/app_voicemail.c: If we encounter something in mailbox - options that we don't grok, then spit out a warning instead of - just silently ignoring it. - - * 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 02:02 +0000 [r147011] Richard Mudgett <rmudgett@digium.com> - - * funcs/func_callerid.c: Independent change from branch issue8824 - that is not part of COLP. (-r142574 rmudgett) - -2008-10-07 00:02 +0000 [r146970] Terry Wilson <twilson@digium.com> - - * 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 23:21 +0000 [r146928] Tilghman Lesher <tlesher@digium.com> - - * include/asterisk/threadstorage.h: Update documentation; - AST_THREADSTORAGE() in trunk only takes a single argument. - -2008-10-06 23:14 +0000 [r146925] Michiel van Baak <michiel@vanbaak.info> - - * res/res_config_odbc.c, build_tools/menuselect-deps.in, configure, - funcs/func_odbc.c, include/asterisk/autoconfig.h.in, - configure.ac, cdr/cdr_adaptive_odbc.c, cdr/cdr_odbc.c, - makeopts.in, res/res_odbc.c, apps/app_voicemail.c: All ODBC parts - can now use either unixodbc or iodbc. This allows for the ODBC - parts to work on OpenBSD as well. 99.99% of the work is done by - seanbright (bow, bow) and I actually did nothing but test and - yell at him that it still didn't work :) Thanks for helping out ! - -2008-10-06 23:08 +0000 [r146875-146923] Jeff Peeler <jpeeler@digium.com> - - * main/features.c, res/res_agi.c, include/asterisk/features.h: - 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 21:18 +0000 [r146807] Michiel van Baak <michiel@vanbaak.info> - - * include/asterisk/endian.h: make aescrypt.c compile on OpenBSD - again - -2008-10-06 21:09 +0000 [r146802] Tilghman Lesher <tlesher@digium.com> - - * 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, channels/chan_iax2.c, - funcs/func_callerid.c, apps/app_speech_utils.c: Merged revisions - 146799 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r146799 | tilghman | 2008-10-06 15:52:04 -0500 (Mon, 06 Oct 2008) - | 8 lines 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) - ........ - -2008-10-06 17:32 +0000 [r146738] Sean Bright <sean.bright@gmail.com> - - * configure, configure.ac: Pretty-print a couple configure options - -2008-10-06 16:52 +0000 [r146713] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_local.c, /: Merged revisions 146711 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r146711 | tilghman | 2008-10-06 11:51:21 -0500 (Mon, 06 - Oct 2008) | 9 lines 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 16:03 +0000 [r146644] Kevin P. Fleming <kpfleming@digium.com> - - * /: Merged revisions 146643 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r146643 | kpfleming | 2008-10-06 10:57:49 -0500 (Mon, 06 Oct - 2008) | 8 lines 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 15:29 +0000 [r146640] Mark Michelson <mmichelson@digium.com> - - * configs/queues.conf.sample, CHANGES, apps/app_queue.c: This - commit introduces a change to how the "joinempty" and - "leavewhenempty" options are configured in queues.conf. Instead - of using vague terms like "yes," "no," "loose," and "strict," we - now accept a comma-separated list of values to determine when to - consider a member available. Extended details can be found in the - queues.conf.sample file. Note also that the above four referenced - values are still accepted for backwards-compatibility, but are - mapped internally to the new method of representing the option. - AST-105 - -2008-10-06 00:36 +0000 [r146555-146597] Sean Bright <sean.bright@gmail.com> - - * utils/Makefile: Make NOISY_BUILD work for the calls to cp in - utils/Makefile - - * utils/Makefile: Quote arguments to cp so we can handle spaces in - our paths. - -2008-10-05 22:11 +0000 [r146514] Russell Bryant <russell@digium.com> - - * utils/muted.c: Make this build on my mac. - -2008-10-05 21:21 +0000 [r146449] Jason Parker <jparker@digium.com> - - * /, channels/chan_sip.c: Recorded merge of revisions 146448 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r146448 | qwell | 2008-10-05 16:17:44 -0500 (Sun, 05 Oct 2008) | - 1 line Fix silly formatting. ........ - -2008-10-05 01:59 +0000 [r146312-146407] Sean Bright <sean.bright@gmail.com> - - * build_tools/make_buildopts_h: This is far from optimal, but I - just found a FreeBSD system without md5 installed on it. So look - around for all of the different binaries that we could possibly - use. I'd wager this gets completely replaced by someone else in - less than 24 hours... :) - - * main/asterisk.c: Fix a bug with the last item in CLI history - getting duplicated when read from the .asterisk_history file (and - subsequently being duplicated when written). We weren't checking - the result of fgets() which meant that we read the same line - twice before feof() actually returned non- zero. Also, stop - writing out an extra blank line between each item in the history - file, fix a minor off-by-one error, and use symbolic constants - rather than a hardcoded integer. - - * 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 22:40 +0000 [r146242] Jeff Peeler <jpeeler@digium.com> - - * main/features.c: remove superfluous reference counting operations - in manage_parkinglot since ao2_interator_next increments the ref - count automatically - -2008-10-03 22:10 +0000 [r146198] Sean Bright <sean.bright@gmail.com> - - * 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 - -2008-10-03 18:30 +0000 [r146081] Tilghman Lesher <tlesher@digium.com> - - * CHANGES: document meetme schedule changes (related to issue - #11040) - -2008-10-03 17:36 +0000 [r146053] Michiel van Baak <michiel@vanbaak.info> - - * CHANGES: put a note in CHANGES about the cli_cleanup done during - AstriDevCon - -2008-10-03 17:35 +0000 [r146052] Terry Wilson <twilson@digium.com> - - * main/dial.c: The dialing API should inherit datastores as well as - variables - -2008-10-02 19:30 +0000 [r145959-145962] Russell Bryant <russell@digium.com> - - * 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 18:02 +0000 [r145915] Michiel van Baak <michiel@vanbaak.info> - - * apps/app_meetme.c: fix the 'meetme list', 'meetme list concise', - 'meetme list $confno' and 'meetme list $confno concise' CLI - commands (closes issue #13586) Reported by: john8675309 Help and - feedback from eliel, thanks! - -2008-10-02 17:16 +0000 [r145846] Tilghman Lesher <tlesher@digium.com> - - * configs/func_odbc.conf.sample, funcs/func_odbc.c, CHANGES: Permit - the syntax and synopsis fields to be set (for func_odbc). - -2008-10-02 16:42 +0000 [r145842] Michiel van Baak <michiel@vanbaak.info> - - * apps/app_meetme.c: make this compile under devmode again - -2008-10-02 15:28 +0000 [r145771] Sean Bright <sean.bright@gmail.com> - - * configure, configure.ac: This is much cleaner, methinks. - -2008-10-02 15:17 +0000 [r145752] Tilghman Lesher <tlesher@digium.com> - - * /, res/res_odbc.c: Merged revisions 145751 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r145751 | tilghman | 2008-10-02 10:13:21 -0500 (Thu, 02 Oct 2008) - | 3 lines 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 23:48 +0000 [r145692] Sean Bright <sean.bright@gmail.com> - - * 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 23:02 +0000 [r145649] Tilghman Lesher <tlesher@digium.com> - - * apps/app_meetme.c, funcs/func_strings.c, - include/asterisk/localtime.h, main/stdtime/localtime.c: Add - schedule extensions to app_meetme. In addition, the reporter - found a problem within strptime(3), which we are correcting here - with ast_strptime(). (closes issue #11040) Reported by: DEA - Patches: 20080910__bug11040.diff.txt uploaded by Corydon76 - (license 14) Tested by: DEA - -2008-10-01 22:23 +0000 [r145553-145606] Mark Michelson <mmichelson@digium.com> - - * 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 17:26 +0000 [r145487] Leif Madsen <lmadsen@digium.com> - - * contrib/scripts/realtime_pgsql.sql, /: Merged revisions 145479 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r145479 | lmadsen | 2008-10-01 13:18:30 -0400 (Wed, 01 Oct 2008) - | 6 lines Update the realtime_pgsql.sql script to create the - setinterfacevar column. (closes issue #13549) Reported by: fiddur - ........ - -2008-10-01 15:44 +0000 [r145428] Tilghman Lesher <tlesher@digium.com> - - * apps/app_sms.c: Initializing buffer prevents a segfault when - arguments are incomplete. (closes issue #13471) Reported by: - alecdavis Patches: 20080916__bug13471.diff.txt uploaded by - Corydon76 (license 14) Tested by: alecdavis - -2008-10-01 14:44 +0000 [r145381] Mark Michelson <mmichelson@digium.com> - - * Makefile: Too many times have I mistyped the word 'install' as - 'isntall' Now this typo shall no longer be a problem since 'make - isntall' just builds the 'install' target. - -2008-10-01 12:29 +0000 [r145329] Russell Bryant <russell@digium.com> - - * CHANGES: tabs to spaces - -2008-09-30 22:21 +0000 [r145249] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_sip.c: (closes issue #13337) Reported by: pj Tested - by: pj Set transport to SIP_TRANSPORT_UDP mode if not specified - which fixes calls to get_transport returning UNKNOWN. - -2008-09-30 21:32 +0000 [r145226] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c, CHANGES: Add support for call pickup on Snom - phones. Asterisk now includes a magic call-id in the dialog-info - event package used with extension state subscriptions on Snom - phones. Then, when the phone sends an INVITE with Replaces for - the special callid, Asterisk will perform a pickup on the - extension that was subscribed to. The original code on this issue - was submitted by xylome. However, contributions have been made by - (at least) mgernoth and pkempgen. The final patch was written by - seanbright, and includes the necessary logic to allow this work - in a technology independent way. (closes issue #5014) Reported - by: xylome Patches: issue5014-trunk.diff uploaded by seanbright - (license 71) - -2008-09-30 21:00 +0000 [r145200] Richard Mudgett <rmudgett@digium.com> - - * channels/misdn/isdn_lib.h, doc/tex/misdn.tex, - channels/chan_misdn.c, channels/misdn/isdn_lib.c: * 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 - -2008-09-28 23:32 +0000 [r145121] Michiel van Baak <michiel@vanbaak.info> - - * channels/chan_unistim.c, res/res_config_pgsql.c, - apps/app_meetme.c, res/ais/clm.c, res/res_limit.c, - main/taskprocessor.c, channels/chan_console.c, apps/app_queue.c, - channels/chan_oss.c, main/astobj2.c, main/cli.c, - channels/chan_dahdi.c, main/manager.c, channels/chan_misdn.c, - channels/chan_features.c, res/res_agi.c, channels/chan_h323.c, - res/ais/evt.c, res/res_config_ldap.c, apps/app_mixmonitor.c, - res/res_clioriginate.c: Merge the cli_cleanup branch. This work - is done by lmadsen, junky and mvanbaak during AstriDevCon. This - is the second audit the CLI got, and this time lmadsen made sure - he had _ALL_ modules loaded that have CLI commands in them. - -2008-09-28 21:39 +0000 [r145076] Tilghman Lesher <tlesher@digium.com> - - * res/res_config_pgsql.c: Change several improper "sizeof" to - "strlen", as sizeof in that context would incorrectly use the - size of a pointer, rather than the length of a string. (Closes - issue #13574) - -2008-09-28 17:08 +0000 [r145027] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_dahdi.c: rename chandup() and clarify its usage - -2008-09-27 16:17 +0000 [r144949-144951] Kevin P. Fleming <kpfleming@digium.com> - - * utils/Makefile: remove incorrect comment - - * agi/Makefile, utils/Makefile, include/asterisk/astmm.h: fix bugs - caused by r144949 when MALLOC_DEBUG is defined - - * include/asterisk.h, /, main/Makefile, main/ast_expr2.y, - Makefile.moddir_rules, utils/astman.c, main/ast_expr2.c, - Makefile, utils/Makefile, main/ast_expr2f.c, pbx/pbx_ael.c, - main/astmm.c, utils/ael_main.c, main/stdtime/localtime.c, - utils/extconf.c, main/ast_expr2.fl: Merged revisions - 144924-144925 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r144924 | kpfleming | 2008-09-27 10:00:48 -0500 (Sat, 27 Sep - 2008) | 6 lines improve header inclusion process in a few small - ways: - it is no longer necessary to forcibly include - asterisk/autoconfig.h; every module already includes asterisk.h - as its first header (even before system headers), which serves - the same purpose - astmm.h is now included by asterisk.h when - needed, instead of being forced by the Makefile; this means - external modules will build properly against installed headers - with MALLOC_DEBUG enabled - simplify the usage of some of these - headers in the AEL-related stuff in the utils directory ........ - r144925 | kpfleming | 2008-09-27 10:13:30 -0500 (Sat, 27 Sep - 2008) | 2 lines fix some minor issues with rev 144924 ........ - -2008-09-27 00:49 +0000 [r144879] Michiel van Baak <michiel@vanbaak.info> - - * channels/chan_dahdi.c, apps/app_queue.c: fix a couple of CLI - commands that did not have a help description. - -2008-09-26 23:12 +0000 [r144829] Joshua Colp <jcolp@digium.com> - - * configs/rtp.conf.sample: Update documentation to include default - setting. This is for you jtodd! - -2008-09-26 18:02 +0000 [r144482-144681] Steve Murphy <murf@digium.com> - - * pbx/pbx_lua.c: (closes issue #13564) Reported by: mnicholson - Patches: pbx_lua9.diff uploaded by mnicholson (license 96) Many - thanks to Matt for his upgrade to the lua dialplan option! the - Description from the bug: This patch adds a stack trace to errors - encountered while executing lua extensions. The patch also - handles out of memory errors reported by lua. - - * main/pbx.c, /: Merged revisions 144677 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r144677 | murf | 2008-09-26 11:47:13 -0600 (Fri, 26 Sep 2008) | - 12 lines (closes issue #13563) Reported by: mnicholson Patches: - found1.diff uploaded by mnicholson (license 96) This patch was - mainly meant to apply to trunk and 1.6.x, but I'm applying it to - 1.4 also, which should be a perfectly harmless fix to the vast - majority of users who are not using external switches, but the - few who might be affected will not have to go to the pain of - filing a bug report. ........ - - * utils/build-extensions-conf.lua (removed): Matt suggests we - remove utils/build-extensions-conf.lua, as per bug 12961, it is - no longer necessary. - - * main/pbx.c, funcs/func_cut.c, channels/chan_oss.c, - apps/app_playback.c: (closes issue #13557) Reported by: - nickpeirson The user attached a patch, but the license is not yet - recorded. I took the liberty of finding and replacing ALL index() - calls with strchr() calls, and that involves more than just - main/pbx.c; chan_oss, app_playback, func_cut also had calls to - index(), and I changed them out. 1.4 had no references to index() - at all. - - * pbx/pbx_lua.c: (closes issue #13559) Reported by: mnicholson - Patches: pbx_lua8.diff uploaded by mnicholson (license 96) - - * pbx/pbx_lua.c, configs/extensions.lua.sample, - include/asterisk/hashtab.h: I added a little verbage to hashtab - for the hashtab_destroy func. It was pretty sparsely documented. - This update fleshes out the pbx_lua module, to add the switch - statements to the extensions in the extensions.lua file, as well - as removing them when the module is unloaded. Many thanks to Matt - Nicholson for his fine contribution! - - * pbx/pbx_lua.c: (closes issue #13558) Reported by: mnicholson - Considering that the example extensions.lua used nothing but - ["12345"] notation, and that the resulting error message: [Sep 24 - 17:01:16] ERROR[12393]: pbx_lua.c:1204 exec: Error executing lua - extension: attempt to call a nil value is not very informative as - to the nature of the problem, I think this bug fix is a big win! - -2008-09-25 01:46 +0000 [r144357] Tilghman Lesher <tlesher@digium.com> - - * /: Recorded merge of revisions 144356 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r144356 | tilghman | 2008-09-24 20:44:47 -0500 (Wed, 24 Sep 2008) - | 6 lines Backport Hebrew language to voicemail. (closes issue - #13155) Reported by: greenfieldtech Patches: - voicemail-hebrew-patch-1.4-SVN.c.patch uploaded by greenfieldtech - (license 369) ........ - -2008-09-24 22:05 +0000 [r144314] Doug Bailey <dbailey@digium.com> - - * res/res_phoneprov.c: Blanch the 404 error message for those with - no sense of humor - -2008-09-24 08:42 +0000 [r144257] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /: Merged revisions 144238 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r144238 | crichter | 2008-09-24 10:20:52 +0200 (Mi, 24 - Sep 2008) | 1 line improved helptext of misdn_set_opt. ........ - -2008-09-24 06:43 +0000 [r144199] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_curl.c: Create a 'hashcompat' option that permits the - results of a CURL() able to be passed directly into the HASH() - function. Requested via the -users list, and committed at - Astricon in the Code Zone. - -2008-09-23 23:33 +0000 [r144149] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: Fix a conflict in flag values - -2008-09-23 16:52 +0000 [r144067] Steve Murphy <murf@digium.com> - - * /, main/features.c: Merged revisions 144066 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r144066 | murf | 2008-09-23 10:41:49 -0600 (Tue, 23 Sep 2008) | - 29 lines (closes issue #13489) Reported by: DougUDI Tested by: - murf (closes issue #13490) Reported by: seanbright Tested by: - murf (closes issue #13467) Reported by: edantie Tested by: murf, - edantie, DougUDI This crash happens because we are unsafely - handling old pointers. The channel whose cdr is being handled, - has been hung up and destroyed already. I reorganized the code a - bit, and tried not to lose the fork-cdr-chain concepts of the - previous code. I now verify that the 'previous' channel (the - channel we had when the bridge was started), still exists, by - looking it up by name in the channel list. I also do not try to - reset the CDR's of channels involved in bridges. Testing shows it - solves the crash problem, and should not negatively impact - previous fixes involving CDR's generated during/after blind - transfers. (The reason we need to reset the CDR's on the - "beginning" channels in the first place). ........ - -2008-09-23 15:37 +0000 [r144025] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: When a promiscuous redirect contained both a - user and host portion in the Contact URI and specifies a - transport, the parsing done in parse_moved_contact resulted in a - malformed URI. This commit fixes the parsing so that a proper - Dial string may be formed when the forwarded call is placed. - (closes issue #13523) Reported by: mattdarnell Patches: - 13523v2.patch uploaded by putnopvut (license 60) Tested by: - mattdarnell - -2008-09-22 22:50 +0000 [r143904] Sean Bright <sean.bright@gmail.com> - - * /, formats/format_pcm.c: Merged revisions 143903 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r143903 | seanbright | 2008-09-22 18:49:00 -0400 (Mon, - 22 Sep 2008) | 8 lines Use the advertised header size in .au - files instead of just assuming they are 24 bytes (the minimum). - (closes issue #13450) Reported by: jamessan Patches: - pcm-header.diff uploaded by jamessan (license 246) ........ - -2008-09-21 09:53 +0000 [r143799-143843] Michiel van Baak <michiel@vanbaak.info> - - * doc/tex/privacy.tex: fix privacymanager example so it shows how - to use the PRIVACYMRGSTATUS variable - - * doc/tex/privacy.tex: document the new context argument for - privacymanager so people can do pattern matching on the input - - * doc/tex/privacy.tex: fix privacy documentation. We no longer do - priority jumping +101 - - * channels/chan_skinny.c: make 'module unload chan_skinny.so' - actually work. (closes issue #13524) Reported by: wedhorn - Patches: unload.diff uploaded by wedhorn (license 30) With small - tweak by me to prevent a crash - -2008-09-20 00:52 +0000 [r143737] Sean Bright <sean.bright@gmail.com> - - * /, contrib/scripts/vmail.cgi: Merged revisions 143736 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r143736 | seanbright | 2008-09-19 20:50:10 -0400 (Fri, 19 Sep - 2008) | 9 lines Make vmail.cgi work with mailboxes defined in - users.conf, too. (closes issue #13187) Reported by: netvoice - Patches: 20080911__bug13187.diff.txt uploaded by Corydon76 - (license 14) (Slightly modified to take alchamist's comments on - mantis into account) Tested by: msales, alchamist, seanbright - ........ - -2008-09-19 21:41 +0000 [r143697] Steve Murphy <murf@digium.com> - - * /: This blocks 143674 from trunk; it appears to already done in - trunk, since ast_odbc_direct_execute creates a new stmt for each - attempt. - -2008-09-19 15:43 +0000 [r143609] Mark Michelson <mmichelson@digium.com> - - * channels/chan_agent.c: We should only unsubscribe to the device - state event subscription if we have previously subscribed. - Otherwise a segfault will occur. (closes issue #13476) Reported - by: jonnt Patches: 13476.patch uploaded by putnopvut (license 60) - Tested by: jonnt - -2008-09-18 23:41 +0000 [r143559] Steve Murphy <murf@digium.com> - - * /, channels/chan_sip.c: Merged revisions 143534 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r143534 | murf | 2008-09-18 16:11:51 -0600 (Thu, 18 Sep 2008) | 1 - line A micro-fix, in sip_park_thread, where d is freed before the - func is done using it. ........ - -2008-09-17 20:57 +0000 [r143405] Tilghman Lesher <tlesher@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 143404 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r143404 | tilghman | 2008-09-17 15:55:47 -0500 (Wed, 17 - Sep 2008) | 6 lines When callerid is blank, we want to use - "unknown caller" in those cases, too. (closes issue #13486) - Reported by: tomo1657 Patches: 20080917__bug13486.diff.txt - uploaded by Corydon76 (license 14) ........ - -2008-09-17 20:25 +0000 [r143340-143400] Mark Michelson <mmichelson@digium.com> - - * main/astmm.c: If attempting to free a NULL pointer when - MALLOC_DEBUG is set, don't bother searching for a region to free, - just immediately exit. This has the dual benefit of suppressing a - warning message about freeing memory at (nil) and of optimizing - the free() replacement by not having to do any futile searching - for the proper region to free. (closes issue #13498) Reported by: - pj Patches: 13498.patch uploaded by putnopvut (license 60) Tested - by: pj - - * /, main/rtp.c: Merged revisions 143337 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r143337 | mmichelson | 2008-09-17 13:24:15 -0500 (Wed, 17 Sep - 2008) | 6 lines Allow for "G.729" if offered in an SDP even - though it is not RFC 3551 compliant. Some Cisco switches will - send this in an SDP, and it doesn't hurt to be able to accept - this. ........ - -2008-09-15 21:31 +0000 [r143141] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 143140 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r143140 | tilghman | 2008-09-15 16:29:32 -0500 (Mon, 15 - Sep 2008) | 6 lines Set the raw formats at the same time as the - other formats. (closes issue #13240) Reported by: jvandal - Patches: 20080813__bug13240.diff.txt uploaded by Corydon76 - (license 14) ........ - -2008-09-14 22:16 +0000 [r143082] Michiel van Baak <michiel@vanbaak.info> - - * channels/chan_skinny.c: plug a couple of memleaks in chan_skinny. - (closes issue #13452) Reported by: pj Patches: memleak5.diff - uploaded by wedhorn (license 30) Tested by: wedhorn, pj, mvanbaak - (closes issue #13294) Reported by: pj - -2008-09-13 14:15 +0000 [r143034] Sean Bright <sean.bright@gmail.com> - - * apps/app_osplookup.c: Everytime a compile fails, a puppy dies. - -2008-09-13 13:54 +0000 [r142992-143031] Tilghman Lesher <tlesher@digium.com> - - * apps/app_dial.c, channels/chan_iax2.c, channels/iax2-parser.c: - Repair IAXVAR implementation so that it works again (regression?) - (closes issue #13354) Reported by: adomjan Patches: - 20080828__bug13354.diff.txt uploaded by Corydon76 (license 14) - 20080829__bug13354__1.6.0.diff.txt uploaded by Corydon76 (license - 14) Tested by: Corydon76, adomjan - - * channels/chan_unistim.c, main/udptl.c, apps/app_meetme.c, - res/res_snmp.c, codecs/codec_adpcm.c, res/res_phoneprov.c, - codecs/codec_gsm.c, apps/app_alarmreceiver.c, - channels/chan_gtalk.c, res/res_http_post.c, - res/res_musiconhold.c, channels/chan_iax2.c, apps/app_followme.c, - res/res_jabber.c, main/enum.c, res/res_config_sqlite.c, - main/config.c, main/loader.c, main/cdr.c, channels/chan_dahdi.c, - channels/chan_phone.c, res/res_smdi.c, main/manager.c, - funcs/func_config.c, apps/app_osplookup.c, - channels/chan_skinny.c, funcs/func_odbc.c, main/features.c, - apps/app_minivm.c, main/http.c, channels/chan_alsa.c, - apps/app_amd.c, apps/app_directory.c, res/res_config_ldap.c, - apps/app_rpt.c, channels/chan_mgcp.c, codecs/codec_lpc10.c, - res/res_config_pgsql.c, main/dnsmgr.c, codecs/codec_g722.c, - channels/chan_sip.c, apps/app_festival.c, codecs/codec_speex.c, - codecs/codec_alaw.c, res/res_adsi.c, include/asterisk/config.h, - channels/chan_agent.c, codecs/codec_g726.c, - channels/chan_console.c, apps/app_queue.c, channels/chan_oss.c, - main/rtp.c, apps/app_playback.c, channels/chan_jingle.c, - channels/chan_h323.c, codecs/codec_ulaw.c, codecs/codec_dahdi.c, - res/res_indications.c, main/asterisk.c, res/res_odbc.c, - main/dsp.c, apps/app_voicemail.c: Create a new config file - status, CONFIG_STATUS_FILEINVALID for differentiating when a file - is invalid from when a file is missing. This is most important - when we have two configuration files. Consider the following - example: Old system: sip.conf users.conf Old result New result - ======== ========== ========== ========== Missing Missing SIP - doesn't load SIP doesn't load Missing OK SIP doesn't load SIP - doesn't load Missing Invalid SIP doesn't load SIP doesn't load OK - Missing SIP loads SIP loads OK OK SIP loads SIP loads OK Invalid - SIP loads incompletely SIP doesn't load Invalid Missing SIP - doesn't load SIP doesn't load Invalid OK SIP doesn't load SIP - doesn't load Invalid Invalid SIP doesn't load SIP doesn't load So - in the case when users.conf doesn't load because there's a typo - that disrupts the syntax, we may only partially load users, - instead of failing with an error, which may cause some calls not - to get processed. Worse yet, the old system would do this with no - indication that anything was even wrong. (closes issue #10690) - Reported by: dtyoo Patches: 20080716__bug10690.diff.txt uploaded - by Corydon76 (license 14) - -2008-09-12 22:24 +0000 [r142929] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_local.c, /: Merged revisions 142927 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r142927 | jpeeler | 2008-09-12 17:22:28 -0500 (Fri, 12 - Sep 2008) | 6 lines (closes issue #12965) Reported by: rlsutton2 - Prevents local channels from playing MOH at each other which was - causing ast_generic_bridge to loop much faster. ........ - -2008-09-12 20:49 +0000 [r142866] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions - 142865 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r142865 | tilghman | 2008-09-12 15:37:18 -0500 (Fri, 12 Sep 2008) - | 11 lines Create rules for disallowing contacts at certain - addresses, which may improve the security of various - installations. As this does not change any default behavior, it - is not classified as a direct security fix for anything within - Asterisk, but may help PBX admins better secure their SIP - servers. (closes issue #11776) Reported by: ibc Patches: - 20080829__bug11776.diff.txt uploaded by Corydon76 (license 14) - Tested by: Corydon76, blitzrage ........ - -2008-09-12 18:22 +0000 [r142808] Michiel van Baak <michiel@vanbaak.info> - - * /: Recorded merge of revisions 142807 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r142807 | mvanbaak | 2008-09-12 19:59:25 +0200 (Fri, 12 Sep 2008) - | 2 lines fix copyright year range ........ - -2008-09-12 16:54 +0000 [r142741-142748] Tilghman Lesher <tlesher@digium.com> - - * main/app.c: When checking for an encoded character, make sure the - string isn't blank, first. (Closes issue #13470) - - * /, apps/app_voicemail.c: Merged revisions 142744 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r142744 | tilghman | 2008-09-12 11:38:02 -0500 (Fri, 12 - Sep 2008) | 4 lines Missing merge from 1.2 fixes errant exit on - DTMF, only when language is Italian (cf commit 34242) (Closes - issue #7353) ........ - - * /, main/file.c: Merged revisions 142740 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r142740 | tilghman | 2008-09-12 11:27:32 -0500 (Fri, 12 Sep 2008) - | 4 lines Don't return a free'd pointer, when a file cannot be - opened. (closes issue #13462) Reported by: wackysalut ........ - -2008-09-12 04:50 +0000 [r142676] Steve Murphy <murf@digium.com> - - * apps/app_dial.c, main/pbx.c, /, main/features.c, - include/asterisk/channel.h, apps/app_queue.c: Merged revisions - 142675 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r142675 | murf | 2008-09-11 22:29:34 -0600 (Thu, 11 Sep 2008) | - 29 lines Tested by: sergee, murf, chris-mac, andrew, KNK This is - a "second attempt" to restore the previous "endbeforeh" behavior - in 1.4 and up. In order to capture information concerning all the - legs of transfers in all their infinite combinations, I was - forced to this particular solution by a chain of logical - necessities, the first being that I was not allowed to rewrite - the CDR mechanism from the ground up! This change basically - leaves the original machinery alone, which allows IVR and local - channel type situations to generate CDR's as normal, but a - channel flag can be set to suppress the normal running of the h - exten. That flag would be set by the code that runs the h exten - from the ast_bridge_call routine, to prevent the h exten from - being run twice. Also, a flag in the ast_bridge_config struct - passed into ast_bridge_call can be used to suppress the running - of the h exten in that routine. This would happen, for instance, - if you use the 'g' option in the Dial app. Running this routine - 'early' allows not only the CDR() func to be used in the h - extension for reading CDR variables, but also allows them to be - modified before the CDR is posted to the backends. While I dearly - hope that this patch overcomes all problems, and introduces no - new problems, reality suggests that surely someone will have - problems. In this case, please re-open 13251 (or 13289), and - we'll see if we can't fix any remaining issues. ** trunk note: - some code to suppress the h exten being run from app_queue was - added; for the 'continue' option available only in trunk/1.6.x. - ........ - -2008-09-12 00:49 +0000 [r142635] Sean Bright <sean.bright@gmail.com> - - * cdr/cdr_adaptive_odbc.c: Build under dev-mode - -2008-09-11 23:12 +0000 [r142576] Steve Murphy <murf@digium.com> - - * /, main/features.c: Merged revisions 142575 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r142575 | murf | 2008-09-11 16:55:49 -0600 (Thu, 11 Sep 2008) | - 20 lines (closes issue #13364) Reported by: mdu113 Well, - fundamentally, the problems revealed in 13364 are because of the - ForkCDR call that is done before the dial. When the bridge is in - place, it's dealing with the first (and wrong) cdr in the list. - So, I wrote a little func to zip down to the first non-locked cdr - in the chain, and thru-out the ast_bridge_call, these results are - used instead of raw chan->cdr and peer->cdr pointers. This - shouldn't affect anyone who isn't forking cdrs before a dial, and - should correct the cdr's of those that do. So, this change ends - up correcting the dstchannel and userfield; the disposition was - fixed by a previous patch, it was OK coming into this problem. - ........ - -2008-09-11 21:45 +0000 [r142536] Tilghman Lesher <tlesher@digium.com> - - * cdr/cdr_adaptive_odbc.c, configs/cdr_adaptive_odbc.conf.sample: - Add usegmtime, as per the recent -users list discussion, and also - add my explanation to the file, since that additional text helps - people understand the concept. - -2008-09-10 22:11 +0000 [r142475] Steve Murphy <murf@digium.com> - - * /, main/features.c: Merged revisions 142474 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r142474 | murf | 2008-09-10 15:58:17 -0600 (Wed, 10 Sep 2008) | - 30 lines (closes issue #12318) Reported by: krtorio I made a - small change to the code that handles local channel situations. - In that code, I copy the answer time from the peer cdr, to the - bridge_cdr, but I wasn't also copying the disposition from the - peer cdr. So, Now I copy the disposition, and I've tested against - these cases: 1. phone 1 never answers the phone; no cdr is - generated at all. this should show up as a manager command - failure or something. 2. phone 2 never answers. CDR is generated, - says NO ANSWER 3. phone 2 is busy. CDR is generated, says BUSY 4. - phone 2 answers: CDR is generated, times are correct; disposition - is ANSWERED, which is correct. The start time is the time that - the manager dialed the first phone. The answer time is the time - the second phone picks up. I purposely left the cid and src - fields blank; since this call really originates from the manager, - there is no 'easy' data to put in these fields. If you feel - strongly that these fields should be filled in, re-open this bug - and I'll dig further. ........ - -2008-09-10 19:09 +0000 [r142417] Sean Bright <sean.bright@gmail.com> - - * /, configure, acinclude.m4: Merged revisions 142416 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r142416 | seanbright | 2008-09-10 15:05:46 -0400 (Wed, - 10 Sep 2008) | 9 lines Fix detection of PWLIB and OpenH323 - version when spacing in the headers isn't consistent. (closes - issue #13426) Reported by: bamby Patches: detect_openh323.diff - uploaded by bamby (license 430) (Modified by me to use sed - instead of tr) ........ - -2008-09-10 16:55 +0000 [r142359] Tilghman Lesher <tlesher@digium.com> - - * /, sounds/Makefile: Merged revisions 142358 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r142358 | tilghman | 2008-09-10 11:54:29 -0500 (Wed, 10 Sep 2008) - | 2 lines Publish new extra sounds version. ........ - -2008-09-10 16:41 +0000 [r142318-142355] Russell Bryant <russell@digium.com> - - * /, main/sched.c: Merged revisions 142354 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r142354 | russell | 2008-09-10 11:39:53 -0500 (Wed, 10 Sep 2008) - | 7 lines It is a normal situation that a task gets put in the - scheduler that should run as soon as possible. Accept "0" as an - acceptable time to run, and also treat negative as "run now", and - don't print a debug message about it. (inspired by a message - asking about the "request to schedule in the past" debug message - on the -dev list) ........ - - * CHANGES: Move last change to CHANGES up to the 1.6.2 section - -2008-09-09 22:08 +0000 [r142280] Philippe Sultan <philippe.sultan@gmail.com> - - * configs/jabber.conf.sample, CHANGES, res/res_jabber.c: Disable - autoprune by default. (closes issue #13411) Reported by: caio1982 - Patches: res_jabber_autoprune1.diff uploaded by caio1982 (license - 22) Tested by: caio1982 - -2008-09-09 19:16 +0000 [r142219] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: Merged revisions 142218 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r142218 | mmichelson | 2008-09-09 14:15:28 -0500 (Tue, 09 Sep - 2008) | 14 lines Make sure that the branch sent in CANCEL - requests matches the branch of the INVITE it is cancelling. - (closes issue #13381) Reported by: atca_pres Patches: - 13381v2.patch uploaded by putnopvut (license 60) Tested by: - atca_pres (closes issue #13198) Reported by: rickead2000 Tested - by: rickead2000 ........ - -2008-09-09 17:30 +0000 [r142181] Richard Mudgett <rmudgett@digium.com> - - * main/callerid.c: Cleaned up comment - -2008-09-09 17:15 +0000 [r142080-142146] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: This is the trunk version of the patch to close - issue 12979. The difference between this and the 1.6.0 and 1.6.1 - versions is that this is a much more invasive change. With this, - we completely get rid of the interfaces list, along with all its - helper functions. Let me take a moment to say that this change - personally excites me since it may mean huge steps forward - regarding proper lock order in app_queue without having to strew - seemingly unnecessary locks all over the place. It also results - in a huge reduction in lines of code and complexity. Way to go - Brett! (closes issue #12979) Reported by: sigxcpu Patches: - 20080710_issue12979_queue_custom_state_interface_trunk_2.diff - uploaded by bbryant (license 36) Tested by: sigxcpu, putnopvut - - * /, channels/chan_sip.c: Merged revisions 142079 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r142079 | mmichelson | 2008-09-09 11:19:17 -0500 (Tue, 09 Sep - 2008) | 21 lines When determining if codecs used by SIP peers - allow the media to be natively bridged, use the jointcapability - instead of the peercapability. It seems that the intent of using - the peercapability was to expand the choice of codecs for the - call to increase the chances of being able to native bridge the - channels. The problem is that if a codec were settled on for the - native bridge and that wasn't a codec that was configured to be - used by Asterisk for that peer, then Asterisk would send a - REINVITE with no codecs in the SDP which is a bug no matter how - you slice it. (closes issue #13076) Reported by: ramonpeek - Patches: 13076.patch uploaded by putnopvut (license 60) Tested - by: tbelder ........ - -2008-09-09 15:44 +0000 [r142064] Russell Bryant <russell@digium.com> - - * /, main/features.c: 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 [r142000] Bradley Latus <brad.latus@gmail.com> - - * include/asterisk/astobj2.h: Minor fix to doco - -2008-09-09 12:32 +0000 [r141995-141998] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Use ast_debug for debug messages. I was - wondering why debug messages weren't showing up when I had set - the debug level high for just app_queue.c. It's because we were - only checking the global option_debug variable instead of using - the awesome macro which checks both the global and file-specific - value - - * channels/chan_oss.c: 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 01:47 +0000 [r141949] Russell Bryant <russell@digium.com> - - * main/channel.c: 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 23:00 +0000 [r141810-141906] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Optimization: The only reason we should check - member status is if the queue has a joinempty or a leavewhenempty - setting which could cause the caller to not join the queue or - exit the queue. Prior to this patch, we could potentially - traverse the entire queue's member list for no reason since even - if the members are currently not available in some way we're - going to let the caller join the queue anyway. - - * channels/chan_sip.c: Um, apparently I didn't actually finish - merging before committing. Bad bad bad - - * /, channels/chan_sip.c: Merged revisions 141809 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r141809 | mmichelson | 2008-09-08 16:10:10 -0500 (Mon, 08 Sep - 2008) | 14 lines Fix pedantic mode of chan_sip to only check the - remote tag of an endpoint once a dialog has been confirmed. Up - until that point, it is possible and legal for the far-end to - send provisional responses with a different To: tag each time. - With this patch applied, these provisional messages will not - cause a matching problem. (closes issue #11536) Reported by: ibc - Patches: 11536v2.patch uploaded by putnopvut (license 60) - ........ - -2008-09-08 21:05 +0000 [r141807] Russell Bryant <russell@digium.com> - - * main/pbx.c, /: 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 20:18 +0000 [r141745] Jason Parker <jparker@digium.com> - - * Makefile, /, redhat (removed): 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 17:13 +0000 [r141682] Sean Bright <sean.bright@gmail.com> - - * build_tools/make_buildopts_h: 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 - -2008-09-07 00:04 +0000 [r141626] Michiel van Baak <michiel@vanbaak.info> - - * funcs/func_curl.c: make func_curl.c compile under devmode. - -2008-09-06 20:19 +0000 [r141566] Steve Murphy <murf@digium.com> - - * /, channels/chan_sip.c: 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 [r141504-141507] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_curl.c: Get rid of the casts that cause warnings on - OpenBSD. The compiler is errantly detecting warnings when we - redefine a structure each time it is used, even though the - structure is identical. Reported by: mvanbaak, via #asterisk-dev - - * /, res/res_agi.c: 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 - ........ - -2008-09-06 12:03 +0000 [r141464] Michiel van Baak <michiel@vanbaak.info> - - * channels/chan_sip.c, channels/chan_iax2.c, main/cli.c: Some fixes - to autocompletion in some commands. Changes applied by this - patch: - Fix autocompletion in 'sip prune realtime', sip peers - where never auto completed. Now we complete this command with: - 'sip prune realtime peer' -> all | like | sip peers Also I have - modified the syntax in the usage, was wrong... - Pass - ast_cli_args->argv and ast_cli_args->argc while running - autocompletion on CLI commands (CLI_GENERATE). With this we avoid - comparisons on ast_cli_args->line like this: strcasestr(a->line, - " description") strcasestr(a->line, "descriptions ") - strcasestr(a->line, "realtime peer"), and so on.. Making the code - more confusing (check the spaces in description!). The only thing - we must be sure is to first check a->pos or a->argc. - Fix 'iax2 - prune realtime' autocompletion, now we autocomplete this command - with 'all' & 'iax2 peers', check a look that iax2 peers where all - the peers, now only the ones in the cache.. (closes issue #13133) - Reported by: eliel Patches: clichanges.patch uploaded by eliel - (license 64) - -2008-09-05 22:03 +0000 [r141367-141425] Mark Michelson <mmichelson@digium.com> - - * funcs/func_curl.c: Fix func_curl compilation - - * /, channels/chan_agent.c: 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 19:12 +0000 [r141328] Tilghman Lesher <tlesher@digium.com> - - * funcs/func_curl.c, CHANGES: Add the CURLOPT dialplan function, - which permits setting various options for use with the CURL - dialplan function. (closes issue #12920) Reported by: davevg - Patches: 20080904__bug12920.diff.txt uploaded by Corydon76 - (license 14) Tested by: Corydon76, davevg - -2008-09-05 14:18 +0000 [r141115-141157] Steve Murphy <murf@digium.com> - - * main/channel.c, /: 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 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 17:27 +0000 [r141039] Jeff Peeler <jpeeler@digium.com> - - * /, main/features.c, res/res_agi.c: 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 20:16 +0000 [r140975] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Fix some locking order issues in app_queue. - This was brought up by atis on IRC a while ago. - -2008-09-03 18:06 +0000 [r140938] Michiel van Baak <michiel@vanbaak.info> - - * channels/chan_skinny.c, CHANGES: Added 'skinny show lines - verbose' This will print the subs and their status for every line - (if any). wedhorn did most of the work with his patch which - introduced 'skinny show debug' but a discussion on IRC stated - that it should be added to 'skinny show lines' Input on the - output format by Qwell on IRC. (closes issue #13344) Reported by: - wedhorn - -2008-09-03 14:41 +0000 [r140860-140887] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Fix compilation - - * /, apps/app_voicemail.c: Merged revisions 140850 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r140850 | mmichelson | 2008-09-03 09:29:15 -0500 (Wed, - 03 Sep 2008) | 9 lines Fix voicemail forwarding when using ODBC - storage. (closes issue #13387) Reported by: moliveras Patches: - 13387.patch uploaded by putnopvut (license 60) Tested by: - putnopvut, moliveras ........ - -2008-09-03 14:01 +0000 [r140824] Steve Murphy <murf@digium.com> - - * res/ael/pval.c, main/pbx.c, res/ael/ael.tab.c, res/ael/ael.y, - res/ael/ael.tab.h: In these changes, I have added some - explanation of changes to the Set and MSet apps, so people aren't - so shocked and surprised when they upgrade from 1.4 to 1.6. Also, - for the sake of those upgrading from 1.4 to 1.6 with AEL, I - provide automatic support for the "old" way of using Set(), that - still does the exact same old thing with quotes and backslashes - and so on as 1.4 did, by having AEL compile in the use of MSet() - instead of Set(), everywhere it inserts this code. But, if the - app_set var is set to 1.6 or higher, it uses the "new", - non-evaluative Set(). This only usually happens if the user - manually inserts this into the asterisk.conf file, or runs the - "make samples" command. - -2008-09-03 13:48 +0000 [r140821] Sean Bright <sean.bright@gmail.com> - - * cdr/cdr_sqlite.c: Move some duplicated code into a separate - function. Also try to do some wacky stuff in the commit message, - like: a newline \n a bell \a a tab \t a format specification %p - That is all. - -2008-09-03 13:41 +0000 [r140817-140820] Russell Bryant <russell@digium.com> - - * main/pbx.c: Formatting change to test something on the svn server - - * /, main/poll.c: 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 23:48 +0000 [r140752] Mark Michelson <mmichelson@digium.com> - - * /, apps/app_voicemail.c: Merged revisions 140751 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r140751 | mmichelson | 2008-09-02 18:47:49 -0500 (Tue, - 02 Sep 2008) | 6 lines After adding the context checking to - app_voicemail for IMAP storage, I left out a crucial place to - copy the context to the vm_state structure. This is the - correction. ........ - -2008-09-02 23:44 +0000 [r140691-140749] Steve Murphy <murf@digium.com> - - * main/cdr.c, /: 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 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 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. ........ - -2008-09-02 18:15 +0000 [r140606] Sean Bright <sean.bright@gmail.com> - - * /, channels/chan_iax2.c: 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 15:11 +0000 [r140563-140566] Russell Bryant <russell@digium.com> - - * codecs/codec_resample.c, apps/app_jack.c: Update instructions for - getting libresample - - * res/ais/lck.c (removed), res/ais/ckpt.c (removed), res/ais/amf.c - (removed): I'm not sure how these files got to trunk (probably my - fault), but they should not be here - -2008-09-02 14:41 +0000 [r140559] Sean Bright <sean.bright@gmail.com> - - * channels/chan_sip.c: When a call is rejected because of - call-limit, the channel driver is behaving as expected, so we - shouldn't report it as an error. Change to LOG_NOTICE instead. - -2008-08-29 17:53 +0000 [r140491] Jeff Peeler <jpeeler@digium.com> - - * main/features.c, CHANGES: Added the option s to the Park - application which will silence the announcement of the parking - space number. Also, fixes the bug of just clearing the flags - instead of actually parsing the arguments to Park. - -2008-08-29 17:47 +0000 [r140418-140489] Mark Michelson <mmichelson@digium.com> - - * main/manager.c, res/ais/lck.c, /, channels/chan_sip.c, - funcs/func_dialgroup.c, res/res_timing_pthread.c, - main/features.c, res/res_phoneprov.c, utils/hashtest2.c, - channels/chan_console.c, main/taskprocessor.c, apps/app_queue.c, - channels/chan_iax2.c, main/config.c: Merged revisions 140488 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r140488 | mmichelson | 2008-08-29 12:34:17 -0500 (Fri, 29 Aug - 2008) | 22 lines After working on the ao2_containers branch, I - noticed something a bit strange. In all cases where we provide a - callback function to ao2_container_alloc, the callback function - would only return 0 or CMP_MATCH. After inspecting the - ao2_callback() code carefully, I found that if you're only - looking for one specific item, then you should return CMP_MATCH | - CMP_STOP. Otherwise, astobj2 will continue traversing the current - bucket until the end searching for more matches. In cases like - chan_iax2 where in 1.4, all the peers are shoved into a single - bucket, this makes for potentially terrible performance since the - entire bucket will be traversed even if the peer is one of the - first ones come across in the bucket. All the changes I have made - were for cases where the callback function defined was passed to - ao2_container_alloc so that calls to ao2_find could find a unique - instance of whatever object was being stored in the container. - ........ - - * main/file.c: Allow for video files to be opened as well as audio - files. (closes issue #13372) Reported by: epicac Patches: - 13372.patch uploaded by putnopvut (license 60) Tested by: epicac - - * /, apps/app_voicemail.c: Merged revisions 140421 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r140421 | mmichelson | 2008-08-29 11:01:07 -0500 (Fri, - 29 Aug 2008) | 12 lines Add context checking when retrieving a - vm_state. This was causing a problem for people who had - identically named mailboxes in separate voicemail contexts. This - commit affects IMAP storage only. (closes issue #13194) Reported - by: moliveras Patches: 13194.patch uploaded by putnopvut (license - 60) Tested by: putnopvut, moliveras ........ - - * channels/chan_sip.c: Merged revisions 140417 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r140417 | mmichelson | 2008-08-29 10:26:52 -0500 (Fri, 29 Aug - 2008) | 10 lines Fix SIP's parsing so that if a port is specified - in a string to Dial(), it is not ignored. (closes issue #13355) - Reported by: acunningham Patches: 13355v2.patch uploaded by - putnopvut (license 60) Tested by: acunningham ........ - -2008-08-27 23:23 +0000 [r140355] Tilghman Lesher <tlesher@digium.com> - - * cdr/cdr_pgsql.c: Oops - -2008-08-27 20:11 +0000 [r140301] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: 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 21:59 +0000 [r140246] Doug Bailey <dbailey@digium.com> - - * channels/chan_dahdi.c: Move the mwi send thread functionality - back into the do_monitor thread so that it is easier to manage - CID spill resources when do_monitor needs to be killed. (closes - issue #13213) Reported by: bbryant - -2008-08-26 18:48 +0000 [r140205] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_dahdi.c, /: Merged revisions 140056 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r140056 | jpeeler | 2008-08-26 10:57:02 -0500 (Tue, 26 - Aug 2008) | 9 lines (closes issue #12071) Reported by: tzafrir - Patches: dahdi_close.diff uploaded by tzafrir (license 46) Tested - by: tzafrir, jpeeler This patch fixes closing open file - descriptors in the case of an error. ........ - -2008-08-26 18:46 +0000 [r140201] Tilghman Lesher <tlesher@digium.com> - - * apps/app_followme.c: OpenBSD compat fix (reminded by mvanbaak on - #asterisk-dev) - -2008-08-26 18:11 +0000 [r140169] Russell Bryant <russell@digium.com> - - * Makefile: 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. - -2008-08-26 18:05 +0000 [r140167] Tilghman Lesher <tlesher@digium.com> - - * configs/followme.conf.sample, apps/app_followme.c: Standardize - the option names for consistency (but continue to work with the - existing names for backwards compatibility). (closes issue - #13370) Reported by: jsturtevant - -2008-08-26 16:10 +0000 [r140061] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 140060 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r140060 | russell | 2008-08-26 11:07:58 -0500 (Tue, 26 Aug 2008) - | 6 lines Fix some bogus scheduler usage in chan_sip. This code - used the return value of a completely unrelated function to - determine whether the scheduler should be run or not. This would - have caused the scheduler to not run in cases where it should - have. Also, leave a note about another scheduler issue that needs - to be addressed at some point. ........ - -2008-08-26 15:57 +0000 [r140057] Steve Murphy <murf@digium.com> - - * main/cdr.c, configs/cdr.conf.sample, CHANGES, - include/asterisk/options.h: (closes issue #13366) Reported by: - erousseau This was a reasonable enhancement request, which was - easy to implement. Since it's an enhancement, it could only be - applied to trunk. Basically, for accounting where "initiated" - seconds are billed for, if the microseconds field on the end time - is greater than the microseconds field for the answer time, add - one second to the billsec field. The implementation was requested - by erousseau, and I've implemented it as requested. I've updated - the CHANGES, the cdr.conf.sample, and the .h files accordingly, - to accept and set a flag for the corresponding new option. cdr.c - adds in the extra second based on the usec fields if the option - is set. Tested, seems to be working fine. - -2008-08-26 15:29 +0000 [r140053] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 140051 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r140051 | russell | 2008-08-26 10:27:23 -0500 (Tue, 26 - Aug 2008) | 15 lines Fix a race condition with the IAX scheduler - thread. A lock and condition are used here to allow newly - scheduled tasks to wake up the scheduler just in case the new - task needs to run sooner than the current wakeup time when the - thread is sleeping. However, there was a race condition such that - a newly scheduled task would not properly wake up the scheduler - or affect the wake up period. The order of execution would have - been: 1) Scheduler thread determines wake up time of N ms. 2) - Another thread schedules a task and signals the condition, with - an execution time of < N ms. 3) Scheduler thread locks and goes - to sleep for N ms. By moving the sleep time determination to - inside the critical section, this possibility is avoided. - ........ - -2008-08-25 23:13 +0000 [r139981] Tilghman Lesher <tlesher@digium.com> - - * Makefile, doc/asterisk.8, include/asterisk/options.h, - main/asterisk.c, main/term.c: Optional light colored background, - for those who use black on white terminals. (closes issue #13306) - Reported by: Corydon76 Patches: 20080814__bug13306__3.diff.txt - uploaded by Corydon76 (license 14) Tested by: Corydon76, pkempgen - -2008-08-25 21:48 +0000 [r139928] Jeff Peeler <jpeeler@digium.com> - - * main/manager.c, /: Merged revisions 139927 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r139927 | jpeeler | 2008-08-25 16:47:33 -0500 (Mon, 25 Aug 2008) - | 3 lines Fix a typo I made. Lesson learned, apply the patch if - one exists. ........ - -2008-08-25 21:32 +0000 [r139915] Sean Bright <sean.bright@gmail.com> - - * build_tools/get_moduleinfo, /, build_tools/get_makeopts: 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 20:59 +0000 [r139870] Terry Wilson <twilson@digium.com> - - * /, channels/chan_sip.c: 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 17:24 +0000 [r139832] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Add output of variables to AgentRingNoAnswer - manager event if eventwhencalled is set to "vars" in queues.conf. - Yay for consistency. (closes issue #13369) Reported by: srt - Patches: 13369_agentringnoanswer_variables.diff uploaded by srt - (license 378) - -2008-08-25 16:02 +0000 [r139775] Tilghman Lesher <tlesher@digium.com> - - * doc/followme.txt (added), apps/app_followme.c: Realtime - capabilities for the Find-Me-Follow-Me application. (closes issue - #13295) Reported by: Corydon76 Patches: - 20080813__followme_realtime_enabled.diff.txt uploaded by - Corydon76 (license 14) Tested by: dferrer - -2008-08-25 15:54 +0000 [r139770] Steve Murphy <murf@digium.com> - - * main/pbx.c, /, main/features.c: 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 16:26 +0000 [r139704-139707] Tilghman Lesher <tlesher@digium.com> - - * cdr/cdr_pgsql.c: Memory leak - - * cdr/cdr_pgsql.c: Eliminate open coding of ast_str - -2008-08-22 22:32 +0000 [r139627-139662] Steve Murphy <murf@digium.com> - - * /, main/features.c: 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 - 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 21:57 +0000 [r139622-139624] Jeff Peeler <jpeeler@digium.com> - - * main/manager.c, /: Merged revisions 139621 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r139621 | jpeeler | 2008-08-22 16:36:13 -0500 (Fri, 22 Aug 2008) - | 5 lines (closes issue #13359) Reported by: Laureano Patches: - originate_channel_check.patch uploaded by Laureano (license 265) - ........ - - * main/features.c: remove extra comma typo - -2008-08-22 20:20 +0000 [r139457-139563] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: The -1 return value from incomplete or - improper headers for the SipNotify manager command was causing - the current manager session to become disconnected. Change the - return value to 0 for these cases. Also change a test for a NULL - pointer to be ast_strlen_zero instead. (closes issue #13351) - Reported by: Laureano Patches: sipnotify_action_fix.patch - uploaded by Laureano (license 265) - - * main/features.c: Add missing unique id to ParkedCallGiveUp and - ParkedCallTimeOut manager events (closes issue #13358) Reported - by: srt Patches: 13358_parking_events.diff uploaded by srt - (license 378) - - * /, include/asterisk/threadstorage.h: 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 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 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 23:41 +0000 [r139391] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_dahdi.c, /: 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 23:00 +0000 [r139345-139346] Dwayne M. Hubbard <dwayne.hubbard@gmail.com> - - * apps/app_receivefax.c (removed), apps/app_sendfax.c (removed): - oops - - * apps/app_receivefax.c (added), apps/app_sendfax.c (added): - initiate T38 negotiation in FaxSend; use channel variables; other - stuff too - -2008-08-21 09:55 +0000 [r139281] Philippe Sultan <philippe.sultan@gmail.com> - - * channels/chan_gtalk.c: 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 22:16 +0000 [r139215] Russell Bryant <russell@digium.com> - - * /, apps/app_chanspy.c: 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 22:06 +0000 [r139210] Jason Parker <jparker@digium.com> - - * channels/chan_sip.c: Fix output of sipshowpeer manager response. - (closes issue #13346) Reported by: srt Patches: - 13346_malformed_sip_show_peer_response.diff uploaded by srt - (license 378) - -2008-08-20 20:03 +0000 [r139153-139154] Shaun Ruffell <sruffell@digium.com> - - * codecs/codec_dahdi.c: Remove extraneous debugging messages. - - * 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 17:25 +0000 [r139083] Steve Murphy <murf@digium.com> - - * main/cdr.c, /: 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 16:16 +0000 [r139020] Michiel van Baak <michiel@vanbaak.info> - - * channels/chan_skinny.c: fix unholding phones after hangup on - older cisco phones. Patch by wedhorn. - -2008-08-20 15:38 +0000 [r138887-139016] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_sip.c: 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 - ........ - - * /, channels/chan_agent.c: Merged revisions 138942 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.4 - ........ r138942 | mmichelson | 2008-08-19 18:17:17 -0500 (Tue, - 19 Aug 2008) | 11 lines Reset agent_pvt variables back to the - values in agents.conf (from what the corresponding channel - variables were set to) when the agent logs out. (closes issue - #13098) Reported by: davidw Patches: - 20080731__issue13098_agent_ackcall_not_reset.diff uploaded by - bbryant (license 36) Tested by: davidw ........ - - * /, apps/app_chanspy.c: Merged revisions 138886 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ - r138886 | mmichelson | 2008-08-19 13:50:53 -0500 (Tue, 19 Aug - 2008) | 23 lines Add a lock and unlock prior to the destruction - of the chanspy_ds lock to ensure that no other threads still have - it locked. While this should not happen under normal - circumstances, it appears that if the spyer and spyee hang up at - nearly the same time, the following may occur. 1. - ast_channel_free is called on the spyee's channel. 2. The chanspy - datastore is removed from the spyee's channel in - ast_channel_free. 3. In the spyer's thread, the spyer attempts to - remove and destroy the datastore from the spyee channel, but the - datastore has already been removed in step 2, so the spyer - continues in the code. 4. The spyee's thread continues and calls - the datastore's destroy callback, chanspy_ds_destroy. This - involves locking the chanspy_ds. 5. Now the spyer attempts to - destroy the chanspy_ds lock. The problem is that in step 4, the - spyee has locked this lock, meaning that the spyer is attempting - to destroy a lock which is currently locked by another thread. - The backtrace provided in issue #12969 supports the idea that - this is possible (and has even occurred). This commit does not - close the issue, but should help in preventing one type of crash - associated with the use of app_chanspy. ........ - -2008-08-19 16:56 +0000 [r138851] Michiel van Baak <michiel@vanbaak.info> - - * channels/chan_skinny.c: chan_skinny now respects callwaiting=no - (closes issue #12691) Reported by: sbisker Patches: - callwaitingv1.diff uploaded by wedhorn (license 30) Tested by: - wedhorn on old skinny phones, mvanbaak on 7960 and 7905 with - latest firmware - -2008-08-19 16:31 +0000 [r138815-138845] Steve Murphy <murf@digium.com> - - * res/ael/ael.tab.c, res/ael/ael.y, res/ael/ael.tab.h, - utils/ael_main.c, utils/conf2ael.c: 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: - 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 15:58 +0000 [r138814] Philippe Sultan <philippe.sultan@gmail.com> - - * res/res_jabber.c: Mention JID rather than SreenName in help - messages - -2008-08-19 00:10 +0000 [r138775-138780] Sean Bright <sean.bright@gmail.com> - - * channels/chan_sip.c: Let it compile now, too (woops) - - * channels/chan_sip.c: And remove code we don't need anymore. - - * channels/chan_sip.c: While we're at it, make this machine - parseable too. - - * channels/chan_sip.c: Change event header to RegistrationTime to - be more consistent (and avoid breaking existing frameworks). - Pointed out by Laureano on #asterisk-dev. - -2008-08-18 21:07 +0000 [r138738] Richard Mudgett <rmudgett@digium.com> - - * channels/misdn/isdn_lib_intern.h, channels/misdn/isdn_lib.h, - doc/tex/misdn.tex, channels/chan_misdn.c, - configs/misdn.conf.sample, channels/misdn/isdn_lib.c, - channels/misdn_config.c: 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/tex/misdn.tex * Updated the - misdn_set_opt application option descriptions. * Fixed some - spelling errors and typos. - -2008-08-18 20:23 +0000 [r138687-138694] Mark Michelson <mmichelson@digium.com> - - * configs/queues.conf.sample, apps/app_queue.c: Change the queue - timeout priority logic into less ugly and confusing code pieces. - Clarify the logic within queues.conf.sample. (closes issue - #12690) Reported by: atis Patches: queue_timeoutpriority.patch - uploaded by atis (license 242) - - * apps/app_queue.c: 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 [r138631] Jason Parker <jparker@digium.com> - - * Makefile: Remove option that isn't valid here. - -2008-08-18 02:13 +0000 [r138518] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_dahdi.c: add missing define for SS7 in - dahdi_restart - -2008-08-17 14:12 +0000 [r138442-138482] Sean Bright <sean.bright@gmail.com> - - * main/features.c: 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: 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: 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: 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: 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 [r138409-138412] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_dahdi.c: Fix compilation warnings (found with - dev-mode) - - * main/pbx.c: Also make sure hinting won't crash on reload. (Closes - issue #13312) - -2008-08-16 01:13 +0000 [r138311-138361] Jeff Peeler <jpeeler@digium.com> - - * channels/chan_dahdi.c, /: 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 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 22:54 +0000 [r138206-138260] Tilghman Lesher <tlesher@digium.com> - - * /, channels/chan_sip.c, configs/sip.conf.sample: 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 ........ - - * main/pbx.c, configs/extensions.conf.sample: Remove deprecated - syntax from sample config file (closes issue #13314) Reported by: - kue - -2008-08-15 20:12 +0000 [r138155] Jeff Peeler <jpeeler@digium.com> - - * 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 19:36 +0000 [r138086-138148] Tilghman Lesher <tlesher@digium.com> - - * main/pbx.c: Change free to ast_free_ptr, too - - * main/pbx.c: e->data can be NULL, so use the safe version of - ast_strdup() (closes issue #13312) Reported by: pj - - * channels/chan_sip.c: regseconds is actually stored as the epoch - time, not registration length - -2008-08-15 15:09 +0000 [r138028] Russell Bryant <russell@digium.com> - - * main/autoservice.c, /: 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 15:03 +0000 [r138024] Tilghman Lesher <tlesher@digium.com> - - * /, funcs/func_strings.c: 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 [r137987] Russell Bryant <russell@digium.com> - - * doc/tex/Makefile: Fix a bashism that causes an error when trying - to build the pdf on ubuntu - -2008-08-14 18:47 +0000 [r137933] Sean Bright <sean.bright@gmail.com> - - * cdr/cdr_sqlite3_custom.c: 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 18:12 +0000 [r137901] Russell Bryant <russell@digium.com> - - * CHANGES: Prepare for adding 1.6.2 changes - -2008-08-14 16:52 +0000 [r137848] Tilghman Lesher <tlesher@digium.com> - - * channels/chan_dahdi.c, /: 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 15:32 +0000 [r137812] Jason Parker <jparker@digium.com> - - * channels/chan_sip.c: Make sure we set the socket port, so we - don't try to use <ip address>:0. (closes issue #13255) Reported - by: falves11 Patches: 13255-socketport.diff uploaded by qwell - (license 4) Tested by: falves11 - -2008-08-14 15:03 +0000 [r137780] Sean Bright <sean.bright@gmail.com> - - * cdr/cdr_tds.c: 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 14:15 +0000 [r137732] Russell Bryant <russell@digium.com> - - * /, configs/sip.conf.sample: 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 02:03 +0000 [r137680] Kevin P. Fleming <kpfleming@digium.com> - - * /, Zaptel-to-DAHDI.txt: 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 ........ - |