diff options
author | russell <russell@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-08-07 21:00:45 +0000 |
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committer | russell <russell@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-08-07 21:00:45 +0000 |
commit | f1404e8f61122d615d804155737d1dba7bbfd98c (patch) | |
tree | 25e1f3259147e1bbdb730f43608c15ce33764483 | |
parent | da9a195ab95a4fb3580fdae026a694e912ec7070 (diff) | |
parent | 65ecf0b84d40c10434155ba1d614646d3ce6da40 (diff) |
Creating tag for the release of asterisk-1.4.10
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.10@78490 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r-- | .lastclean | 1 | ||||
-rw-r--r-- | .version | 1 | ||||
-rw-r--r-- | ChangeLog | 10427 | ||||
-rw-r--r-- | res/res_config_odbc.c | 20 |
4 files changed, 12 insertions, 10437 deletions
diff --git a/.lastclean b/.lastclean deleted file mode 100644 index 9902f1784..000000000 --- a/.lastclean +++ /dev/null @@ -1 +0,0 @@ -28 diff --git a/.version b/.version deleted file mode 100644 index ac9f79cab..000000000 --- a/.version +++ /dev/null @@ -1 +0,0 @@ -1.4.10 diff --git a/ChangeLog b/ChangeLog deleted file mode 100644 index 7f90c48d9..000000000 --- a/ChangeLog +++ /dev/null @@ -1,10427 +0,0 @@ -2007-08-07 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.10 released. - -2007-08-07 19:43 +0000 [r78450] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: The logic behind inboxcount's return value - was reversed in has_voicemail and message_count. (closes issue - #10401, reported by st1710, patched by me) - -2007-08-07 19:34 +0000 [r78437] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * res/res_odbc.c: Don't free the environment handle when the - connection fails, because other connections might be depending - upon it - -2007-08-07 19:11 +0000 [r78416] Jason Parker <jparker@digium.com> - - * channels/chan_sip.c: Allow chan_sip to build in devmode - -2007-08-07 19:09 +0000 [r78415] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c, res/res_config_odbc.c, - apps/app_directory.c: Reconnection doesn't happen automatically - when a DB goes down (fixes issue #9389) - -2007-08-07 18:25 +0000 [r78375] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Properly check the capabilities count to - avoid a segfault. (ASA-2007-019) - -2007-08-07 17:45 +0000 [r78371] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c, /: Merged revisions 78370 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r78370 | russell | 2007-08-07 12:44:04 -0500 (Tue, 07 Aug 2007) | - 4 lines Revert patch committed for issue #9660. It broke E&M - trunks. (closes issue #10360) (closes issue #10364) ........ - -2007-08-06 21:41 +0000 [r78275] Joshua Colp <jcolp@digium.com> - - * main/channel.c: Add additional DTMF log messages to help when - debugging issues. - -2007-08-06 20:44 +0000 [r78184-78242] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Fix an issue where dynamic threads can get - free'd, but still exist in the dynamic thread list. (closes issue - #10392, patch from Mihai, with credit to his colleague, Pete) - - * include/asterisk/linkedlists.h: Fix the return value of - AST_LIST_REMOVE(). This shouldn't be causing any problems, - though, because the only code that uses the return value only - checks to see if it is NULL. (closes issue #10390, pointed out by - mihai) - -2007-08-06 16:32 +0000 [r78182] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: It is possible for a transfer to occur - before the remote device has our tag in which case they send none - in the transfer. In this case we need to not fail the transfer - dialog lookup. - -2007-08-06 16:30 +0000 [r78180] Jason Parker <jparker@digium.com> - - * main/config.c: Fix an issue with using UpdateConfig (manager - action) where escaped semicolons in a config would be converted - to just semicolons (\; to ;) Issue 9938 - -2007-08-06 15:27 +0000 [r78166-78172] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: (closes issue #10355) Reported by: wdecarne Now that - we pass through RTP timestamp information we need to make the - allowed timestamp skew considerably less. There are situations - where a source may change and due to the timestamp difference the - receiver will experience an audio gap since we did not indicate - by setting the marker bit that the source changed. - - * configure, configure.ac: (closes issue #10383) Reported by: rizzo - Include stdlib.h so NULL gets defined for gethostbyname_r checks. - -2007-08-06 13:33 +0000 [r78164] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: Fixed a mistake I made in realtime_peer - which caused it to return NULL every time. Thanks to Jon Fealy - for emailing me the correction. - -2007-08-05 14:18 +0000 [r78146] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * cdr/cdr_pgsql.c: Portability fix for devmode compiling (closes - bug #10382) - -2007-08-05 04:15 +0000 [r78143] Russell Bryant <russell@digium.com> - - * include/asterisk/lock.h: Fix compilation failure when - MALLOC_DEBUG is enabled, but DEBUG_THREADS is not - -2007-08-05 03:29 +0000 [r78139] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * channels/chan_sip.c: If peer is not found, the error message is - misleading (should be peer not found, not ACL failure) - -2007-08-03 20:25 +0000 [r78103] Mark Michelson <mmichelson@digium.com> - - * main/config.c, channels/chan_sip.c, include/asterisk/config.h: - Changed the behavior of sip's realtime_peer function to match the - corresponding way of matching for non-realtime peers. Now matches - are made on both the IP address and port number, or if the - insecure setting is set to "port" then just match on the IP - address. In order to accomplish this, I also added a new API - call, ast_category_root, which returns the first variable of an - ast_category struct - -2007-08-03 20:14 +0000 [r78028-78101] Russell Bryant <russell@digium.com> - - * apps/app_voicemail.c: (closes issue #10194) Reported by: - blitzrage Patches: bug0010194 uploaded by vovochka Tested by: - blitzrage Fix a problem when you call Voicemail() with multiple - mailboxes specified and ODBC_STORAGE is in use. The audio part of - the message was only given to the first mailbox specified. - - * main/utils.c, include/asterisk/lock.h, main/astmm.c: Add some - improvements to lock debugging. These changes take effect with - DEBUG_THREADS enabled and provide the following: * This will keep - track of which locks are held by which thread as well as which - lock a thread is waiting for in a thread-local data structure. A - reference to this structure is available on the stack in the - dummy_start() function, which is the common entry point for all - threads. This information can be easily retrieved using gdb if - you switch to the dummy_start() stack frame of any thread and - print the contents of the lock_info variable. * All of the - thread-local structures for keeping track of this lock - information are also stored in a list so that the information can - be dumped to the CLI using the "core show locks" CLI command. - This introduces a little bit of a performance hit as it requires - additional underlying locking operations inside of every - lock/unlock on an ast_mutex. However, the benefits of having this - information available at the CLI is huge, especially considering - this is only done in DEBUG_THREADS mode. It means that in most - cases where we debug deadlocks, we no longer have to request - access to the machine to analyze the contents of ast_mutex_t - structures. We can now just ask them to get the output of "core - show locks", which gives us all of the information we needed in - most cases. I also had to make some additional changes to astmm.c - to make this work when both MALLOC_DEBUG and DEBUG_THREADS are - enabled. I disabled tracking of one of the locks in astmm.c - because it gets used inside the replacement memory allocation - routines, and the lock tracking code allocates memory. This - caused infinite recursion. - - * channels/chan_iax2.c: Only pass through HOLD and UNHOLD control - frames when the mohinterpret option is set to "passthrough". This - was pointed out by Kevin in the middle of a training session. - - * channels/chan_iax2.c: Don't reuse the timespec that was set to 0 - in the previous timedwait as it will just return immediately. - Also, fix some logic so the thread's lock isn't unlocked twice in - the weird case of dynamic threads getting acquired right after a - timeout. (pointed out by SteveK) - -2007-08-02 21:53 +0000 [r77993-77996] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c, configs/skinny.conf.sample: Make sure we - actually allow 6 chars to be sent. Also make note of the "A" - option of date format. Issue 9779, modifications by DEA, wedhorn, - and myself. - - * channels/chan_skinny.c: If a device disconnects, the session will - go away. If this happens during call setup, we need to give up. - Issue 10325. - -2007-08-02 19:25 +0000 [r77949] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Fix the case where a dynamic thread times - out waiting for something to do during the first time it runs. - This shouldn't ever happen, but we should account for it anyway. - (pointed out by pete, who works with mihai) - -2007-08-02 18:42 +0000 [r77947] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Make sure we clear the prompt status - message on a hangup. Also rearrange messages to better fit with - what a wireshark trace shows it should be. Issue 10299, initial - patch and solution by sbisker, modified by me to fit with - wireshark trace. - -2007-08-02 18:21 +0000 [r77945] Steve Murphy <murf@digium.com> - - * main/fskmodem.c, /: Merged revisions 77942 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r77942 | murf | 2007-08-02 11:56:37 -0600 (Thu, 02 Aug 2007) | 1 - line This patch hopefully solves 10141; The user is running with - it, and it doesn't appear to harm asterisk's operation, and may - prevent a crash. I'll store it in 1.2, as we have shut down - support on 1.2, but since I developed the patch before support - finished, and it might affect 1.4 and trunk, I'm going ahead with - it. ........ - -2007-08-02 18:04 +0000 [r77939-77943] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Fix another race condition in the handling - of dynamic threads. If the dynamic thread timed out waiting for - something to do, but was acquired to perform an action - immediately afterwords, then wait on the condition again to give - the other thread a chance to finish setting up the data for what - action this thread should perform. Otherwise, if it immediately - continues, it will perform the wrong action. (reported on IRC by - mihai, patch by me) (related to issue #10289) - - * channels/chan_iax2.c: Add another sanity check to - vnak_retransmit(). This check ensures that frames that have - already been marked for deletion don't get retransmitted. (closes - issue #10361, patch from mihai) - -2007-08-02 15:15 +0000 [r77890-77894] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Make sure that we show the correct - extension if dialed from a macro "From: 5555" rather than "From: - s" Issue 10358, initial patch by DEA, reworked by me to use S_OR, - tested by sbisker - - * channels/chan_skinny.c: Put in some additional debug information - for softkey/stimulus messages. Issue 10291, patch by DEA. - -2007-08-01 22:16 +0000 [r77887] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Fix some race conditions which have been - causing weird problems in chan_iax2. The most notable problem is - that people have been seeing storms of VNAK frames being sent due - to really old frames mysteriously being in the retransmission - queue and never getting removed. It was possible that a dynamic - thread got created, but did not acquire its lock before the - thread that created it signals it to perform an action. When this - happens, the thread will sleep until it hits a timeout, and then - get destroyed. So, the action never gets performed and in some - cases, means a frame doesn't get transmitted and never gets freed - since the scheduler never gets a chance to reschedule - transmission. Another less severe race condition is in the - handling of a timeout for a dynamic thread. It was possible for - it to be acquired to perform at action at the same time that it - hit a timeout. When this occurs, whatever action it was acquired - for would never get performed. (patch contributed by Mihai and - SteveK) (closes issue #10289) (closes issue #10248) (closes issue - #10232) (possibly related to issue #10359) - -2007-08-01 22:14 +0000 [r77886] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Voicemail with ODBC_STORAGE defined does - not compile cleanly (missing def) - -2007-08-01 21:08 +0000 [r77883] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Fix an issue that caused one-way audio on - some newer devices (specifically the 7921), due to sending - packets in the wrong order during hangup. Also make sure we clear - tones/messages on the correct line/instance. Issue 10291, patch - by DEA, tested by sbisker and myself. - -2007-08-01 18:08 +0000 [r77863-77871] Joshua Colp <jcolp@digium.com> - - * main/cli.c: (closes issue #10351) Reported by: ftarz Some - platforms don't like it when you pass NULL to vsnprintf so pass - "" instead. - - * include/asterisk/threadstorage.h, channels/chan_mgcp.c, - apps/app_voicemail.c, main/acl.c, utils/smsq.c, - channels/chan_iax2.c: Add some fixes for building on Solaris. - - * main/utils.c: Whoops, I meant R_5 not R5. - - * configure, configure.ac: And for my last trick... make sure that - if gethostbyname_r is exported by a library that it is used. - - * configure, include/asterisk/autoconfig.h.in, configure.ac, - main/utils.c: Extend autoconf logic to determine which version of - gethostbyname_r is on the system. - -2007-08-01 14:08 +0000 [r77852-77854] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Fixes an issue I introduced to queues wherein a - queue with joinempty=yes would kick people out of the queue - because of erroneously thinking the 'n' option was in use. - (closes issue #10320, reported by jfitzgibbon, patched by me, - tested by blitzrage and me) Thank you blitzrage for all the - testing you've done lately with queues! It's much appreciated! - - * apps/app_queue.c: If a queue uses dynamic realtime members, then - the member list should be updated after each attempt to call the - queue. This fixes an issue where if a caller calls into a queue - where no one is logged in, they would wait forever even if a - member logged in at some point. (closes issue #10346, reported by - and tested by blitzrage, patched by me) - -2007-07-31 21:09 +0000 [r77845-77846] Jim Dixon <telesistant@hotmail.com> - - * apps/app_rpt.c: Much newer version, 0.70 with much additions - - * main/dsp.c, channels/chan_zap.c: Made VAST improvements in DTMF - receiver in RADIO_RELAX mode (thanx Steve W9SH), and oversight in - logic in TONE_VERIFY/RELAX mode in chan_zap. - -2007-07-31 20:59 +0000 [r77844] Steve Murphy <murf@digium.com> - - * /, contrib/scripts/ast_grab_core: Merged revisions 77842 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r77842 | murf | 2007-07-31 13:19:35 -0600 (Tue, 31 Jul 2007) | 1 - line This probably isn't super-general, but it's a first stab at - using kill -11 to generate a core file instead of gcore. ........ - -2007-07-31 16:17 +0000 [r77831] Joshua Colp <jcolp@digium.com> - - * res/res_speech.c, include/asterisk/speech.h: Add a flag to the - speech API that allows an engine to set whether it received - results or not. - -2007-07-31 15:53 +0000 [r77827] Kevin P. Fleming <kpfleming@digium.com> - - * build_tools/cflags.xml: DETECT_DEADLOCKS can't be enabled without - DEBUG_THREADS or it does nothing - -2007-07-31 15:21 +0000 [r77824] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: This patch makes Asterisk send 100 Trying - provisional responses upon receipt of re-invites. This makes it - so that if there are two or more Asterisk servers between - endpoints, the Asterisk servers will not keep retransmitting the - re-invites. (closes issue #10274, reported by cstadlmann, patched - by me with approval from file) - -2007-07-30 20:17 +0000 [r77795] Jason Parker <jparker@digium.com> - - * main/say.c: Applications like SayAlpha() should not hang up the - channel if you request an "unknown" character such as a comma. - Instead, skip the character and move on. Issue 10083, initial - patch by jsmith, modified by me. - -2007-07-30 20:16 +0000 [r77785-77794] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Fix an issue that could potentially cause - corruption of the global iax frame queue. In the network_thread() - loop, it traverses the list using the AST_LIST_TRAVERSE_SAFE - macro. However, to remove an element of the list within this - loop, it used AST_LIST_REMOVE, instead of - AST_LIST_REMOVE_CURRENT, which I believe could leave some of the - internal variables of the SAFE macro invalid. Mihai says that he - already made this change in his local copy and it didn't help his - VNAK storm issues, but I still think it's wrong. :) - - * res/res_agi.c: (closes issue #10279) Reported by: seanbright - Patches: res_agi.carefulwrite.1.4.07252007.patch uploaded by - seanbright (license 71) res_agi.carefulwrite.trunk.07252007.patch - uploaded by seanbright (license 71) Allow the "agi_network: yes" - line to be printed out in the AGI debug output. Also, allow - partial writes to be handled when writing out this line just like - it is for all of the others. - - * main/channel.c: file and I both committed changes for issue - #10301. Remove a duplicated assignment to restore the original - value of the previous channel. - -2007-07-30 18:43 +0000 [r77783] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, res/res_agi.c: Merged revisions 77782 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r77782 | tilghman | 2007-07-30 13:40:54 -0500 (Mon, 30 Jul 2007) - | 2 lines Revert change in revision 71656, even though it fixed a - bug, because many people were depending upon the (broken) - behavior. ........ - -2007-07-30 17:29 +0000 [r77780] Russell Bryant <russell@digium.com> - - * main/channel.c: (closes issue #10301) Reported by: fnordian - Patches: asterisk-1.4.9-channel.c.patch uploaded by fnordian - (license 110) Additional changes by me Fix some problems in - channel_find_locked() which can cause an infinite loop. The - reference to the previous channel is set to NULL in some cases. - These changes ensure that the reference to the previous channel - gets restored before needing it again. I'm not convinced that the - code that is setting it to NULL is really the right thing to do. - However, I am making these changes to fix the obvious problem and - just leaving an XXX comment that it needs a better explanation - that what is there now. - -2007-07-30 17:11 +0000 [r77768-77778] Joshua Colp <jcolp@digium.com> - - * res/res_features.c: (closes issue #10327) Reported by: kkiely - Instead of directly mucking with the extension/context/priority - of the channel we are transferring when it has a PBX simply call - ast_async_goto on it. This will ensure that the channel gets - handled properly and sent to the right place. - - * main/channel.c: (closes issue #10301) Reported by: fnordian - Patches: asterisk-1.4.9-channel.c.patch uploaded by fnordian - (license 110) Restore previous behavior where if we failed to - lock the channel we wanted we would return to exactly the same - point as if we had just reentered the function. - - * /, apps/app_macro.c: Merged revisions 77767 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r77767 | file | 2007-07-30 11:50:02 -0300 (Mon, 30 Jul 2007) | 4 - lines (closes issue #10334) Reported by: ramonpeek Pass through - the return value from macro_exec through the MacroIf application. - ........ - -2007-07-27 18:15 +0000 [r77571] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * res/res_odbc.c: Missing newline - -2007-07-27 17:04 +0000 [r77536-77540] Joshua Colp <jcolp@digium.com> - - * cdr/cdr_pgsql.c: (closes issue #10310) Reported by: prashant_jois - Patches: cdr_pgsql.patch uploaded by prashant (license 114) - Finish the Postgresql connection after the log messages are - printed so we don't access invalid memory. - - * channels/chan_sip.c: (closes issue #10323) Reported by: julianjm - Patches: chan_sip_device_state_hold_fix.v1.diff.txt uploaded by - julianjm (license 99) Clear ONHOLD flag when decrementing the - onHold peer count. If we did not do this the count may keep - decreasing. - -2007-07-27 14:30 +0000 [r77490] Mark Michelson <mmichelson@digium.com> - - * channels/chan_sip.c: "re-invite" was misspelled - -2007-07-26 23:19 +0000 [r77460] Joshua Colp <jcolp@digium.com> - - * main/channel.c: (closes issue #10302) Reported by: litnialex If a - DTMF end frame comes from a channel without a begin and it is - going to a technology that only accepts end frames (aka INFO) - then use the minimum DTMF duration if one is not in the frame - already. - -2007-07-26 22:16 +0000 [r77424-77429] Kevin P. Fleming <kpfleming@digium.com> - - * doc/mp3.txt: change protocol for downloads as well - - * doc/mp3.txt, sounds/Makefile: use new canonical name for download - server - -2007-07-26 21:23 +0000 [r77410] Russell Bryant <russell@digium.com> - - * Makefile, build_tools/make_buildopts_h: AST_DEVMODE was defined - in trunk, but not in 1.4. When Asterisk is compiled under dev - mode, AST_DEVMODE will get defined in buildopts.h. Change 1.4 to - define it in the same way that trunk does. Also, revert the - change that added this define in the Makefile The advantage to - doing it this way is that buildopts.h gets installed when you - install Asterisk. Then, when building any out of tree modules, or - building asterisk-addons, these modules know which options the - rest of Asterisk was built with. - -2007-07-26 20:35 +0000 [r77380] Mark Michelson <mmichelson@digium.com> - - * Makefile, main/logger.c: Fixes to get ast_backtrace working - properly. The AST_DEVMODE macro was never defined so the majority - of ast_backtrace never attempted compilation. The makefile now - defines AST_DEVMODE if configure was run with --enable-dev-mode. - Also, changes were made to acccomodate 64 bit systems in - ast_backtrace. Thanks to qwell, kpfleming, and Corydon76 for - their roles in allowing me to get this committed - -2007-07-26 19:32 +0000 [r77348-77350] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * main/logger.c: Missed one - - * main/logger.c: Oops, that builtin define should be all-lowercase. - -2007-07-26 18:30 +0000 [r77318] Mark Michelson <mmichelson@digium.com> - - * cdr/cdr_pgsql.c: Two consecutive calls to PQfinish could occur, - meaning free gets called on the same variable twice. This patch - sets the connection to NULL after calls to PQfinish so that the - problem does not occur. Also in this patch, prashant_jois - informed me that it is safe to pass a null pointer to PQfinish, - so I have removed the check for conn's existence from - my_unload_module. (closes issue 10295, reported by junky, patched - by me with input from prashant_jois) - -2007-07-25 22:39 +0000 [r77191] Steve Murphy <murf@digium.com> - - * apps/app_meetme.c: This fix solves problem with intense squelch - noise when someone joins conf in bug 9430; We repro'd the problem - with meetme opts of 'CciMo'; Josh Colp supplied this patch, and - I'm applying it. It looks like playing the recorded username will - louse up the next thing played into the channel. Josh rearranged - the code so as to start things over before playing data directly - into the conference. - -2007-07-25 22:16 +0000 [r77176] Joshua Colp <jcolp@digium.com> - - * apps/app_speech_utils.c: (closes issue #10303) Reported by: jtodd - Add SPEECH_DTMF_TERMINATOR variable so the user can specify the - digit to terminate a DTMF string with. If none is specified then - no terminator will be used. - -2007-07-25 21:52 +0000 [r77154] Mark Michelson <mmichelson@digium.com> - - * main/channel.c: chan->emulate_dtmf_duration is an unsigned int, - not a signed int, so use %u instead of %d in the format string - -2007-07-25 20:23 +0000 [r77116-77136] Jason Parker <jparker@digium.com> - - * /: so are my fingers... - - * /: autotagexternals script is still obviously misbehaving... - - * /: use autotagged externals - -2007-07-25 17:14 +0000 [r77071] Joshua Colp <jcolp@digium.com> - - * configure, acinclude.m4: Fix autoconf logic for finding OpenH323 - when it is not in the first place searched (/usr/share/openh323). - -2007-07-25 09:34 +0000 [r77022] Luigi Rizzo <rizzo@icir.org> - - * main/rtp.c: set the sequence number in a frame for all frame - types - -2007-07-25 00:18 +0000 [r76983] Steve Murphy <murf@digium.com> - - * channels/chan_zap.c, /: Merged revisions 76978 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r76978 | murf | 2007-07-24 18:07:24 -0600 (Tue, 24 Jul 2007) | 1 - line this fixes bug 10293, where the error message because - defaultzone or loadzone was not defined was confusing ........ - -2007-07-24 22:12 +0000 [r76891-76937] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, include/asterisk/lock.h: Merged revisions 76934 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r76934 | tilghman | 2007-07-24 17:11:33 -0500 (Tue, 24 - Jul 2007) | 2 lines Oops, res contains the error code, not errno. - I was wondering why a mutex was reporting "No such file or - directory"... ........ - - * main/app.c: Found another place where we should be using the - umask (thanks jcmoore) - -2007-07-24 Jason Parker <jparker@digium.com> - - * Asterisk 1.4.9 released. - -2007-07-24 16:42 +0000 [r76803-76805] Jason Parker <jparker@digium.com> - - * /: Blocked revisions 76802 via svnmerge ........ r76802 | qwell | - 2007-07-24 11:32:04 -0500 (Tue, 24 Jul 2007) | 3 lines Don't - create the Asterisk channel until we are starting the PBX on it. - (ASA-2007-018) ........ - - * channels/chan_iax2.c: Don't create the Asterisk channel until we - are starting the PBX on it. (ASA-2007-018) - -2007-07-24 16:26 +0000 [r76801] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Added a membercount variable to call_queue - struct which keeps track of the number of logged in members in a - particular queue. This makes it so that the 'n' option for - Queue() can act properly depending on which strategy is used. If - the strategy is roundrobin, rrmemory, or ringall, we want to ring - each phone once before moving on in the dialplan. However, if any - other strategy is used, we will only ring one phone since it - cannot be guaranteed that a different phone will ring on - subsequent attempts to ring a phone. As a side effect of this, - the QUEUE_MEMBER_COUNT dialplan function now just reads the - membercount variable instead of traversing through the member - list to figure out how many members there are. Special thanks to - blitzrage for helping to test this out. (closes issue #10127, - reported by bcnit, patched by me, tested by blitzrage) - -2007-07-23 22:38 +0000 [r76708] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: It was our stated intention for 1.4 that - files created in app_voicemail should depend upon the umask. - Unfortunately, mkstemp() creates files with mode 0600, regardless - of the umask. This corrects that deficiency. - -2007-07-23 18:59 +0000 [r76656] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Fix some incorrect softkey labels in - messages. Don't try to play dialtone in some unimplemented - features. - -2007-07-23 18:29 +0000 [r76654] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_agent.c: Merged revisions 76653 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r76653 | file | 2007-07-23 15:28:13 -0300 (Mon, 23 Jul - 2007) | 4 lines (closes issue #5866) Reported by: tyler Do not - force channel format changes when a generator is present. The - generator may have changed the formats itself and changing them - back would cause issues. ........ - -2007-07-23 17:57 +0000 [r76620] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Don't try to queue up hold/unhold frames - on a non-existent channel. Issue 10276. - -2007-07-23 17:48 +0000 [r76519-76618] Joshua Colp <jcolp@digium.com> - - * apps/app_morsecode.c: Allow app_morsecode to build on PPC Linux - by putting the value of the digit char in an int. - - * /, channels/chan_sip.c: Merged revisions 76560 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r76560 | file | 2007-07-23 11:32:07 -0300 (Mon, 23 Jul 2007) | 6 - lines (closes issue #10236) Reported by: homesick Patches: - rpid_1.4_75840.patch uploaded by homesick (license 91) Accept - Remote Party ID on guest calls. ........ - - * channels/chan_skinny.c: (closes issue #10268) Reported by: - mvanbaak Patches: chan_skinny_openbsd.diff uploaded by mvanbaak - (license 7) Add another OS that has to use the Macros for byte - ordering. - -2007-07-23 12:25 +0000 [r76485] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Use a signed integer for storing the number - of bytes in the packet read from the network. Using an unsigned - value here made it impossible to handle an error returned from - recvfrom(). Furthermore, in the case that recvfrom() did return - an error, this would cause a crash due to a heap overflow. - (closes issue #10265, reported by and fix suggested by - timrobbins) - -2007-07-22 21:42 +0000 [r76410] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /: Blocked revisions 76409 via svnmerge ........ r76409 | - tilghman | 2007-07-22 16:39:55 -0500 (Sun, 22 Jul 2007) | 2 lines - We should not use C++ reserved words in API headers (closes issue - #10266) ........ - -2007-07-21 02:02 +0000 [r76227] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 76226 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r76226 | russell | 2007-07-20 21:01:46 -0500 (Fri, 20 Jul 2007) | - 4 lines Backport a fix for a memory leak that was fixed in trunk - in reivision 76221 by rizzo. The memory used for the localaddr - list was not freed during a configuration reload. ........ - -2007-07-20 21:36 +0000 [r76211] Steve Murphy <murf@digium.com> - - * sounds/Makefile: This patch from 10249 is worth applying! It - prevents downloading sound files if they are already downloaded. - Darn Practical, if you ask me - -2007-07-20 21:03 +0000 [r76174-76178] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Allow getting a call from an existing - "sub" channel. Cancel ringing if endpoint hangs up before - answering. Fixes were backported from trunk (there was apparently - a bit of confusion during merge of a previous patch). (closes - issue #10241) - - * main/manager.c: Eliminate a compiler warning with gcc 4.2 by - constifying a char * - - * channels/chan_skinny.c: It's possible for sub->owner to be NULL - here if you cancel the call immediately after/during sending a - digit. - -2007-07-20 18:42 +0000 [r76139] Mark Michelson <mmichelson@digium.com> - - * apps/app_directory.c: When using users.conf for the entries in - the directory, if multiple users had the same last name, only the - first user listed would be available in the directory. (closes - issue #10200, reported by mrskippy, patched by me) - -2007-07-20 18:22 +0000 [r76132] Russell Bryant <russell@digium.com> - - * main/channel.c: Use the define that specifies the default length - of an artificially created DTMF digit in the ast_senddigit() - function. The define is set to 100ms by default, which is the - same thing that this function was using. But, using the define - lets changes take effect in this case, as well as the others - where it was already used. - -2007-07-20 17:20 +0000 [r76054-76087] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 76080 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r76080 | file | 2007-07-20 14:16:48 -0300 (Fri, 20 Jul 2007) | 6 - lines (closes issue #10247) Reported by: fkasumovic Patches: - chan_sip.patch uploaded by fkasumovic (license #101) Drop any - peer realm authentication entries when reloading so multiple - entries do not get added to the peer. ........ - - * res/res_convert.c: (closes issue #10246) Reported by: fkasumovic - Patches: res_conver.patch uploaded by fkasumovic (license #101) - Use the last occurance of . to find the extension, not the first - occurance. - - * apps/app_queue.c: Move makeannouncement variable declaration to - proper place. - -2007-07-19 20:36 +0000 [r75980] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Remove some duplicate code. - -2007-07-19 18:59 +0000 [r75969-75978] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: The diff on this looks pretty big but all I did - was remove a pointless if statement (always evaluates true). - - * apps/app_queue.c: Changes in handling return values of several - functions in app_queue. This all started as a fix for issue - #10008 but now includes all of the following changes: 1. - Simplifying the code to handle positive return values from ast - API calls. 2. Removing the background_file function. 3. The fix - for issue #10008 (closes issue #10008, reported and patched by - dimas) - -2007-07-19 15:53 +0000 [r75928] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 75927 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r75927 | russell | 2007-07-19 10:49:42 -0500 (Thu, 19 Jul 2007) | - 6 lines When processing full frames, take sequence number - wraparound into account when deciding whether or not we need to - request retransmissions by sending a VNAK. This code could cause - VNAKs to be sent erroneously in some cases, and to not be sent in - other cases when it should have been. (closes issue #10237, - reported and patched by mihai) ........ - -2007-07-18 22:59 +0000 [r75807] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Need to make sure we set milliseconds and - timestamp - pointed out by the recent ast_ time stuff from - Tilghman - -2007-07-18 21:09 +0000 [r75759] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 75757 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r75757 | russell | 2007-07-18 16:09:13 -0500 (Wed, 18 Jul 2007) | - 5 lines When traversing the queue of frames for possible - retransmission after receiving a VNAK, handle sequence number - wraparound so that all frames that should be retransmitted - actually do get retransmitted. (issue #10227, reported and - patched by mihai) ........ - -2007-07-18 20:40 +0000 [r75749] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c, /: Merged revisions 75748 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r75748 | tilghman | 2007-07-18 15:31:36 -0500 (Wed, 18 Jul 2007) - | 2 lines Store prior to copy (closes issue #10193) ........ - -2007-07-18 20:17 +0000 [r75732] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Umm, why are we transmitting dialtone on - cfwdall? - -2007-07-18 20:00 +0000 [r75712] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c, channels/chan_sip.c, channels/chan_agent.c, - pbx/pbx_realtime.c: Backport GCC 4.2 fixes. Without these - Asterisk won't build under devmode using GCC 4.2. - -2007-07-18 19:54 +0000 [r75707-75711] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Fixes for 7935/7936 conference phones. - Issue 9245, patch by slimey. - - * channels/chan_skinny.c: Fix issues with new 79x1 phones. Issue - 9887, patches by DEA - -2007-07-18 17:56 +0000 [r75658] Dwayne M. Hubbard <dhubbard@digium.com> - - * /, apps/app_queue.c: Merged revisions 75657 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r75657 | dhubbard | 2007-07-18 12:48:33 -0500 (Wed, 18 Jul 2007) - | 1 line removed the word 'pissed' from ast_log(...) function - call for BE-90 ........ - -2007-07-18 15:44 +0000 [r75583-75623] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Few more places that needs to check for - onhold state. - - * channels/chan_sip.c: (closes issue #10165) Reported by: elandivar - It is possible for hold status to exist without call limits set, - so we need to ensure update_call_counter is executed regardless. - - * channels/chan_h323.c: Don't bother reloading chan_h323 if it did - not load successfully in the first place. This would otherwise - cause a crash. - - * pbx/pbx_dundi.c: (closes issue #10224) Reported by: irroot Record - the threadid of each running thread before shutting them down as - the thread themselves may change the value. - -2007-07-18 12:29 +0000 [r75529] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_meetme.c: Using a freed frame causes crashes (closes - issue #9317) - -2007-07-17 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.8 released. - -2007-07-17 20:57 +0000 [r75441-75450] Russell Bryant <russell@digium.com> - - * /, channels/chan_skinny.c: Merged revisions 75449 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r75449 | russell | 2007-07-17 15:57:09 -0500 (Tue, 17 - Jul 2007) | 3 lines Properly check for the length in the skinny - packet to prevent an invalid memcpy. (ASA-2007-016) ........ - - * main/rtp.c: cast arguments to ast_log so that it builds without - warnings for me - - * channels/iax2-parser.c, channels/iax2-parser.h, /, - channels/chan_iax2.c: Merged revisions 75444 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r75444 | russell | 2007-07-17 15:45:27 -0500 (Tue, 17 Jul 2007) | - 5 lines Ensure that when encoding the contents of an ast_frame - into an iax_frame, that the size of the destination buffer is - known in the iax_frame so that code won't write past the end of - the allocated buffer when sending outgoing frames. (ASA-2007-014) - ........ - - * /, channels/chan_iax2.c: Merged revisions 75440 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r75440 | russell | 2007-07-17 15:41:41 -0500 (Tue, 17 Jul 2007) | - 4 lines After parsing information elements in IAX frames, set the - data length to zero, so that code later on does not think it has - data to copy. (ASA-2007-015) ........ - -2007-07-17 20:40 +0000 [r75439] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Ensure that the pointer to STUN data does not go to - unaccessible memory. (ASA-2007-017) - -2007-07-17 20:33 +0000 [r75437] Russell Bryant <russell@digium.com> - - * res/res_agi.c: (issue #10210) Reported by: juggie Patches: - 10210-1.4-grr.patch uploaded by juggie (license #24) Tested by: - juggie, blitzrage Log a warning if someone uses DeadAGI on a live - channel. - -2007-07-17 20:03 +0000 [r75405] Mark Michelson <mmichelson@digium.com> - - * apps/app_dial.c: Fixing an error I made earlier. ast_fileexists - can return -1 on failure, so I need to be sure that we only enter - the if statement if it is successful. Related to my fix to issue - #10186 - -2007-07-17 20:01 +0000 [r75401-75403] Russell Bryant <russell@digium.com> - - * main/pbx.c: (closes issue #10209) Reported by: juggie Patches: - 10209-trunk-2.patch uploaded by juggie Tested by: juggie, - blitzrage In ast_pbx_run(), mark a channel as hung up after an - application returned -1, or when it runs out of extensions to - execute. This is so that code can detect that this channel has - been hung up for things like making sure DeadAGI is used on - actual dead channels, and is beneficial for other things, like - making sure someone doesn't try to start spying on a channel that - is about to go away. - - * res/res_agi.c: Remove a duplicated newline character in AGI debug - output. (closes issue #10207, patch by seanbright) - -2007-07-16 20:53 +0000 [r75258-75306] Kevin P. Fleming <kpfleming@digium.com> - - * main/dns.c, /: Merged revisions 75304 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r75304 | kpfleming | 2007-07-16 15:46:58 -0500 (Mon, 16 Jul 2007) - | 3 lines provide proper copyright/license attribution for this - structure that was copied from a BSD-licensed header file long, - long ago... ........ - - * /: another fix that is not needed here (finishing up 75251) - -2007-07-16 18:16 +0000 [r75253] Mark Michelson <mmichelson@digium.com> - - * apps/app_dial.c: Restoring functionality from 1.2 wherein - Retrydial will not exit if there is no announce file specified. - This change makes it so that if there is no announce file - specified, the application will continue until finished (or - caller hangs up). If a bogus announce file is specified, then a - warning message will be printed saying that the file could not be - found, but execution will still continue. (closes issue #10186, - reported by jon, patched by me) - -2007-07-16 18:12 +0000 [r75252] Kevin P. Fleming <kpfleming@digium.com> - - * /: block change that is not relevant here - -2007-07-13 20:36 +0000 [r75108] Russell Bryant <russell@digium.com> - - * /, res/res_musiconhold.c: Merged revisions 75107 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r75107 | russell | 2007-07-13 15:35:22 -0500 (Fri, 13 - Jul 2007) | 3 lines Fix a couple potential minor memory leaks. - load_moh_classes() could return without destroying the loaded - configuration. ........ - -2007-07-13 20:15 +0000 [r75078] Mark Michelson <mmichelson@digium.com> - - * apps/app_chanspy.c, /: Merged revisions 75066 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r75066 | mmichelson | 2007-07-13 15:10:39 -0500 (Fri, 13 Jul - 2007) | 5 lines Fixed an issue where chanspy flags were - uninitialized if no options were passed. What triggered this - investigation was an IRC chat where some people's quiet flags - were set while others' weren't even though none of them had - specified the q option. ........ - -2007-07-13 20:10 +0000 [r75053-75067] Russell Bryant <russell@digium.com> - - * /, res/res_musiconhold.c: Merged revisions 75059 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r75059 | russell | 2007-07-13 15:07:21 -0500 (Fri, 13 - Jul 2007) | 6 lines Ensure that adding a user to the list of - users of a specific music on hold class is not done at the same - time as any of the other operations on this list to prevent list - corruption. Using the global moh_data lock for this is not ideal, - but it is what is used to protect these lists everywhere else in - the module, and I am only changing what is necessary to fix the - bug. ........ - - * channels/chan_zap.c, /: Merged revisions 75052 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r75052 | russell | 2007-07-13 14:10:00 -0500 (Fri, 13 Jul 2007) | - 12 lines (closes issue #9660) Reported by: mmacvicar Patches - submitted by: bbryant, russell Tested by: mmacvicar, marco, - arcivanov, jmhunter, explidous When using a TDM400P (and probably - other analog cards) there was a chance that you could hang up and - pick the phone back up where it has been long enough to be not - considered a flash hook, but too soon such that the device - reports that it is busy and the person on the phone will only - hear silence. This patch makes chan_zap more tolerant of this and - gives the device a couple of seconds to succeed so the person on - the phone happily gets their dialtone. ........ - -2007-07-12 23:00 +0000 [r74998] Mark Michelson <mmichelson@digium.com> - - * channels/chan_agent.c: Change to my previous fix regarding agent - logoff soft. Now uses deferlogoff instead of loginstart since - loginstart is used after logoff. Thanks to makoto for pointing - this out and suggesting the fix. (closes issue #10178, reported - and patched by makoto, with modification by me) - -2007-07-12 20:42 +0000 [r74955] Steve Murphy <murf@digium.com> - - * channels/chan_sip.c: This patch resolves 10143; thanks to irroot - for the patch; looked acceptable. Let the community decide if it - messes things up - -2007-07-12 19:17 +0000 [r74888-74922] Joshua Colp <jcolp@digium.com> - - * main/channel.c: Whoops... didn't want this to be returned to 0 - each iteration. - - * main/channel.c: When waiting for a digit ensure that a begin - frame was received with it, not just an end frame. (issue #10084 - reported by rushowr) - -2007-07-12 16:53 +0000 [r74839-74866] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: It helps if I actually add this stuff for - the 7921 too - otherwise it won't actually do much of anything. - - * channels/chan_skinny.c: Add device ID for 7921 wireless skinny - phone - - * channels/chan_skinny.c: Fix dialing in skinny that was broken in - some cases. Issue 10136, fix provided by DEA. - -2007-07-12 15:53 +0000 [r74815] Joshua Colp <jcolp@digium.com> - - * /, res/res_musiconhold.c: Merged revisions 74814 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r74814 | file | 2007-07-12 12:51:24 -0300 (Thu, 12 Jul - 2007) | 2 lines Only print out a warning for situations where it - is actually helpful. (issue #10187 reported by denke) ........ - -2007-07-11 22:57 +0000 [r74767] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 74766 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r74766 | russell | 2007-07-11 17:53:26 -0500 (Wed, 11 Jul 2007) | - 5 lines The function make_trunk() can fail and return -1 instead - of a valid new call number. Fix the uses of this function to - handle this instead of treating it as the new call number. This - would cause a deadlock and memory corruption. (possible cause of - issue #9614 and others, patch by me) ........ - -2007-07-11 21:14 +0000 [r74722] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_agent.c: Merged revisions 74719 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r74719 | mmichelson | 2007-07-11 16:12:30 -0500 (Wed, 11 - Jul 2007) | 5 lines The cli command "agent logoff Agent/x soft" - did not work...at all. Now it does. (closes issue #10178, - reported and patched by makoto, with slight modification for 1.4 - and trunk by me) ........ - -2007-07-11 18:34 +0000 [r74657] Russell Bryant <russell@digium.com> - - * res/res_config_odbc.c, /: Merged revisions 74656 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r74656 | russell | 2007-07-11 13:33:23 -0500 (Wed, 11 - Jul 2007) | 4 lines Make sure that the ESCAPE immediately follows - the condition that uses LIKE. This fixes realtime extensions with - ODBC. (closes issue #10175, reported by stuarth, patch by me) - ........ - -2007-07-11 18:18 +0000 [r74628-74642] Steve Murphy <murf@digium.com> - - * Makefile: This fixes 10172, where the entire man8 dir gets - removed during an uninstall of asterisk - - * utils/expr2.testinput, doc/channelvariables.txt, UPGRADE.txt: - further reversion of previously applied floating point stuff for - expr2 - -2007-07-11 17:16 +0000 [r74515-74590] Joshua Colp <jcolp@digium.com> - - * /: Blocked revisions 74587 via svnmerge ........ r74587 | file | - 2007-07-11 14:15:11 -0300 (Wed, 11 Jul 2007) | 2 lines Use some - Makefile magic to determine if linux/compiler.h is present. - (issue #10174 reported by francesco_r) ........ - - * channels/chan_phone.c, configure, - include/asterisk/autoconfig.h.in, configure.ac: Instead of - figuring out kernel versions that have compiler.h and not... - let's just use autoconf to check for it's presence. (issue #10174 - reported by francesco_r) - - * channels/chan_phone.c: Only check if we need to do a SIGMA based - tone generation if we have a card. (issue #10179 reported by - mikowhy) - -2007-07-10 23:32 +0000 [r74476] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Forwarding a message with IMAP storage was - storing the message in the sender's box instead of the forwarded - mailbox. (closes issue #10138, reported and patched by jaroth) - -2007-07-10 19:58 +0000 [r74374-74428] Jason Parker <jparker@digium.com> - - * /, apps/app_queue.c: Merged revisions 74427 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r74427 | qwell | 2007-07-10 14:57:20 -0500 (Tue, 10 Jul 2007) | 6 - lines Fix an issue where it was possible to have a service level - of over 100% Between the time recalc_holdtime and update_queue - was called, it was possible that the call could have been hungup. - Move both additions to the same place, so this won't happen. - Issue 10158, initial patch by makoto, modified by me. ........ - - * main/dns.c: Don't use #if to check if something is defined - use - #ifdef instead. Pointed out by kpfleming - - * /, channels/chan_agent.c: Merged revisions 74376 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r74376 | qwell | 2007-07-10 14:03:45 -0500 (Tue, 10 Jul - 2007) | 4 lines Fix an issue with wrapuptime not working when - using AgentLogin. Issue 10169, patch by makoto, with a minor mod - by me to not re-break issue 9618 ........ - - * main/dns.c, /, configure, include/asterisk/autoconfig.h.in, - configure.ac: Merged revisions 74373 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r74373 | qwell | 2007-07-10 13:37:23 -0500 (Tue, 10 Jul 2007) | 5 - lines Use res_ndestroy on systems that have it. Otherwise, use - res_nclose. This prevents a memleak on NetBSD - and possibly - others. Issue 10133, patch by me, reported and tested by scw - ........ - -2007-07-10 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.7.1 released. - -2007-07-10 16:00 +0000 [r74323] Russell Bryant <russell@digium.com> - - * res/res_musiconhold.c: fix an uninitialized variable - -2007-07-10 15:38 +0000 [r74317] Jason Parker <jparker@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 74316 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r74316 | qwell | 2007-07-10 10:37:54 -0500 (Tue, 10 Jul 2007) | 4 - lines Fix a small typo in description in of Voicemail() - application. Issue 10170, patch by casper. ........ - -2007-07-10 15:31 +0000 [r74314] Russell Bryant <russell@digium.com> - - * res/res_config_odbc.c, /: Merged revisions 74313 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r74313 | russell | 2007-07-10 10:30:20 -0500 (Tue, 10 - Jul 2007) | 3 lines Only use ESCAPE when LIKE is used. (issue - #10075, this part reported by jmls on IRC, patch by me) ........ - -2007-07-10 14:50 +0000 [r74262-74265] Joshua Colp <jcolp@digium.com> - - * /, main/app.c: Merged revisions 74264 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r74264 | file | 2007-07-10 11:48:00 -0300 (Tue, 10 Jul 2007) | 2 - lines Ensure the group information category exists before trying - to do a string comparison with it. (issue #10171 reported by - mlegas) ........ - - * channels/chan_sip.c: Only spit out an inringing warning message - when it is applicable. Since call limits are already toast in - realtime let's not scare the user if they are using it. (issue - #10166 reported by bcnit) - -2007-07-09 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.7 released. - -2007-07-09 21:31 +0000 [r74162-74211] Russell Bryant <russell@digium.com> - - * configure, configure.ac: Update the configure script to check for - a required function that is not present in the 1.2 version of - libpri. This will prevent the configure script from thinking that - it has compatible libpri support for Asterisk 1.4, when it - actually does not because the installed version is from 1.2. - - * /: Blocked revisions 74165 via svnmerge ........ r74165 | russell - | 2007-07-09 16:00:17 -0500 (Mon, 09 Jul 2007) | 4 lines When the - specified class isn't found, properly fall back to the channel's - music class or the default. (issue #10123, reported by blitzrage, - patches from juggie, qwell, and me) ........ - - * res/res_musiconhold.c: (closes issue #10123) Reported by: - blitzrage Patches submitted by: juggie, qwell, me Tested by: - blitzrage When trying to find a music on hold class to use, try - all of the options, instead of only the first one that is set. - Also, change the MusicOnHold applications to not hang up on the - channel when a class can not be found. - -2007-07-09 20:19 +0000 [r74159] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c, /: Merged revisions 74158 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r74158 | qwell | 2007-07-09 15:18:15 -0500 (Mon, 09 Jul 2007) | 8 - lines Several chan_zap options were not working on reload because - they were arbitrarily disallowed when reloading some/most PRI - options (such as signalling) was disallowed. Options such as - polarityonanswerdelay and answeronpolarityswitch can safely be - changed on a reload. This corrects that behavior. Issue 9186, - patch by tzafrir. ........ - -2007-07-09 18:38 +0000 [r74120-74122] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Forgot to get rid of an extraneous debug - message. - - * apps/app_queue.c: The n option for Queue should make the queue - exit immediately after failure to reach any members and should - not be dependent on the timeout value passed to Queue (closes - issue #10127, reported by bcnit, repaired by me) - -2007-07-09 15:32 +0000 [r74082] Joshua Colp <jcolp@digium.com> - - * channels/chan_skinny.c: Only destroy the scheduler context if it - was allocated. (issue #10124 reported by gzero) - -2007-07-09 14:57 +0000 [r74047] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Fixed a logic error in leave_voicemail. - Pass the mailbox instead of the context to inbox_count when the - context is "default." (closes issue #10135, reported by yannj, - repaired by me) - -2007-07-09 14:49 +0000 [r74043-74045] Joshua Colp <jcolp@digium.com> - - * channels/chan_skinny.c, pbx/pbx_dundi.c: Few minor thread - synchronization tweaks. (issue #10124 reported by gzero) - - * configure, acinclude.m4: Use AC_CHECK_HEADER to check for - ptlib/openh323 to allow for cross compiling. (issue #9675 - reported by zandbelt) - -2007-07-09 04:03 +0000 [r73985] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * main/ast_expr2f.c: Doxygen formatting fixes; fixes errors while - 'make progdocs'. (Closes issue #10104) - -2007-07-09 03:13 +0000 [r73930-73980] Joshua Colp <jcolp@digium.com> - - * main/cdr.c: Give Agent channel names priority when doing CDR - merging. (issue #10011 reported by krtorio) - - * pbx/pbx_config.c: Add a few sanity checks when writing out the - dialplan. (issue #10157 reported by dome) - -2007-07-08 09:47 +0000 [r73849] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: While tracking down a bug, I need some more - history. Dumphistory is very useful, indeed. - -2007-07-06 23:02 +0000 [r73769] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 73768 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r73768 | russell | 2007-07-06 18:01:22 -0500 (Fri, 06 Jul 2007) | - 4 lines If a sip_pvt struct has already registered an extension - state callback, remove the old one before adding a new one. If - this isn't done, Asterisk will crash. (issue #10120) ........ - -2007-07-06 16:36 +0000 [r73727] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Fixing a rare case which causes voicemail - to crash when compiled with IMAP storage. inboxcount has the - possibility of finding an "interactive" vm_state when no - persistent "non-interactive" vm_state exists for that mailbox. If - this should happen when someone attempts to leave a message, it - results in a crash. This patch, along with my commit in revision - 72670 fix issue 10053, reported by jaroth. closes issue #10053 - -2007-07-06 16:12 +0000 [r73679-73696] Russell Bryant <russell@digium.com> - - * res/res_config_odbc.c, /: Merged revisions 73684 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r73684 | russell | 2007-07-06 11:06:27 -0500 (Fri, 06 - Jul 2007) | 8 lines (closes issue #10075) Reported by: apsaras - Patches submitted by: Corydon76 Tested by: apsaras Fix a problem - with MSSQL 2005 by explicitly stating that '\' is being used as - an escape character. ........ - - * /, channels/chan_sip.c: Merged revisions 73678 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r73678 | russell | 2007-07-06 10:55:41 -0500 (Fri, 06 Jul 2007) | - 7 lines (closes issue #10125) Reported by: makoto Patches - submitted by: makoto This fixes a crash in chan_sip that happens - when the bindaddr setting is not valid on Asterisk startup, gets - fixed, and then a reload gets issued. ........ - -2007-07-06 15:27 +0000 [r73675] Mark Michelson <mmichelson@digium.com> - - * /, channels/chan_agent.c: Merged revisions 73674 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r73674 | mmichelson | 2007-07-06 10:26:40 -0500 (Fri, 06 - Jul 2007) | 5 lines Fixed a bug wherein agents get stuck busy. - (issue 9618, reported by jiddings, patched by moi) closes issue - #9618 ........ - -2007-07-06 03:34 +0000 [r73551-73629] Russell Bryant <russell@digium.com> - - * BUGS: fix a little spelling error - - * channels/chan_sip.c: Fix a crash in chan_sip. Don't try to stop - the monitor thread if it was never started. (closes issue #10124, - reported by gzero, fixed by me) - - * channels/chan_iax2.c: copy from the correct buffer when deferring - a full frame (related to issue #9937) - - * channels/chan_iax2.c: * Store the call number that a thread is - processing without the full frame bit set to ease debugging * - When deferring a full frame for processing, stick it into the - queue for the thread that is processing frames for that call, not - the one that read the current frame and is about to go back into - the idle list (related to issue #9937) - -2007-07-05 22:20 +0000 [r73548] Kevin P. Fleming <kpfleming@digium.com> - - * /, channels/chan_sip.c: Merged revisions 73547 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r73547 | kpfleming | 2007-07-05 17:11:51 -0500 (Thu, 05 Jul 2007) - | 2 lines we shouldn't allow G.723.1 endpoints to use VAD, just - like we don't support it for G.729 ........ - -2007-07-05 20:50 +0000 [r73512] Russell Bryant <russell@digium.com> - - * res/res_features.c: Pass HOLD and UNHOLD frames to the other - channel when they are returned from a native bridge function. - This fixes a problem where when two zap channels are natively - bridged and one does a flash hook, the other channel did not - receive music on hold. (Reported to me directly by Doug Bailey at - Digium) - -2007-07-05 19:18 +0000 [r73467] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 73466 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r73466 | file | 2007-07-05 16:15:18 -0300 (Thu, 05 Jul 2007) | 2 - lines Copy language information to the dialog structure when - calling a peer for situations where a PBX may be started on the - dialed channel. (issue #10121 reported by clegall_proformatique) - ........ - -2007-07-05 15:59 +0000 [r73400] Mark Michelson <mmichelson@digium.com> - - * apps/app_queue.c: Correcting a minor CLI bug I found. When - issuing the queue show command, if you type queue show and then - press tab, you can continue pressing tab and it will keep - auto-completing queue names even though only 1 queue can be used - as an argument. - -2007-07-05 15:28 +0000 [r73398] Russell Bryant <russell@digium.com> - - * channels/chan_vpb.cc, channels/Makefile: Make this module build - for me in dev-mode - -2007-07-05 14:21 +0000 [r73316-73355] Joshua Colp <jcolp@digium.com> - - * apps/app_chanspy.c, main/channel.c, /: Merged revisions 73349 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r73349 | file | 2007-07-05 11:19:14 -0300 (Thu, 05 Jul 2007) | 2 - lines Tweak spy locking. (issue #9951 reported by welles) - ........ - - * channels/chan_local.c, /: Merged revisions 73318 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r73318 | file | 2007-07-05 10:26:02 -0300 (Thu, 05 Jul - 2007) | 2 lines Actually check to make sure a PBX was started on - one of the Local channels instead of blindly assuming it was. - (issue #10112 reported by makoto) ........ - - * /, apps/app_queue.c: Merged revisions 73315 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r73315 | file | 2007-07-05 10:19:17 -0300 (Thu, 05 Jul 2007) | 2 - lines Reset ServicelevelPerf variable back to 0 if we are unable - to calculate it each time... otherwise we will get previous - values. (issue #10117 reported by noriyuki) ........ - -2007-07-04 14:53 +0000 [r73208-73253] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, /: Merged revisions 73252 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r73252 | crichter | 2007-07-04 16:50:58 +0200 (Mi, 04 - Jul 2007) | 1 line bchannel configurations like echocancel and - volume control, need to be setuped on inbound calls too. ........ - - * channels/chan_misdn.c, /: Merged revisions 73207 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r73207 | crichter | 2007-07-04 10:20:54 +0200 (Mi, 04 - Jul 2007) | 1 line bad bug in overlapdial case, we called - start_pbx multiple times, because the state wasn't changed.. - ........ - -2007-07-03 20:17 +0000 [r73143] Steve Murphy <murf@digium.com> - - * main/ast_expr2.fl, main/ast_expr2.c, main/Makefile, - main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2f.c: Removing - expr floating patch from 1.4; too much of a behavior change. If - you want this fix, try trunk instead. bug 9508. - -2007-07-03 15:42 +0000 [r73104-73106] Jason Parker <jparker@digium.com> - - * /: What the heck. This should not have happened. - - * /: use autotagged externals - -2007-07-03 12:38 +0000 [r73053] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_dial.c, /: Merged revisions 73052 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r73052 | tilghman | 2007-07-03 07:34:14 -0500 (Tue, 03 Jul 2007) - | 2 lines RetryDial should accept a 0 argument, but it does not, - because atoi does not distinguish between 0 and error (closes - issue #10106) ........ - -2007-07-03 08:17 +0000 [r73005] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /: Merged revisions 73004 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r73004 | crichter | 2007-07-03 10:04:35 +0200 (Di, 03 - Jul 2007) | 1 line fixed issue, that misdn_l2l1_check could only - be called from mISDN Source channels.. #9449 ........ - -2007-07-02 20:16 +0000 [r72933] Steve Murphy <murf@digium.com> - - * main/ast_expr2.fl, main/ast_expr2.c, utils/expr2.testinput, - main/Makefile, main/ast_expr2.h, main/ast_expr2.y, - main/ast_expr2f.c, doc/channelvariables.txt, UPGRADE.txt: support - for floating point numbers added to ast_expr2 $\[...\] exprs. - Fixes bug 9508, where the expr code fails with fp numbers. The - MATH function returns fp numbers by default, so this fix is - considered necessary. - -2007-07-02 18:18 +0000 [r72926] Russell Bryant <russell@digium.com> - - * main/manager.c: Remove a bogus comment and add proper locking to - the handler function for the CLI command to show information on - manager actions. - -2007-07-02 17:59 +0000 [r72925] Jason Parker <jparker@digium.com> - - * /: Blocked revisions 72924 via svnmerge ........ r72924 | qwell | - 2007-07-02 12:58:25 -0500 (Mon, 02 Jul 2007) | 4 lines Fix an - issue with playing "oclock" multiple times in French with 24 hour - time format. Issue 10101 ........ - -2007-07-02 14:32 +0000 [r72888] Joshua Colp <jcolp@digium.com> - - * main/channel.c: Added additional DTMF debug messages for when - emulation occurs. - -2007-07-02 08:41 +0000 [r72850-72852] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged - revisions 72585 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r72585 | crichter | 2007-06-29 15:08:26 +0200 (Fr, 29 Jun 2007) | - 1 line check if the bchannel stack id is already used, if so - don't use it a second time. Also added a release_chan lock, so - that the same chan_list object cannot be freed twice. chan_misdn - does not crash anymore on heavy load with these changes. ........ - - * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, - channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c: - Merged revisions 72099 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r72099 | crichter | 2007-06-27 15:22:37 +0200 (Mi, 27 Jun 2007) | - 1 line simplified generation for dummy bchannels, also we mark - them as dummies, so they are not used later as real-bchannels, - optimized the RESTART mechanisms, we block a channel now on - cause:44, and send out a RESTART automatically, then on reception - of RESTART_ACKNOWLEDGE we unblock the channel again. ........ - - * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, /: Merged - revisions 72087 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r72087 | crichter | 2007-06-27 11:26:53 +0200 (Mi, 27 Jun 2007) | - 1 line simplified channel finding and locking a lot. removed - unnecessary #ifdefed areas. ........ - -2007-07-01 23:52 +0000 [r72806] Russell Bryant <russell@digium.com> - - * pbx/pbx_spool.c, /: Merged revisions 72805 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r72805 | russell | 2007-07-01 18:51:34 -0500 (Sun, 01 Jul 2007) | - 5 lines When appending lines to call files to keep track of - retries, write a leading newline just in case the original call - file did not have a newline at the end. This fix is in response - to a problem I saw reported on the asterisk-users mailing list. - ........ - -2007-06-30 16:50 +0000 [r72705-72766] Russell Bryant <russell@digium.com> - - * configure, configure.ac: Tweak the configure script so that error - output isn't spewed to the console when searching for GTK2 libs, - and they aren't found. - - * formats/format_pcm.c: give format_pcm a more concise destription - -2007-06-29 19:07 +0000 [r72665] Luigi Rizzo <rizzo@icir.org> - - * main/utils.c: Use !defined(HAVE_GETHOSTBYNAME_R) to check for - absence of the function. This was already done in trunk. - -2007-06-29 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.6 released. - -2007-06-29 16:31 +0000 [r72630] Russell Bryant <russell@digium.com> - - * /: Blocked revisions 72629 via svnmerge ........ r72629 | russell - | 2007-06-29 11:30:56 -0500 (Fri, 29 Jun 2007) | 4 lines Backport - changes that make chan_iax2 not start the PBX on an incoming - channel until the three-way call setup is completed. These - changes are already in 1.4 and trunk. ........ - -2007-06-29 14:26 +0000 [r72597-72599] Joshua Colp <jcolp@digium.com> - - * main/cdr.c: Minor change for older GCC versions. - - * Makefile, configure, configure.ac, makeopts.in: Backport fix for - GCC versions without support for declaration-after-statement. - -2007-06-29 04:47 +0000 [r72554-72556] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * main/manager.c: Issue 10055 - Change memory allocation to use the - heap for a command, since the output has the potential to - overflow the stack (as it did here) - - * res/res_jabber.c: Fix 1.4 breakage - -2007-06-28 19:44 +0000 [r72493] Russell Bryant <russell@digium.com> - - * configure, include/asterisk/autoconfig.h.in: regenerate the - configure script for rizzo - -2007-06-28 19:29 +0000 [r72453-72489] Luigi Rizzo <rizzo@icir.org> - - * configure.ac: add a check for gethostbyname_r so we can simplify - the handling e.g. in utils.c Also add comments on a couple of - features which are not working on FreeBSD. All the above has been - already done in trunk so the merge must be blocked. Can someone - please regenerate ./configure ? - - * Makefile, channels/chan_zap.c, main/say.c: Add - -Wdeclaration-after-statement to AST_DEVMODE flags to catch - variable declarations in the middle of a block. Fix the few - instances of the above spotted out by the compiler. All of this - has been already done or is not applicable in trunk, so the merge - of this change will be blocked. - - * apps/app_meetme.c: cast a time_t so that it does not conflict - with the print format. This change was already done on trunk so - this change needs to be blocked from merging. - -2007-06-27 23:29 +0000 [r72383] Brett Bryant <bbryant@digium.com> - - * main/asterisk.c, /: Merged revisions 72373 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r72373 | bbryant | 2007-06-27 18:22:13 -0500 (Wed, 27 Jun 2007) | - 3 lines Reinstating patch. This actually fixes the problem, - however I was running a development branch without it and - mistakenly thought it wasn't fixed. Fixes issue #10010, and - #9654: 100% CPU usage caused by an asterisk console losing it's - controlling terminal. ........ - -2007-06-27 23:25 +0000 [r72381] Joshua Colp <jcolp@digium.com> - - * apps/app_mixmonitor.c, /: Merged revisions 72378 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r72378 | file | 2007-06-27 19:24:01 -0400 (Wed, 27 Jun - 2007) | 2 lines Update documentation to clarify variable usage - with MixMonitor. (issue #9494 reported by netoguy) ........ - -2007-06-27 23:03 +0000 [r72335] Brett Bryant <bbryant@digium.com> - - * main/asterisk.c, /: Merged revisions 72333 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r72333 | bbryant | 2007-06-27 17:58:53 -0500 (Wed, 27 Jun 2007) | - 2 lines Reverted changes for earlier revisions 72259 to 72261. - Issue #9654, #10010 ........ - -2007-06-27 22:58 +0000 [r72328-72331] Joshua Colp <jcolp@digium.com> - - * channels/chan_gtalk.c: Make payload IDs for iLBC/Speex match to - our list. Since these are dynamic payloads the other side - shouldn't care. (issue #9426 reported by irroot) - - * /, apps/app_queue.c: Merged revisions 72327 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r72327 | file | 2007-06-27 18:43:11 -0400 (Wed, 27 Jun 2007) | 2 - lines Fix issue where queue log events might be missing. (issue - #7765 reported by mtryfoss) ........ - -2007-06-27 21:08 +0000 [r72272] Russell Bryant <russell@digium.com> - - * /, pbx/pbx_config.c: Merged revisions 72267 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r72267 | russell | 2007-06-27 16:06:45 -0500 (Wed, 27 Jun 2007) | - 5 lines Fix a minor issue with parsing the priority number. You - could have as much whitespace as you want around a numeric - priority, but you couldn't have any whitespace around a special - priority like "n" or "hint". (issue #10039, reported by mitheloc, - fixed by me) ........ - -2007-06-27 20:46 +0000 [r72260] Brett Bryant <bbryant@digium.com> - - * main/asterisk.c, /: Merged revisions 72259 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r72259 | bbryant | 2007-06-27 15:43:53 -0500 (Wed, 27 Jun 2007) | - 4 lines Fixes 100% load when controlling terminal disappears. - Issue #9654, #10010 ........ - -2007-06-27 20:25 +0000 [r72257] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /: Merged revisions 72256 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r72256 | file | 2007-06-27 16:23:24 -0400 (Wed, 27 Jun 2007) | 2 - lines I may possibly get shot for doing this... but... defer CDR - processing until after the channel has been dealt with. This - should eliminate all of the issues with channels going funky - (SIP/PRI) when you are posting CDRs to a database that is either - slow or unavailable and do not want to enable batching. ........ - -2007-06-27 19:13 +0000 [r72205] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c: use the proper type for storing group number - bits so that if someone specifies 'group=42' it will actually - work instead of being silently ignored - -2007-06-27 18:40 +0000 [r72182-72185] Jason Parker <jparker@digium.com> - - * /: Blocked revisions 72184 via svnmerge ........ r72184 | qwell | - 2007-06-27 13:40:15 -0500 (Wed, 27 Jun 2007) | 4 lines Fix - another problem in voicemail with missing symbols. Issue 10074, - patch by kryptolus, extended to include #if 0'd blocks (just in - case) ........ - - * apps/app_voicemail.c: Fix another problem in voicemail with - missing symbols. Issue 10074, patch by kryptolus, extended to - include #if 0'd blocks (just in case) - -2007-06-27 17:31 +0000 [r72148] Joshua Colp <jcolp@digium.com> - - * main/channel.c: Make the ast_read_noaudio API call behave better - under circumstances where DTMF emulation was happening and a - generator was setup. (issue #10065 reported by stevefeinstein) - -2007-06-27 17:10 +0000 [r72125] Jason Parker <jparker@digium.com> - - * channels/chan_gtalk.c: Don't modify a variable that we don't want - modified. Make a copy of it instead. Issue 10029, patch by - phsultan with slight modifications by me (to remove needless - casts). - -2007-06-27 16:34 +0000 [r72112] Russell Bryant <russell@digium.com> - - * main/rtp.c: Only output debug information related to RTCP - timestamps when RTCP debug is turned on (issue #10066, patch by - me) - -2007-06-27 07:58 +0000 [r72042] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, /: Merged revisions 72040-72041 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r72040 | crichter | 2007-06-27 09:49:27 +0200 (Mi, 27 Jun 2007) | - 1 line for inbound TE calls, we setup the bchannel when we get - the CONNECT_ACKNOWLEDGE, to make sure mISDN has everything ready. - removed some #if 0 areas which weren't used anymore. ........ - r72041 | crichter | 2007-06-27 09:54:30 +0200 (Mi, 27 Jun 2007) | - 1 line isdn_lib.c didn't compile ........ - -2007-06-27 00:58 +0000 [r72006] Joshua Colp <jcolp@digium.com> - - * pbx/pbx_dundi.c: Make unloading of pbx_dundi actually work. - -2007-06-26 23:02 +0000 [r71953] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Removing a pointless line. This variable - was already set earlier and between then and this line, there is - no way that the values on the right side of the assignment could - have changed. - -2007-06-26 20:36 +0000 [r71915] Jason Parker <jparker@digium.com> - - * main/rtp.c: Don't dereference a pointer that may be NULL here. - Issue 10017. - -2007-06-26 19:00 +0000 [r71877] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: A few changes, the ultimate goal of which - is to keep better track of the number of messages that a mailbox - currently has. A description of the changes: 1. Changed the - "updated" field of the vm_state struct to act more as a binary - semaphore than a counting semaphore, since its current - implementation made the inboxcount function not work properly. - This change falls in line with a change made by UPenn with their - IMAP setup and helps to sync our changes with theirs. 2. - Eliminated some redundant calls to get_vm_state_by_mailbox inside - leave_voicemail 3. Use the play_folder variable to keep track of - the number of old and new messages in a mailbox as the messages - are deleted 4. Added an increment to the number of new messages - that was not there previously in the leave_voicemail function - -2007-06-26 17:49 +0000 [r71848] Jason Parker <jparker@digium.com> - - * /: Blocked revisions 71847 via svnmerge ........ r71847 | qwell | - 2007-06-26 12:49:14 -0500 (Tue, 26 Jun 2007) | 4 lines Don't try - to install an init script that doesn't exist. Reported to me on - #asterisk on Freenode IRC. ........ - -2007-06-26 15:47 +0000 [r71796] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Fixing bug where the authuser was - mistakenly pulled from the mailbox string instead of the IMAP - user. (closes issue 10054, reported and patched by jaroth) - -2007-06-26 12:27 +0000 [r71657-71751] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c, /: Merged revisions 71750 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r71750 | tilghman | 2007-06-26 07:25:58 -0500 (Tue, 26 Jun 2007) - | 2 lines Issue 10062 - Trying to move a message without - selecting one first results in memory corruption ........ - - * /, res/res_agi.c: Merged revisions 71656 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r71656 | tilghman | 2007-06-25 13:12:37 -0500 (Mon, 25 Jun 2007) - | 2 lines Issue 10035 - handle_exec returns a result inconsistent - with all of the other AGI commands ........ - -2007-06-25 14:13 +0000 [r71522-71576] Joshua Colp <jcolp@digium.com> - - * channels/chan_h323.c: Build a peer as well when hash323 is - enabled in users.conf (issue #9599 reported by asagage) - - * channels/chan_agent.c: Minor tweak for queueing up the unhold - frame... this will teach me to do bugs while half asleep. (issue - #10046 reported by dimas) - -2007-06-25 12:40 +0000 [r71519] Russell Bryant <russell@digium.com> - - * doc/asterisk-mib.txt: Fix a typo in the Asterisk mib. (issue - #10048, Matti) - -2007-06-25 01:10 +0000 [r71412-71430] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 71414 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r71414 | file | 2007-06-24 21:02:49 -0400 (Sun, 24 Jun 2007) | 2 - lines Ignore other URIs after the first in a 300 Multiple Choice - response. (issue #10041 reported by homesick) ........ - - * main/cdr.c: Fix it so 1.4 actually compiles on my box. - - * channels/chan_agent.c: Check to make sure the channel pointer is - present before queueing up an unhold frame on it. (issue #10046 - reported by dimas) - -2007-06-24 20:16 +0000 [r71362-71371] Russell Bryant <russell@digium.com> - - * build_tools/prep_tarball: Include the menuselect-tree file in - tarballs to make builds from tarballs a little bit faster - - * main/asterisk.c, /: Merged revisions 71358 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r71358 | russell | 2007-06-24 15:04:21 -0500 (Sun, 24 Jun 2007) | - 2 lines Revert the patch from issue 9654 due to an unexpected - side effect ........ - -2007-06-24 17:50 +0000 [r71289-71291] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * res/res_features.c: Issue 10044 - chan->cdr is NULL here, so - peer->cdr is what we really wanted to use - - * main/db.c, main/manager.c, /: Merged revisions 71288 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r71288 | tilghman | 2007-06-24 12:32:21 -0500 (Sun, 24 - Jun 2007) | 2 lines Issue 10043 - There is a legitimate need to - be able to set variables to the empty string. ........ - -2007-06-23 03:29 +0000 [r71230] Steve Murphy <murf@digium.com> - - * main/cdr.c, res/res_features.c: This patch is meant to fix 8433; - where clid and src are lost via bridging. - -2007-06-22 22:44 +0000 [r71214] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, - channels/chan_misdn.c, /: Merged revisions 70341 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r70341 | crichter | 2007-06-20 17:29:09 +0200 (Mi, 20 - Jun 2007) | 1 line fixed a bug that was introduced by copy and - paste in the last commit ..bchannels weren't cleaned properly. - ........ - -2007-06-22 16:05 +0000 [r71128] Joshua Colp <jcolp@digium.com> - - * /: Blocked revisions 71124 via svnmerge ........ r71124 | file | - 2007-06-22 12:02:40 -0400 (Fri, 22 Jun 2007) | 2 lines Send an - unhold indication when going off hold. (issue #10036 reported by - speedy) ........ - -2007-06-22 15:38 +0000 [r71096-71123] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged - revisions 70672 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r70672 | crichter | 2007-06-21 15:11:29 +0200 (Do, 21 Jun 2007) | - 1 line we activate the bchannels in TE mode on incoming calls - only when we want to connect the call. ........ - - * channels/misdn/isdn_lib.c, /: Merged revisions 70342 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r70342 | crichter | 2007-06-20 17:42:39 +0200 (Mi, 20 - Jun 2007) | 1 line forgot one place .. ........ - - * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, - channels/chan_misdn.c, /: Merged revisions 70311 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r70311 | crichter | 2007-06-20 16:47:59 +0200 (Mi, 20 - Jun 2007) | 1 line on receiption of cause:44 we mark the channel - as in use and inform the user about the situation, we need to - test the RESTART stuff then. Also shuffled the - empty_chan_in_stack function after the bchannel cleaning - functions, to avoid race conditions. ........ - - * channels/chan_misdn.c, /: Merged revisions 69887 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r69887 | crichter | 2007-06-19 15:23:04 +0200 (Di, 19 - Jun 2007) | 1 line when we send out a SETUP, but get no response, - we should cleanup everything after reception of a hangup. - ........ - - * /, channels/misdn/isdn_msg_parser.c: Merged revisions 69053 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r69053 | crichter | 2007-06-13 11:55:54 +0200 (Mi, 13 Jun 2007) | - 1 line restart indicator 0x80 is correct, at least that's what - libpri does. ........ - - * channels/chan_misdn.c, /: Merged revisions 68887 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r68887 | crichter | 2007-06-12 10:35:22 +0200 (Di, 12 - Jun 2007) | 1 line if the bridged partner is mISDN too we should - not send dtmf tones, they are transmitted inband always ........ - - * channels/chan_misdn.c, /: Merged revisions 68874 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r68874 | crichter | 2007-06-12 09:48:52 +0200 (Di, 12 - Jun 2007) | 1 line if we have already some digits, we just stop - the tones. ........ - -2007-06-22 15:00 +0000 [r71068] Jason Parker <jparker@digium.com> - - * apps/app_speech_utils.c, /, res/res_agi.c, main/file.c: Merged - revisions 71065 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r71065 | qwell | 2007-06-22 09:52:18 -0500 (Fri, 22 Jun 2007) | 4 - lines Fix a few silly usages of ast_playstream() - it only ever - returns 0... Issue 10035 ........ - -2007-06-22 14:53 +0000 [r71066] Brett Bryant <bbryant@digium.com> - - * main/asterisk.c, /: Merged revisions 71064 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r71064 | bbryant | 2007-06-22 09:39:34 -0500 (Fri, 22 Jun 2007) | - 10 lines Fixed infinite loop when controlling terminal was lost - and return value of input function wasn't checked for errors. - This would cause 100% cpu to be taken up. (closes issue #9654, - issue #10010) Reported by: mnicholson, and eserra Idea for the - patch from mnicholson, patched by me ........ - -2007-06-22 14:10 +0000 [r71063] Steve Murphy <murf@digium.com> - - * main/cdr.c: My conditions for merging amaflags info was naive; - DOCUMENTATION is the default, although null is possible; theft of - user-settable fields is not good. Just copy them, leave them - alone. - -2007-06-22 03:14 +0000 [r71003] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Fix a small typo which ... well ... - completely broke chan_iax2. oops! (issue #9937, patch by me) - -2007-06-21 22:34 +0000 [r70949] Steve Murphy <murf@digium.com> - - * main/cdr.c, /: Merged revisions 70948 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r70948 | murf | 2007-06-21 16:29:50 -0600 (Thu, 21 Jun 2007) | 1 - line This little fix is in response to bug 10016, but may not - cure it. The code is wrong, clearly. In a situation where you set - the CDR's amaflags, and then ForkCDR, and then set the new CDR's - amaflags to some other value, you will see that all CDRs have had - their amaflags changed. This is not good. So I fixed it. ........ - -2007-06-21 21:40 +0000 [r70899] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 70898 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r70898 | file | 2007-06-21 17:37:55 -0400 (Thu, 21 Jun 2007) | 2 - lines Don't explode if the gain option is specified without a - value. (issue #9274 reported by mfarver) ........ - -2007-06-21 21:14 +0000 [r70866-70883] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Put the thread reading from the socket back - in the idle list if it deferred the processing of a full frame to - another thread - - * channels/chan_iax2.c: If a full frame is received while one of - the iax2 threads is in the middle of handling a full frame for - the same call, queue it up for processing by that same thread - later instead of dropping it. (issue #9937, patch by me) - -2007-06-21 20:19 +0000 [r70841] Steve Murphy <murf@digium.com> - - * cdr/cdr_custom.c, /: Merged revisions 70804 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r70804 | murf | 2007-06-21 13:13:17 -0600 (Thu, 21 Jun 2007) | 1 - line it was pointed out that the cdr_custom config load could get - a lock, and under certain circumstances, would never release it. - I also noted that the situation where more than one mapping spec - was warned about, but did not ignore further mappings as it had - promised. I think I have fixed both situations. ........ - -2007-06-21 19:49 +0000 [r70808] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: When volgain is used don't leave a - temporary file behind. (Closes Issue 8514, Reported and patched - by ulogic, code reviewed by Jason Parker) - -2007-06-21 15:22 +0000 [r70727] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Do not Packet2Packet bridge if packetization settings - do not allow it. (issue #9117 reported by phsultan) - -2007-06-21 15:21 +0000 [r70726] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c: Remove a couple of duplicate unlocks - -2007-06-21 13:58 +0000 [r70677] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Fix building with ODBC storage enabled. - (issue #10025 reported by denisgalvao) - -2007-06-21 13:00 +0000 [r70656] Steve Murphy <murf@digium.com> - - * main/cdr.c: Via complaints aired in asterisk-users, I submit - these changes, which allow cdr updates to see macro - context/exten, whether hung up or not - -2007-06-20 23:32 +0000 [r70554-70612] Jason Parker <jparker@digium.com> - - * cdr/cdr_pgsql.c: Fix some potential memory leaks in cdr_pgsql. - Issue 10020, patch by my, with credit to prashant_jois for - pointing out the problem. - - * cdr/cdr_pgsql.c: Fix a stupid mistake in my last cdr_pgsql race - condition fix - - * cdr/cdr_pgsql.c: Fix a race condition in cdr_pgsql that can occur - when reloading the module. Issue 10022, patch by me, with credit - to prashant_jois for finding the bug. - -2007-06-20 22:22 +0000 [r70552] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 70551 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r70551 | file | 2007-06-20 18:20:16 -0400 (Wed, 20 Jun 2007) | 2 - lines Don't overwrite the configured username setting upon a - REGISTER. (issue #8565 reported by jsmith) ........ - -2007-06-20 20:53 +0000 [r70494] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Make sure we clear the previously dialed - number if it did not exist. Issue 9958. - -2007-06-20 19:29 +0000 [r70445] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_dial.c, /: Merged revisions 70444 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r70444 | tilghman | 2007-06-20 14:25:54 -0500 (Wed, 20 Jun 2007) - | 2 lines Issue 9997 - Timelimit times out the wrong channel - ........ - -2007-06-20 18:46 +0000 [r70397] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c, /: Merged revisions 70396 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r70396 | russell | 2007-06-20 13:45:38 -0500 (Wed, 20 Jun 2007) | - 5 lines Fix a problem where an established call would not be - properly disconnected when a PRI disconnect is received depending - on which cause code was received. (issue #9588, original patch by - softins, updated patch from jtexter3, and some additional - feedback from mhardeman) ........ - -2007-06-20 17:52 +0000 [r70198-70360] Joshua Colp <jcolp@digium.com> - - * main/rtp.c, main/frame.c: Put the speex packetization values back - in but disable it when setting up the smoother. - - * main/frame.c: Don't do packetization/smoother stuff with speex, - it doesn't work. - -2007-06-20 00:03 +0000 [r70084-70164] Russell Bryant <russell@digium.com> - - * contrib/scripts/ast_grab_core: don't delete the backtrace in - ast_grab_core - - * channels/chan_gtalk.c: Only attempt to queue a hangup on the - owner channel if it actually exists. (issue #9795, patch from - zandbelt) - -2007-06-19 18:23 +0000 [r70062] Steve Murphy <murf@digium.com> - - * main/channel.c, /: Merged revisions 70053 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r70053 | murf | 2007-06-19 12:07:59 -0600 (Tue, 19 Jun 2007) | 1 - line This fixes 9246, where channel variables are not available - in the 'h' exten, on a 'ZOMBIE' channel. The fix is to - consolidate the channel variables during a masquerade, and then - copy the merged variables back onto the clone, so the zombie has - the same vars that the 'original' has. ........ - -2007-06-19 17:07 +0000 [r70003] Joshua Colp <jcolp@digium.com> - - * main/rtp.c, /: Merged revisions 69992 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r69992 | file | 2007-06-19 13:00:58 -0400 (Tue, 19 Jun 2007) | 2 - lines Handle the CC field in the RTP header. (issue #9384 - reported by DoodleHu) ........ - -2007-06-19 16:46 +0000 [r69991] Russell Bryant <russell@digium.com> - - * /: Blocked revisions 69990 via svnmerge ........ r69990 | russell - | 2007-06-19 11:45:37 -0500 (Tue, 19 Jun 2007) | 12 lines - Backport fix for crashes related to subscriptions from 1.4 ... - Fix a crash that could occur when handing device state changes. - When the state of a device changes, the device state thread tells - the extension state handling code that it changed. Then, the - extension state code calls the callback in chan_sip so that it - can update subscriptions to that extension. A pointer to a - sip_pvt structure is passed to this function as the call which - needs a NOTIFY sent. However, there was no locking done to ensure - that the pvt struct didn't disappear during this process. (issue - #9946, reported by tdonahue, patch by me, patch updated to trunk - to use the sip_pvt lock wrappers by eliel) ........ - -2007-06-19 16:24 +0000 [r69987] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /: Merged revisions 69986 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r69986 | file | 2007-06-19 12:21:29 -0400 (Tue, 19 Jun 2007) | 2 - lines Update BRIDGEPEER variable if set to the new channel name - when a masquerade happens. (issue #9699 reported by dimas) - ........ - -2007-06-19 15:22 +0000 [r69944] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Fix a crash that could occur when handing - device state changes. When the state of a device changes, the - device state thread tells the extension state handling code that - it changed. Then, the extension state code calls the callback in - chan_sip so that it can update subscriptions to that extension. A - pointer to a sip_pvt structure is passed to this function as the - call which needs a NOTIFY sent. However, there was no locking - done to ensure that the pvt struct didn't disappear during this - process. (issue #9946, reported by tdonahue, patch by me, patch - updated to trunk to use the sip_pvt lock wrappers by eliel) - -2007-06-19 13:55 +0000 [r69805-69895] Joshua Colp <jcolp@digium.com> - - * /, apps/app_meetme.c: Merged revisions 69894 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r69894 | file | 2007-06-19 09:54:03 -0400 (Tue, 19 Jun 2007) | 2 - lines Perform an extra hangup check just in case. (issue #9589 - reported by bcnit) ........ - - * /, res/res_features.c: Merged revisions 69846 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r69846 | file | 2007-06-19 08:57:55 -0400 (Tue, 19 Jun 2007) | 2 - lines Add parked call extension AFTER the parking slot has been - announced, otherwise two threads will try to handle the same - channel and it will go kaboom. (issue #9191 reported by japple) - ........ - - * main/callerid.c: Fix for building on PowerPC under Linux. - -2007-06-18 19:48 +0000 [r69796] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * channels/chan_sip.c: Issue 10005 - Segfault with missing - arguments, plus fix a missing define for SIP INFO channels - -2007-06-18 19:00 +0000 [r69775-69794] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Don't count RTP timeout when involved in a - T38 fax session. (issue #9222 reported by ivoc) - - * /, channels/chan_sip.c: Merged revisions 69765 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r69765 | file | 2007-06-18 14:13:03 -0400 (Mon, 18 Jun 2007) | 2 - lines Set the peer name on the dialog to the one configured in - sip.conf and NOT the username to be used for authentication - attempts. (issue #9967 reported by achauvin) ........ - -2007-06-18 17:46 +0000 [r69744] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * contrib/scripts/safe_asterisk, /: Merged revisions 69743 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r69743 | tilghman | 2007-06-18 12:45:15 -0500 (Mon, 18 Jun 2007) - | 2 lines Issue 9998 - Remove SIG prefix, since it's not - supported by ksh ........ - -2007-06-18 16:51 +0000 [r69708] Joshua Colp <jcolp@digium.com> - - * main/dnsmgr.c: Remember the DNS lookup done when dnsmgr is called - for the first time so that it does not needlessly spit out - changed messages when the host really didn't change. - -2007-06-18 16:35 +0000 [r69689-69702] Russell Bryant <russell@digium.com> - - * res/res_odbc.c, apps/app_voicemail.c, res/res_config_odbc.c, - build_tools/menuselect-deps.in, configure, funcs/func_odbc.c, - include/asterisk/autoconfig.h.in, configure.ac, cdr/cdr_odbc.c: - To prevent 92138749238754 more reports of "I have unixodbc - installed, but still can't build *_odbc.so!", check for ltdl - directly, instead of just listing it as another library to - include in the unixodbc check in the configure script. This also - makes ltdl show up as a dependency in menuselect so people know - what to go install. (related to issue #9989, patch by me) - - * build_tools/prep_moduledeps: Change the use of "echo -e" to - "printf". On systems where /bin/sh is not bash, most of the lines - in menuselect-tree were getting a "-e" at the beginning of every - line. I'm surprised nobody noticed this, but I think the XML - parser was being very nice and ignoring them. - -2007-06-18 16:04 +0000 [r69661-69668] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Don't defer the BYE till later on a transfer - when the transfer itself goes kaboom and has no hope of working. - - * channels/chan_sip.c: Few minor transfer tweaks. We can't unlock - something we never locked, and better handle a specific scenario - with doing an attended transfer between two non-bridged calls. - -2007-06-18 15:46 +0000 [r69660] Russell Bryant <russell@digium.com> - - * Makefile: Tweak paths for BSD systems (issue #10001, stuarth) - -2007-06-18 13:55 +0000 [r69625] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Fix issue where it would be possible for the - negotiated codecs to get set back to nothing. (issue #9992 - reported by yehavi) - -2007-06-15 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.5 released. - -2007-06-15 20:18 +0000 [r69579] Russell Bryant <russell@digium.com> - - * res/res_features.c: Fix a silly deadlock in res_features that I - found while debugging on one of blitzrage's test machines. It was - one of the situations where he was seeing hung channels, and may - be the cause of some of the reports from other people. (related - to issue #9235) - -2007-06-15 19:23 +0000 [r69558] Joshua Colp <jcolp@digium.com> - - * apps/app_speech_utils.c: Add support for setting the maximum - length of acceptable DTMF in SpeechBackground. - -2007-06-15 15:27 +0000 [r69518] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c: The SLATRUNK_STATUS variable indicated - "SUCCESS" for both an answer of the incoming call on the trunk, - or if the trunk reached its ring timeout. This patch changes the - variable to say "RINGTIMEOUT" in that case. (issue #9973, - reported by n00dle, patch by me) - -2007-06-14 23:22 +0000 [r69434-69470] Jason Parker <jparker@digium.com> - - * main/config.c, /: Merged revisions 69469 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r69469 | qwell | 2007-06-14 18:21:45 -0500 (Thu, 14 Jun 2007) | 4 - lines Fix an issue where the line number in an unterminated - comment block error message would show the wrong line number. - "Reported" to me on #asterisk (somebody posted an error message, - and I happened to catch it) ........ - - * sounds/Makefile: Update to latest versions of sound files. - -2007-06-14 21:50 +0000 [r69392] Kevin P. Fleming <kpfleming@digium.com> - - * cdr/cdr_tds.c, cdr/cdr_csv.c, main/cdr.c, channels/chan_phone.c, - cdr/cdr_sqlite.c, main/logger.c, main/callerid.c, cdr/cdr_odbc.c, - main/asterisk.c, channels/chan_mgcp.c, cdr/cdr_manager.c, - apps/app_voicemail.c, include/asterisk/utils.h, main/pbx.c, - main/say.c, cdr/cdr_pgsql.c, cdr/cdr_radius.c, - channels/chan_iax2.c: use ast_localtime() in every place - localtime_r() was being used - -2007-06-14 21:08 +0000 [r69358] Russell Bryant <russell@digium.com> - - * main/say.c: Fix some problems with saying dates and times for the - "tw" langauge (issue #9964, ljmid) - -2007-06-14 15:21 +0000 [r69259] Jason Parker <jparker@digium.com> - - * funcs/func_groupcount.c, /: Merged revisions 69258 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r69258 | qwell | 2007-06-14 10:15:53 -0500 (Thu, 14 Jun - 2007) | 4 lines Change a quite broken while loop to a for loop, - so "continue;" works as expected instead of eating 99% CPU... - Issue 9966, patch by me. ........ - -2007-06-13 21:19 +0000 [r69184-69222] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Whoops... - - * channels/chan_iax2.c: Let's make chan_iax2 media only native - transfers actually work. (issue #9376 reported by simone - cittadini) - - * channels/iax2-parser.c: Add TXMEDIA to list so that it is - properly displayed during iax2 packet output. - -2007-06-13 19:57 +0000 [r69183] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Move the logic for destroying a call when no - response is received to a BYE outside of the block that checks - for FLAG_FATAL to be set. This flag is only set when the packet - is transmitted with the reliability set to XMIT_CRITICAL when the - original packet is transmitted. A BYE is always sent with it set - to XMIT_RELIABLE, meaning this code could never be encountered. - This resulted in seeing some SIP channels that would never go - away with the last packet sent being a BYE. (part of issue #9235, - patch from jcmoore) - -2007-06-13 19:41 +0000 [r69181] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Contains a patch for fixing an encoding - problem when using Outlook to view voicemail emails and - attachments. This fix has also been tested on Thunderbird, - Evolution, Pine, and Mutt. (Issue 9336, reported by marwick, - patched by mutterc) - -2007-06-13 19:08 +0000 [r69128-69144] Joshua Colp <jcolp@digium.com> - - * apps/app_meetme.c: Really ignore NULL frames and check whether - the channel hungup or not. (issue #9912 reported by junky) - - * /, main/app.c: Merged revisions 69127 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r69127 | file | 2007-06-13 14:12:48 -0400 (Wed, 13 Jun 2007) | 2 - lines Return group counting to previous behavior where you could - only have one group per category. (issue #9711 reported by - irroot) ........ - -2007-06-13 16:56 +0000 [r69016-69071] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Clarify a bit of logic. This doesn't change - behavior in any way, but it is helpful when following the logic - to debug problems like 9235. - - * channels/chan_iax2.c: Fix a place where a chan_iax2 pvt struct - was accessed without the lock held. This issue was reported to me - via email by Dmitry Mishchenko. Thanks! - - * cdr/cdr_pgsql.c: Fix a memory leak pointed out by prashant_jois - in #asterisk-bugs. PQclear() was not called on the result - structure after doing a PQexec(). Also, fix up some formatting in - passing. - -2007-06-12 19:36 +0000 [r69012-69014] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Change the full frame dropping log message - to debug to avoid future bug reports. - - * channels/chan_iax2.c: Schedule the sending of a PING packet a - second later than previously so that it does not collide with the - LAGRQ. - -2007-06-12 19:13 +0000 [r69010] Russell Bryant <russell@digium.com> - - * main/channel.c: In ast_channel_make_compatible(), just return if - the channels' read and write formats already match up. There are - code paths that call this function on a pair of channels multiple - times. This made calls fail that were using g729 in some cases. - The reason is that codec_g729a will unregister itself from the - list of available translators will all licenses are in use. So, - the first time the function got called, the right translation - path was allocated. However, the second time it got called, the - code would not find a translation path to/from g729 and make the - call fail, even if the channel actually already had a g729 - translation path allocated. (SPD-32) - -2007-06-12 14:23 +0000 [r68922] Joshua Colp <jcolp@digium.com> - - * main/rtp.c, /: Merged revisions 68921 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r68921 | file | 2007-06-12 10:18:57 -0400 (Tue, 12 Jun 2007) | 2 - lines Bring RTP back to Asterisk at the end of a native bridge no - matter what. ........ - -2007-06-11 21:20 +0000 [r68814] Jason Parker <jparker@digium.com> - - * include/asterisk/time.h: Solaris 10 sometimes (?) needs this - include in order to have NULL defined. - -2007-06-11 20:45 +0000 [r68781] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_directory.c: Issue 9947 - fn2 was unused / incorrectly - used - -2007-06-11 16:57 +0000 [r68733] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c: - Merged revisions 68732 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r68732 | crichter | 2007-06-11 18:49:00 +0200 (Mo, 11 Jun 2007) | - 1 line added check for NULL Pointer when calling misdn_new. - Asterisk does not allow us to create channels anymore when stop - gracefully is used :). also modified the restart_indicator to 0 - ........ - -2007-06-11 14:33 +0000 [r68683] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /: Merged revisions 68682 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r68682 | file | 2007-06-11 10:29:58 -0400 (Mon, 11 Jun 2007) | 2 - lines Improve deadlock handling of the channel list. (issue #8376 - reported by one47) ........ - -2007-06-11 10:29 +0000 [r68644] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, - channels/chan_misdn.c, /, channels/misdn/ie.c, - channels/misdn/isdn_msg_parser.c: Merged revisions 68631 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r68631 | crichter | 2007-06-11 11:18:01 +0200 (Mo, 11 Jun 2007) | - 1 line fixed problem that the dummybc chanels had no lock, - checking for the lock now. Also fixed the channel restart stuff, - we can now specify and restart particular channels too. ........ - -2007-06-11 04:21 +0000 [r68595] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * pbx/pbx_config.c: "dialplan save" produced garbage in the config - file - -2007-06-08 22:23 +0000 [r68527] Russell Bryant <russell@digium.com> - - * /, apps/app_dictate.c: Merged revisions 68526 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r68526 | russell | 2007-06-08 17:22:36 -0500 (Fri, 08 Jun 2007) | - 4 lines Don't automatically hang up after running Dictate so that - callers can exit cleanly using '#' (closes issue #9577, patch - from Thomas Andrews) ........ - -2007-06-08 15:52 +0000 [r68450] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_iax2.c: actually remember the type/subclass of full - frames that are in process - -2007-06-08 00:17 +0000 [r68370-68401] Joshua Colp <jcolp@digium.com> - - * /, main/say.c: Merged revisions 68397 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r68397 | file | 2007-06-07 20:15:33 -0400 (Thu, 07 Jun 2007) | 2 - lines Don't call ast_waitstream_full when the control file - descriptor and audio file descriptor are not set, simply call - ast_waitstream! (issue #8530 reported by rickead2000) ........ - - * main/dnsmgr.c, /: Merged revisions 68368 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r68368 | file | 2007-06-07 19:59:04 -0400 (Thu, 07 Jun 2007) | 2 - lines Do a DNS lookup immediately upon calling the dnsmgr - function, don't wait until a refresh happens. (issue #9097 - reported by plack) ........ - -2007-06-07 23:14 +0000 [r68354] Russell Bryant <russell@digium.com> - - * /, main/say.c: Merged revisions 68351 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r68351 | russell | 2007-06-07 18:13:33 -0500 (Thu, 07 Jun 2007) | - 3 lines Fix a problem where saying a character wouldn't properly - break out when the caller pressed '#' (issue #8113, reported by - patbaker82, patch from jamesgolovich (hey, long time no see!) and - patbaker82) ........ - -2007-06-07 23:00 +0000 [r68326] Jason Parker <jparker@digium.com> - - * apps/app_voicemail.c: Fix incorrect French syntax of "old - messages". Request for feedback was sent to asterisk-dev mailing - list, with little response. Issue 9118, patch by junky. - -2007-06-07 22:14 +0000 [r68313] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_iax2.c: some improvements to the IAX2 full frame - dropping logic recently added: - use inaddrcmp(), since we have - it - output the type of frame and subclass being dropped, and the - type/subclass that is already being processed (which caused the - drop) - -2007-06-07 21:16 +0000 [r68280] Russell Bryant <russell@digium.com> - - * channels/chan_agent.c, apps/app_queue.c: Fix loading persistent - queue members when using realtime configuration for queues. Also, - remove an unneeded leading slash for the astdb family. (issue - #9911, patch by atis) - -2007-06-07 20:25 +0000 [r68211-68249] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Fix an issue with newer phones which - require packets be padded out to the correct length. Issue 9887, - patch by DEA. - - * apps/app_voicemail.c, /: Merged revisions 68204 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r68204 | qwell | 2007-06-07 15:02:50 -0500 (Thu, 07 Jun 2007) | 4 - lines Don't try to save voicemail greetings unless the user - presses '1' to accept/save. Issue 9904, patch by me. ........ - -2007-06-07 19:47 +0000 [r68198] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Submitting a fix for Issue 8016. Added a - check to make sure that greetings get stored properly. (Issue - 8016, reported by edhorton, patched by alamantia with - modification by me. Thanks to Jason Parker for the advice on - this). - -2007-06-07 19:46 +0000 [r68196] Olle Johansson <oej@edvina.net> - - * channels/chan_features.c: Disable chan_features by default in - menuselect - -2007-06-07 19:30 +0000 [r68192] Russell Bryant <russell@digium.com> - - * main/strcompat.c: Include stdarg.h for build issues on Solaris - (issue #9381) - -2007-06-07 18:39 +0000 [r68071-68157] Joshua Colp <jcolp@digium.com> - - * main/channel.c: Fix logic when doing a name based channel search - for a structure when you want to start from a specific point in - the channel list. (issue #9324 reported by slavon) - - * apps/app_dial.c, /: Merged revisions 68070 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r68070 | file | 2007-06-07 10:19:40 -0400 (Thu, 07 Jun 2007) | 2 - lines Allow the 'g' option to work if used with the 'S' option. - (issue #9888 reported by gasparz) ........ - -2007-06-07 10:00 +0000 [r67993-68030] Olle Johansson <oej@edvina.net> - - * res/res_jabber.c: Adding a few Todo's to res_jabber so we don't - forget. - - * res/res_jabber.c: Ok, we found out that this is not about if you - have any *active* clients using TLS, but if you have initialized - TLS at all during the lifetime of the module. So if you reload to - disable TLS, it won't help. - - * res/res_jabber.c: If you have a jabber client that uses TLS, - refuse unload. Bad fix, but will prevent crashes while we are - trying to find a workaround. Iksemel development seems to have - stalled and we might have to stop using the TCP/TLS connections - in that library and use our own, which would scale better from a - poll/select perspective I guess. It would also make it easier to - migrate to OpenSSL and stop Asterisk from depending on both - OpenSSL and GnuTLS. - - * include/asterisk/jabber.h, res/res_jabber.c: Issue #9738 - Make - sure we can unload res_jabber. Patch by phsultan - thanks! Due to - a bug in the iksemel library, this will not work if you are using - GTLS in the connection. That's being investigated. If you figure - out a way to handle that without us having to patch iksemel, let - us know in the bug report. Thanks. - -2007-06-07 00:10 +0000 [r67924-67941] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 67938 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r67938 | file | 2007-06-06 20:09:13 -0400 (Wed, 06 Jun 2007) | 2 - lines Only notify the devicestate system of a peer state change - when the peer is built from the config file. (issue #9900 - reported by arkadia) ........ - - * main/file.c: Properly handle cases where a stream can't be - written to. (issue #9757 reported by junky) - -2007-06-06 22:08 +0000 [r67862-67872] Russell Bryant <russell@digium.com> - - * res/res_snmp.c: Disable reload functionality in res_snmp. It is - not possible to initialize the snmp library more than once - without completely unloading the module and loading it again. - (issue #9571, reported by hristo, additional helpful debug - information from festr, patch from me) - - * channels/chan_sip.c: Fix a crash when doing call pickups with SIP - phones. The code unlocked the channel when it should not have. - (issue #9652, reported by corruptor, fixed by me) - -2007-06-06 19:26 +0000 [r67804] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Fix for Issue 9810. There was a segfault - under a specific set of circumstances: 1. VoiceMailMain was - configured in the dialplan with an extension as its argument 2. A - message was left for this mailbox 3. Tried to call VoiceMailMain - but hung up before entering password. This was fixed by checking - that a pointer was non-null prior to trying to dereference it. - (Issue 9810, reported by xmarksthespot, patched by Corydon76 with - modifications by me). - -2007-06-06 16:55 +0000 [r67716] Russell Bryant <russell@digium.com> - - * main/channel.c, /, include/asterisk/linkedlists.h: Merged - revisions 67715 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r67715 | russell | 2007-06-06 11:40:51 -0500 (Wed, 06 Jun 2007) | - 5 lines We have some bug reports showing crashes due to a double - free of a channel. Add a sanity check to ast_channel_free() to - make sure we don't go on trying to free a channel that wasn't - found in the channel list. (issue #8850, and others...) ........ - -2007-06-06 13:30 +0000 [r67594-67650] Joshua Colp <jcolp@digium.com> - - * main/rtp.c, /: Merged revisions 67649 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r67649 | file | 2007-06-06 09:28:34 -0400 (Wed, 06 Jun 2007) | 2 - lines Reinvite the RTP back to the Asterisk machine when the - timeout happens. (issue #9888 reported by gasparz) ........ - - * main/translate.c: Fix plc_samples warning when registering a - translator. (issue #9897 reported by xylome) - - * apps/app_directed_pickup.c: Include macroexten while searching - for a channel to pick up in case they are in a macro. (issue - #9491 reported by jamesb63) - - * res/res_agi.c: Make the new "agi debug off" CLI command work. - (issue #9890 reported by eliel) - - * /, main/devicestate.c: Merged revisions 67593 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r67593 | file | 2007-06-06 08:18:36 -0400 (Wed, 06 Jun 2007) | 2 - lines Revert channel name splitting fix for Zap. The moral of the - story is don't use - in your user/peer names. (issue #9668 - reported by stevedavies) ........ - -2007-06-05 23:01 +0000 [r67558] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c: Fix some crashes related to the use of the - "meetme" CLI command. The code for this command was not locking - the conference list at all. (issue #9351, reported by and patch - submitted by Junk-Y, committed patch is different and by me) - -2007-06-05 21:30 +0000 [r67526] Steve Murphy <murf@digium.com> - - * pbx/ael/ael.tab.c, pbx/ael/ael.y, pbx/pbx_ael.c: this fixes bug - 9883, wherein macros were not allowing the includes construct. - fixed and tested, looks OK. Now includes can serve as an adjunct - to catch. - -2007-06-05 20:53 +0000 [r67457-67492] Russell Bryant <russell@digium.com> - - * include/asterisk/linkedlists.h: This bug has been hanging over my - head ever since I wrote this SLA code. Every time I tried to go - debug it by adding some debug output, the behavior would change. - It turns out I wasn't crazy. I had the following piece of code: - if (remove) AST_LIST_REMOVE_CURRENT(...); Well, - AST_LIST_REMOVE_CURRENT was not wrapped in braces, so my - conditional statement didn't do much good at all. It always ran - at least all of the macro minus the first statement, so I was - seeing list entries magically disappear when they weren't - supposed to. After many hours of debugging, I have come to this - extremely irritating fix. :) (issues #9581, #9497) - - * channels/chan_zap.c: Suppress a bunch of debug output unless - option_debug is on - -2007-06-05 18:32 +0000 [r67424] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Fix for bug number 9786, wherein voicemails - saved to IMAP storage using extensions other than gsm were unable - to be played over the phone. (Issue 9786, reporter: - xmarksthespot, Patched by xmarksthe spot with revisions by me, - reviewed by Russell Bryant). - -2007-06-05 18:18 +0000 [r67421] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Correctly update date/time on devices - throughout the life of the device, instead of just at - registration. Issue 9152, yet another patch by DEA. - -2007-06-05 18:17 +0000 [r67420] Steve Murphy <murf@digium.com> - - * pbx/pbx_ael.c: Added code to automatically add a default case to - switches that don't have one. In some cases, rather than fall - thru, it results in a goto with -1 result, which terminates the - extension; a sort of dialplan seqfault, sort of. This was - required to fix bug reported in 9881 - -2007-06-05 17:07 +0000 [r67360-67372] Russell Bryant <russell@digium.com> - - * main/channel.c: Handle a failure in malloc() in - ast_safe_string_alloc() - - * main/channel.c: Fix a problem that showed itself by causing Zap - channel names to be completely bogus on my machine. - ast_safe_string_alloc() was broken. It called vsnprintf() on a - va_args list twice without re-initializing it. After the first - usage, va_end() and va_start() must be called again. - -2007-06-05 16:14 +0000 [r67329-67334] Christian Richter <christian.richter@beronet.com> - - * /, channels/misdn/chan_misdn_config.h: Merged revisions 67307 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r67307 | crichter | 2007-06-05 17:42:03 +0200 (Di, 05 Jun 2007) | - 1 line briding is a bool, fixed copy and paste issue. ........ - - * channels/chan_misdn.c, /: Merged revisions 67306 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r67306 | crichter | 2007-06-05 17:39:43 +0200 (Di, 05 - Jun 2007) | 1 line simplified the EVENT_SETUP handling in the - cb_events function a lot. Commented the different possibilities a - bit and made functions of shared code. When the dialed extension - does not exist in the extensions.conf we'll jump into the 'i' - extension if this does exist, else we disconnect the call with - the cause:1 = No Route to Destination. ........ - -2007-06-05 15:51 +0000 [r67308] Russell Bryant <russell@digium.com> - - * main/asterisk.c, main/loader.c, include/asterisk/module.h: When - shutting down "gracefully", go through and run the unload() - callbacks for all of the modules. "stop now" is considered a - non-graceful shutdown and will not go through this process. - (issue #9804, reported by chrisost, patch by me) - -2007-06-05 15:22 +0000 [r67304] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Only muck with the thread structure if an - idle one was found/created. - -2007-06-05 14:35 +0000 [r67270] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_iax2.c: ensure that a burst of full frames - (AST_FRAME_DTMF being the prime example) will not be processed - out of order... this is a brute force fix, but seems to be the - safest fix for now (thanks to the Digium PQ department for - finding this bug) - -2007-06-05 10:25 +0000 [r67210] Christian Richter <christian.richter@beronet.com> - - * channels/misdn_config.c, channels/chan_misdn.c, /, - channels/misdn/chan_misdn_config.h: Merged revisions 67209 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r67209 | crichter | 2007-06-05 12:05:45 +0200 (Di, 05 Jun 2007) | - 1 line added possibility to deactivate bridging per port ........ - -2007-06-04 23:43 +0000 [r67162] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, funcs/func_math.c: Merged revisions 67161 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r67161 | tilghman | 2007-06-04 18:41:49 -0500 (Mon, 04 Jun 2007) - | 2 lines According to MATH, 0+1181000386 = 1181000448. Oops. - ........ - -2007-06-04 23:31 +0000 [r67158] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Fix up a bunch of places where the iax2 pvt - structure can disappear and the code did not account for it and - crashes. (issues #9642, #9569, #9666, probably others ... based - on the work by stevedavies and mihai, with additional changes - from me) - -2007-06-04 23:26 +0000 [r67121-67156] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Fix for skinny keepalives. If there is no - traffic from the phone for (keep_alive * 1100) ms (arbitrarily - adding 10% for network issues, etc), unregister the device. Issue - 8394, patch by DEA. - - * channels/chan_mgcp.c: Fixes for dtmf/dialing with mgcp (similar - to the recent fix for chan_skinny) Issue 9855, patch by DEA. - -2007-06-04 22:28 +0000 [r67119] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Add comments for two functions that get - called with the appropriate call locked, but perform operations - that could result in the pvt structure getting destroyed before - returning again, causing numerous seg faults all over the module. - (inspired by issues #9642, #9569, and #9666, and the work done by - stevedavies and mihai) - -2007-06-04 21:59 +0000 [r67073] Steve Murphy <murf@digium.com> - - * main/cdr.c: This typo has been here since 1.4 forked. It has been - the source of heartburn to many a dialplan/CDR programmer. - -2007-06-04 21:47 +0000 [r67071] Russell Bryant <russell@digium.com> - - * main/rtp.c: Add a missing \n. (pointed out by jcmoore on IRC) - -2007-06-04 19:31 +0000 [r67064-67068] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Better handle SIP devices that say they have - SDP content... but really don't. (issue #9398 reported by - mthomasslo) - - * apps/app_dial.c: Initialize cidname variable to nothing since it - may be used without having been touched. (issue #9661 reported by - dimas) - - * res/res_features.c: Returning a value that indicates the parking - of a call was a success when it really wasn't (because the - parking slot selected was in use) is the wrong thing to do. - (issue #9723 reported by mdu113) - -2007-06-04 17:11 +0000 [r67061] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * contrib/init.d/rc.debian.asterisk, - contrib/init.d/rc.mandrake.asterisk, /, - contrib/init.d/rc.redhat.asterisk, - contrib/init.d/rc.gentoo.asterisk, - contrib/init.d/rc.mandrake.zaptel, - contrib/init.d/rc.slackware.asterisk: Merged revisions 67060 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r67060 | tilghman | 2007-06-04 12:10:30 -0500 (Mon, 04 Jun 2007) - | 2 lines Add revision Id tags (by request of tzafrir) ........ - -2007-06-04 16:02 +0000 [r67026] Russell Bryant <russell@digium.com> - - * configure, configure.ac: Change the configure script to build a - test program against libcurl to make sure the results from - curl-config can be used to compile successfully. This is intended - to help prevent a situation where you are cross compiling, and - the configure script finds the curl library installed on the - host. (issue #9865, reported and patched by zandbelt) - -2007-06-04 15:50 +0000 [r67021] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * res/res_jabber.c: Issue 9739 - Malformed jid causes a crash - -2007-06-04 15:47 +0000 [r67018-67020] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Resolve a deadlock in chan_iax2. When - handling an implicit ACK to a frame that was marked as the final - transmission for a call, don't call iax2_destroy() for that call - while the global frame queue is still locked. There is a very - nice explanation of the deadlock in the report. (issue #9663, - thorough report and patch from stevedavies, additional positive - test reports from mihai and joff_oconnell) - - * include/asterisk/stringfields.h: Fix some compiler warnings in - C++ modules. (issue #9866, reported by osk, patch by Corydon76) - -2007-06-01 21:45 +0000 [r66919] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * funcs/func_odbc.c: On some drivers, deallocating the statement - handle isn't enough. We also have to clear the cursor (nice, - Oracle) - -2007-06-01 21:31 +0000 [r66897-66917] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Removing extraneous debugging lines from - revision 66897. Sorry :) - - * apps/app_voicemail.c: Submitting a fix for voicemail with IMAP - storage. Attachments with format specified as gsm were duplicated - (i.e. two attachments) were left. Thank you very much to - xmarksthespot for submitting the patch that fixed this. (Issues - 9787 and 8873, Reported by xmarksthespot and jerjer, patched by - xmarksthespot) - -2007-06-01 19:41 +0000 [r66879-66881] Russell Bryant <russell@digium.com> - - * channels/chan_skinny.c: Changes to the way DTMF is handled in the - core broke dialing in chan_skinny. This patch makes chan_skinny - usable again. I did not end up testing this, but there are - multiple positive test reports listed in the bug report. (issue - #9596, reported by pj, testing by pj and mvanbaak, and the fix - was written by DEA) - - * apps/app_page.c: List app_meetme as a module that app_page - depends on. - -2007-05-31 23:03 +0000 [r66821] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * doc/asterisk.8: Issue 9850 - update preferred command line syntax - -2007-05-31 18:41 +0000 [r66775] Russell Bryant <russell@digium.com> - - * res/res_speech.c, include/asterisk/app.h, - include/asterisk/speech.h: Change a couple of header files to not - use "new", which is a reserved keyword in C++. (issue #9830, - reported by osk) - -2007-05-31 17:15 +0000 [r66770] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, apps/app_macro.c: Merged revisions 66744 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r66744 | tilghman | 2007-05-31 10:58:45 -0500 (Thu, 31 May 2007) - | 2 lines Issue 9818 - Fix for issue 8329 breaks pbx_realtime. - Issue 8329 will remain unfixed for pbx_realtime, but only because - we lack core API to do it. ........ - -2007-05-31 16:14 +0000 [r66768] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 66764 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r66764 | file | 2007-05-31 12:12:39 -0400 (Thu, 31 May 2007) | 2 - lines It is now possible for this path of execution to have the - frame pointer be NULL, therefore we need to check for it before - trying to access it. (issue #9836 reported by barthpbx) ........ - -2007-05-30 23:26 +0000 [r66671] Mark Michelson <mmichelson@digium.com> - - * apps/app_voicemail.c: Fixed seg-faults when recording greetings - in voicemail with IMAP enabled. (Issue No. 9735, reported by - xmarksthespot, patched by me) - -2007-05-30 17:28 +0000 [r66602-66639] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Silly me for having out of date source! Oh - well... I'm still leaving my comment. - - * channels/chan_sip.c: When calling some peer/host that may not - exist/reply back... don't keep the dialog in memory for all of - eternity. - - * channels/chan_zap.c, channels/chan_features.c: Change how channel - names are generated a bit. (issue #9825 reported by eldadran) - -2007-05-29 21:56 +0000 [r66538] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, funcs/func_strings.c: Merged revisions 66537 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r66537 | tilghman | 2007-05-29 16:49:35 -0500 (Tue, 29 May 2007) - | 2 lines If the value of a variable passed to FIELDQTY is blank, - then FIELDQTY should return 0, not 1. ........ - -2007-05-29 19:32 +0000 [r66474-66503] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Properly handle 408 request timeout - - according to the RFC, the dialog dies if a request in a dialog - gets this response. - - * channels/chan_sip.c: Don't issue hangup on hangup on hangup on - hangup (for jcmoore) - -2007-05-29 16:44 +0000 [r66437] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Handle cases where a frame may have no data. (issue - #9519 reported by dmb) - -2007-05-29 16:07 +0000 [r66404-66414] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Don't reset hangupcause if we already have - one - - * channels/chan_sip.c: Tracking down hanging channels, killing them - one by one. Issue #9235 and related - -2007-05-29 15:43 +0000 [r66398] Joshua Colp <jcolp@digium.com> - - * doc/datastores.txt: Update datastores documentation. (issue #9801 - reported by mnicholson) - -2007-05-29 09:41 +0000 [r66363] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 66349 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r66349 | oej | 2007-05-29 09:53:14 +0200 (Tue, 29 May 2007) | 2 - lines Issue #9802 - Change inuse counter on CANCEL ........ - -2007-05-28 23:16 +0000 [r66312] Joshua Colp <jcolp@digium.com> - - * channels/chan_zap.c: Make the usedistinctiveringdetection option - work again. (issue #9823 reported by premeau) - -2007-05-27 04:12 +0000 [r66244] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c: I don't know what this was trying to do, but - it's clearly incorrect. Issues 9808 and 9809. - -2007-05-25 14:43 +0000 [r66160] Kevin P. Fleming <kpfleming@digium.com> - - * configure, configure.ac: have to check for OSP toolkit _after_ - checking for OpenSSL - -2007-05-25 14:41 +0000 [r66159] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, main/say.c: Merged revisions 66127 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r66127 | tilghman | 2007-05-25 08:46:35 -0500 (Fri, 25 May 2007) - | 2 lines Issue 9791 - Fix pronunciation of seconds in Dutch - ........ - -2007-05-25 14:28 +0000 [r66157] Kevin P. Fleming <kpfleming@digium.com> - - * configure, configure.ac, channels/chan_gtalk.c, makeopts.in, - res/res_jabber.c: handle the GNUTLS library properly in the - configure script and build system don't build in OSP support - unless we have found and are allowed to use SSL support - -2007-05-24 22:23 +0000 [r66076] Russell Bryant <russell@digium.com> - - * main/channel.c: if the string field init fails, clean up the - stuff that was allocated already - -2007-05-24 22:16 +0000 [r66074] Joshua Colp <jcolp@digium.com> - - * main/slinfactory.c: Fix slinfactory logic when dealing with - frames coming in that may already be in the signed linear format. - -2007-05-24 22:07 +0000 [r66068-66070] Russell Bryant <russell@digium.com> - - * main/channel.c: Check the result of ast_string_field_init() in - ast_channel_alloc() - - * main/rtp.c: Make 1.4 build on my machine, too.. - -2007-05-24 20:54 +0000 [r66029-66030] Jason Parker <jparker@digium.com> - - * configure: Rebuild configure script for previous ar fix. - - * configure.ac: Following moving strip to AC_PATH_TOOL, we need to - do something similar for ar. - -2007-05-24 20:42 +0000 [r65978-66026] Russell Bryant <russell@digium.com> - - * configure, include/asterisk/autoconfig.h.in, configure.ac: - Checking for the strip application needs to be done with - AC_PATH_TOOL instead of AC_PATH_PROG to properly handle cross - compilation environments. - - * Makefile: Clear CFLAGS before running make for menuselect. (issue - #9784, reported by ovi, patch by me) - -2007-05-24 18:28 +0000 [r65965-65967] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_gtalk.c: oops, use #ifdef instead of #if - - * channels/chan_gtalk.c: don't reference GnuTLS headers and - functions unless the configure script found it - - * main/rtp.c: don't use uninitialized variables - -2007-05-24 15:27 +0000 [r65902] Joshua Colp <jcolp@digium.com> - - * main/manager.c: Add the ability to blacklist certain commands - from being executed using the Command AMI action. (issue #9240 - reported by junky) - -2007-05-24 15:26 +0000 [r65892-65901] Olle Johansson <oej@edvina.net> - - * channels/chan_gtalk.c: Issue 7672 - fix by zandbelt - Asterisk - core dump since the GnuTLS interface did not support - multithreading correctly. - - * channels/chan_gtalk.c: Issue 8193 - NAT issues with gtalk/STUN. - Patch by phsultan. Thanks! - -2007-05-24 15:16 +0000 [r65877-65883] Jason Parker <jparker@digium.com> - - * .cleancount: Update cleancount for that last commit - just for - good measure. - - * include/asterisk/translate.h, codecs/codec_speex.c, - main/translate.c, codecs/codec_ilbc.c: Fix handling of - zero-length frames when a codec is capable of native PLC. Issue - 9183, patch by Mihai. - -2007-05-24 15:08 +0000 [r65866] Dwayne M. Hubbard <dhubbard@digium.com> - - * funcs/func_math.c: merged qwell's func_math patch for issue 9507 - -2007-05-24 15:08 +0000 [r65863] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: I like it when the RTP stack compiles myself... - -2007-05-24 15:05 +0000 [r65857] Olle Johansson <oej@edvina.net> - - * channels/chan_gtalk.c: Issue 7686, fix by phsultan, NAT issues - when calling from gtalk to SIP over nat. - -2007-05-24 15:04 +0000 [r65842-65853] Russell Bryant <russell@digium.com> - - * apps/app_festival.c: Ensure that frames are fully initialized. - This will probably fix getting weird timestamp log messages in - logs when using the Festival app. (issue #9781, patch by me) - - * main/rtp.c: Fix the calculation of the RTT for RTCP. The previous - code would result in oscillating and incorrect data. - Additionally, the RTT would sometimes report negative values due - to incorrect calculations. (issue #9601, patch from davetroy) - -2007-05-24 14:48 +0000 [r65841] Olle Johansson <oej@edvina.net> - - * channels/chan_gtalk.c: Issue #8536 - Caller ID not set in CDR for - jingle - -2007-05-24 14:42 +0000 [r65839] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 65837 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r65837 | file | 2007-05-24 10:40:38 -0400 (Thu, 24 May 2007) | 2 - lines Allow RFC2833 to be negotiated when an INVITE comes in - without SDP and is not matched to a user or peer. (issue #9546 - reported by mcrawford) ........ - -2007-05-24 14:38 +0000 [r65836] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c, res/res_jabber.c: Issue 8409 - phsultan - - Fix "login" as component to jabber server. ...and, by accident, - fix a bug in chan_sip for stopping a loop on retransmits of BYE - requests. - -2007-05-24 09:37 +0000 [r65768] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /: Merged revisions 65767 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r65767 | crichter | 2007-05-24 11:19:58 +0200 (Do, 24 - Mai 2007) | 1 line we should only activate the generator in - chan_misdn, when asterisk hask not yet taken the call - (WAITING4DIGS state). Alerting audio will be generated fomr - asterisk for example. ........ - -2007-05-23 20:59 +0000 [r65677-65685] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_iax2.c: start the delayed PBX when receive voice or - video full frames as well, and comment this delayed-PBX activity - - * /, channels/chan_sip.c: Merged revisions 65682 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r65682 | kpfleming | 2007-05-23 16:46:22 -0400 (Wed, 23 May 2007) - | 2 lines ensure that variables are set on a newly created - channel before we start a PBX on it ........ - - * channels/chan_iax2.c: clear the 'delay PBX' flag when we are - ready to start the PBX - - * channels/chan_iax2.c: don't start a PBX on a new incoming IAX2 - channel until we have some sort of response to our ACCEPT (ACK or - anything else) - - * /, channels/chan_iax2.c: Merged revisions 65676 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r65676 | kpfleming | 2007-05-23 16:06:13 -0400 (Wed, 23 May 2007) - | 2 lines if we are going to set variables on a newly created - channel, it should be done *before* we start the PBX on it - ........ - -2007-05-23 13:07 +0000 [r65589] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c, /: Merged revisions 65588 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r65588 | russell | 2007-05-23 08:06:17 -0500 (Wed, 23 May 2007) | - 3 lines Revert revision 62417 as someone reported problems with - it to Mark. This was related to issue #9588. ........ - -2007-05-22 20:25 +0000 [r65541] Kevin P. Fleming <kpfleming@digium.com> - - * build_tools/make_version: when building a version string for a - developer branch, include the base branch in the version string - -2007-05-22 18:40 +0000 [r65501] Russell Bryant <russell@digium.com> - - * apps/app_voicemail.c, channels/chan_zap.c: List res_smdi as a - dependency for app_voicemail and chan_zap (Thanks to mnicholson - for pointing it out) - -2007-05-22 15:04 +0000 [r65452] Joshua Colp <jcolp@digium.com> - - * apps/app_meetme.c: Remove a double const. - -2007-05-22 14:02 +0000 [r65408] BJ Weschke <bweschke@btwtech.com> - - * apps/app_followme.c: Fix a problem with flag recognition. - -2007-05-22 13:09 +0000 [r65394] Russell Bryant <russell@digium.com> - - * /, apps/app_queue.c: Merged revisions 65389 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r65389 | russell | 2007-05-22 08:07:03 -0500 (Tue, 22 May 2007) | - 4 lines Fix a memory leak that I just noticed in the device state - handling in app_queue. On most device state changes, it would - leak roughly 8 to 64 bytes (the length of the name of the - device). ........ - -2007-05-22 08:12 +0000 [r65342] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /: Merged revisions 65328 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r65328 | crichter | 2007-05-22 09:46:39 +0200 (Di, 22 - Mai 2007) | 1 line we stop the tones only when we're in the - pre-call phase, otherwise e.g. when in CONNECTED state we should - not stop tones when we receive an Information Message ........ - -2007-05-20 17:59 +0000 [r65250] Joshua Colp <jcolp@digium.com> - - * res/res_agi.c: res_agi needs to export two symbols - (ast_agi_register and ast_agi_unregister) for usage by others. - (issue #9755 reported by mnicholson) - -2007-05-18 22:26 +0000 [r65200-65201] Steve Murphy <murf@digium.com> - - * main/cdr.c: Ugh. The svnmerge didn't catch the shift from cdr.c - to main/cdr.c, and neither did I. This is the remainder of the - 9717 patch, the fix for the run-away FAIL status for a call - - * apps/app_dial.c, /, include/asterisk/cdr.h: Merged revisions - 65172 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r65172 | murf | 2007-05-18 14:56:20 -0600 (Fri, 18 May 2007) | 1 - line This update will fix the situation that occurs as described - by 9717, where when several targets are specified for a dial, if - any one them reports FAIL, the whole call gets FAIL, even though - others were ringing OK. I rearranged the priorities, so that a - new disposition, NULL, is at the lowest level, and the - disposition get init'd to NULL. Then, next up is FAIL, and next - up is BUSY, then NOANSWER, then ANSWERED. All the related set - routines will only do so if the disposition value to be set to is - greater than what's already there. This gives the intended - effect. So, if all the targets are busy, you'd get BUSY for the - call disposition. If all get BUSY, but one, and that one rings is - not answered, you get NOANSWER. If by some freak of nature, the - NULL value doesn't get overridden, then the disp2str routine will - report NOANSWER as before. ........ - -2007-05-18 18:16 +0000 [r65041-65123] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 65122 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r65122 | oej | 2007-05-18 20:10:46 +0200 (Fri, 18 May 2007) | 2 - lines Not getting an ACK to a 200 OK in the initial invite is - critical to the call. ........ - - * /, channels/chan_sip.c: Merged revisions 65075 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r65075 | oej | 2007-05-18 17:12:09 +0200 (Fri, 18 May 2007) | 5 - lines Issue 9235 - part of the problem, maybe not all. Please - retry with this patch (and no other patch) if you have problems - with hanging SIP channels. Thank you. A special Thank You to - WeBRainstorm that gave me access to his system. ........ - - * channels/chan_sip.c: - Adding support for putting calls OFF hold - with a re-invite with blank SDP. This was a bug found while doing - tests at SIPit in Antwerp. - In order to not duplicate code, I - restructured some of the code for putting calls on/off hold. - Thanks DEA for reminding me. This fix has been asleep in the - videocaps branch until now. - -2007-05-18 12:40 +0000 [r65039] Christian Richter <christian.richter@beronet.com> - - * /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: Merged - revisions 65007 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r65007 | crichter | 2007-05-18 13:23:11 +0200 (Fr, 18 Mai 2007) | - 1 line fixed a warning regarding Keypad encoding. encode the IE - sending_complete at the right position. ........ - -2007-05-18 10:37 +0000 [r64974] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue 9487 - stop media flows at hangup of - call - -2007-05-18 08:58 +0000 [r64904] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /: Merged revisions 64902 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r64902 | crichter | 2007-05-18 10:24:08 +0200 (Fr, 18 - Mai 2007) | 1 line we *need* to send a PROCEEDING when - sending_complete is set, even if need_more_infos is requested. - ........ - -2007-05-18 02:48 +0000 [r64868] Russell Bryant <russell@digium.com> - - * apps/app_queue.c: Fix a small bug I noticed while working on - something else. app_queue did not unregister its device state - monitoring callback in unload_module(). So, this would make - Asterisk crash on the first device state change after you unload - the module. - -2007-05-17 21:19 +0000 [r64820] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, include/asterisk/linkedlists.h: Merged revisions 64819 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r64819 | tilghman | 2007-05-17 16:14:36 -0500 (Thu, 17 May 2007) - | 2 lines How is it that we never caught that this is returning - the opposite of our documentation, until now? ........ - -2007-05-17 16:53 +0000 [r64761] Jason Parker <jparker@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 64758 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r64758 | qwell | 2007-05-17 11:52:38 -0500 (Thu, 17 May 2007) | 4 - lines If we have a negative current message, we shouldn't go back - even further... Issue 9727. ........ - -2007-05-17 16:52 +0000 [r64756-64759] Russell Bryant <russell@digium.com> - - * contrib/scripts/astxs (removed): Remove script that is no longer - functional since the build system was redone. (issue #9340, - reported by junky) - - * apps/app_dial.c: Increase the size of a buffer to support longer - dial strings for channels. (issue #9291, reported and fix - suggested by meni) - -2007-05-17 16:10 +0000 [r64720-64754] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Even more direct RTP setup fixes! Don't - allow a codec that isn't supported to creep into the SDP of - either side. (issue #9446 reported by marcelbarbulescu) - - * apps/app_voicemail.c: Fix authuser support. (issue #9740 reported - by xmarksthespot) - -2007-05-17 06:13 +0000 [r64686] Russell Bryant <russell@digium.com> - - * README: Update the main README to reflect the new build process - for 1.4 and above. (issue #9725, patch by eliel) - -2007-05-16 11:01 +0000 [r64516-64609] Olle Johansson <oej@edvina.net> - - * /: Blocking patch already in this code - - * channels/chan_sip.c: Fix auth on BYE. (Different patch than for - 1.2) - - * channels/chan_sip.c: Issue #9681 - Handle www-auth on BYE - - * channels/chan_sip.c: Final part of issue #9483 - fixing - transfer() of sip calls in the dial plan (twilson) - - * channels/chan_sip.c: Issue #9439 - properly handle username - parameters in SIP uri. - - * /, channels/chan_sip.c: Merged revisions 64535 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r64535 | oej | 2007-05-16 11:08:22 +0200 (Wed, 16 May 2007) | 2 - lines Support SIP uri's starting with SIP: and sip: (reported by - Tony Mountfield on the mailing list. Thanks!) ........ - - * /, channels/chan_sip.c: Merged following patch with a lot of - changes for 1.4 ------ Merged revisions 64514 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r64514 | oej | 2007-05-16 10:25:56 +0200 (Wed, 16 May 2007) | 6 - lines Issue #9726 - rlister - Better logging for ACL denials - While at it, also added better logging and handling of peers that - are not supposed to register. My patch, stole the issue report - from Russell. My apologies, Russell :-) ........ - -2007-05-16 08:44 +0000 [r64515] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /: Merged revisions 64513 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r64513 | crichter | 2007-05-16 10:23:42 +0200 (Mi, 16 - Mai 2007) | 1 line in the case immediate=yes, we directly jump - into the dialplan, where people can use PlayTones to indicate a - Dialtone, so we don't need to to that by ourself. also we should - not do a dialtone_indicate for incoming calls on a TE port in - overlapdialmode. ........ - -2007-05-15 19:52 +0000 [r64353-64426] Russell Bryant <russell@digium.com> - - * res/res_features.c: Properly fix a problem that occurs when you - set PARKINGEXTEN to an exten where a call is already parked. - (issue #9723, patch by me) - - * res/res_features.c: When someone requests a specific parking - space using the PARKINGEXTEN variable, ensure that no other - caller is already there. (issue #9723, reported by mdu113, patch - by me) - -2007-05-14 19:26 +0000 [r64324] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Change -2 to XMIT_ERROR to clarify a bit - more - -2007-05-14 19:13 +0000 [r64306] Russell Bryant <russell@digium.com> - - * channels/chan_alsa.c: Properly handle AST_CONTROL_PROGRESS by - just ignoring it. An unknown indication will trigger an error and - cause sounds to stop, which in this case, is ringing. - -2007-05-14 18:52 +0000 [r64280] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Handle network errors, like host or network - unreachable, in a better way. This means that calls to hosts or - qualify (OPTION) messages will fail quicker if the TCP/IP stack - tells us that there is an issue. Since this is an unconnected UDP - socket, we will not get error messages directly in most cases, - but maybe on the second and third try. This is already - implemented in trunk. - -2007-05-14 18:48 +0000 [r64240-64278] Joshua Colp <jcolp@digium.com> - - * codecs/codec_speex.c: Properly set datalen field when doing PLC - in codec_speex. (issue #9722 reported by mihai) - - * /, main/devicestate.c: Merged revisions 64275 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r64275 | file | 2007-05-14 14:34:06 -0400 (Mon, 14 May 2007) | 2 - lines Only perform stripping of - strings from the channel name - for Zap channels. Anywhere else we might remove a legitimate part - of a device name. (issue #9668 reported by stevedavies) ........ - - * main/channel.c: Fix scenario where if a phone that simply called - Echo() put itself on hold it could never get off hold. - -2007-05-14 13:58 +0000 [r64193] Steve Murphy <murf@digium.com> - - * main/cdr.c, main/pbx.c, channels/chan_local.c: As per 9570, - worrisome CDR warnings have been removed, that are either not - helpful, or not relevant. - -2007-05-14 10:39 +0000 [r64157] Olle Johansson <oej@edvina.net> - - * main/channel.c: Add hangupcause when we lack codecs for - transcoding - -2007-05-12 22:27 +0000 [r64044-64114] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: This concludes my final adventure with - bitmasks and the onhold flag. Would anyone care for some peanuts? - - * channels/chan_sip.c: Tweak hold flags some more. They can be of - three states when active: active, inactive, one direction. - - * channels/chan_sip.c: Ensure the onhold flag is set no matter what - when being put on hold. - -2007-05-11 20:16 +0000 [r63982] Jason Parker <jparker@digium.com> - - * main/manager.c: Hide manager password from "manager show user - foo". I realize that there are other ways to get this, but we - really don't need to just show it in plain text so easily. Issue - 9273, patch by junky - -2007-05-11 16:35 +0000 [r63905] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * contrib/scripts/safe_asterisk, Makefile, /: Merged revisions - 63903 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r63903 | tilghman | 2007-05-11 11:31:03 -0500 (Fri, 11 May 2007) - | 2 lines Issue 9121 - fixups for safe_asterisk script ........ - -2007-05-11 16:05 +0000 [r63886] Russell Bryant <russell@digium.com> - - * main/manager.c: When MD5 authentication is not possible because - there is no challenge present, either because the Challenge - action was never issued, or some other reason, give a proper - error message and return an error instead of claiming that the - user wasn't found. (reported by jsmith on IRC) - -2007-05-11 15:43 +0000 [r63872] Joshua Colp <jcolp@digium.com> - - * res/res_features.c: Make the PARKINGEXTEN feature of parking - actually work. (issue #9708 reported by mdu113) - -2007-05-10 23:15 +0000 [r63830] Jason Parker <jparker@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 63828 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r63828 | qwell | 2007-05-10 18:14:55 -0500 (Thu, 10 May 2007) | 4 - lines Fix an issue with trying to kill a thread before it gets - created. Issue 9709, patch by nic_bellamy. ........ - -2007-05-10 22:23 +0000 [r63804] Russell Bryant <russell@digium.com> - - * main/manager.c: Strip terminal escape sequences from CLI command - output that is going to be sent out over the manager interface. - (issue #9659, reported by pari, fixed by me) - -2007-05-10 20:48 +0000 [r63750] Doug Bailey <dbailey@digium.com> - - * main/callerid.c: Add test for negative offsets in cid data to - prevent infinite loops. - -2007-05-10 20:46 +0000 [r63749] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 63748 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r63748 | oej | 2007-05-10 22:38:54 +0200 (Thu, 10 May 2007) | 4 - lines Do not allocate SIP pvt's for PEERs we can not reach. This - was seen as a lot of dialogs being created then immediately - destroyed at reload/restart of the SIP channel. ........ - -2007-05-09 19:22 +0000 [r63656-63698] Joshua Colp <jcolp@digium.com> - - * main/channel.c: Use the DTMF frame on the channel when returning - a DTMF frame from AST_FRAME_NULL or AST_FRAME_VOICE. - - * channels/chan_sip.c: Do not prematurely go on hold if sendonly - was not actually set. - -2007-05-09 17:25 +0000 [r63654] Matthew Fredrickson <creslin@digium.com> - - * channels/chan_zap.c, /: Merged revisions 63653 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r63653 | mattf | 2007-05-09 12:20:20 -0500 (Wed, 09 May 2007) | 2 - lines Make sure we only create a DSP if it's requested on - SUB_REAL ........ - -2007-05-09 16:55 +0000 [r63612] Russell Bryant <russell@digium.com> - - * main/channel.c: Modify ast_senddigit_begin() to use the same - assumptions used elsewhere in the code in that if a channel does - not have a send_digit_begin() callback, it only cares about DTMF - END events. (pointed out by Michael Neuhauser on the asterisk-dev - list) - -2007-05-09 16:54 +0000 [r63611] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 63610 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r63610 | file | 2007-05-09 12:51:03 -0400 (Wed, 09 May 2007) | 2 - lines Properly handle hints that point to multiple devices in - chan_sip. Why chan_sip is even doing this I have no idea but I - would rather not go into a rant. (issue #9536 reported by - rlister) ........ - -2007-05-09 16:43 +0000 [r63608] Russell Bryant <russell@digium.com> - - * main/channel.c: Only call ast_senddigit_begin() in - ast_senddigit() if the channel has a send_digit_begin() callback. - Checking the END_DTMF_ONLY flag was the wrong thing to do, - because that flag indicates that a *bridged* channel only wants - DTMF END events coming from this channel. - -2007-05-09 14:50 +0000 [r63566] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, apps/app_directory.c: Merged revisions 63565 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r63565 | tilghman | 2007-05-09 09:48:06 -0500 (Wed, 09 May 2007) - | 2 lines Replicate fix from 51158 (app_voicemail) to - app_directory (Issue 9224) ........ - -2007-05-09 13:24 +0000 [r63535] Russell Bryant <russell@digium.com> - - * Makefile: I have seen multiple people post questions trying to - figure out what the message "The configure script must be - executed before running 'make'" means. So, add another like that - says to specifically run ./configure. If this isn't obvious - enough, then they should be using something like AsteriskNOW and - not installing from source. - -2007-05-09 13:17 +0000 [r63534] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /, - channels/misdn/isdn_msg_parser.c: Merged revisions - 62945,63402,63519 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r62945 | crichter | 2007-05-03 17:39:21 +0200 (Do, 03 Mai 2007) | - 1 line when we're in state WAITING4DIGS, we use the asterisk - tone-generator which prods us, so we can't just return -1 in - misdn_write in this case. Added a MISDN_KEYPAD channel variable, - and fixed the sending of keypad. this enables us to modify the - call forward parameters in the switch. ........ r63402 | crichter - | 2007-05-08 17:07:37 +0200 (Di, 08 Mai 2007) | 1 line added - application misdn_check_l2l1 which tries to pull up the L1/L2 on - all ports that have the layers down in a group. It waits then for - a timeout. This helps for scenarios where multiple PMP BRIs are - grouped together, or where a provider has a faulty PTP - Implementation, that looses the L2 after a while. ........ r63519 - | crichter | 2007-05-09 13:26:16 +0200 (Mi, 09 Mai 2007) | 1 line - release_chan frees ch, so we should never touch ch after - release_chan, this may cause segfaults. ........ - -2007-05-09 13:04 +0000 [r63532] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Don't retransmit 200 OK's on ignore status. - (Reported on asterisk-users) - -2007-05-08 22:38 +0000 [r63478] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, apps/app_macro.c: Merged revisions 63477 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r63477 | tilghman | 2007-05-08 17:19:15 -0500 (Tue, 08 May 2007) - | 2 lines Issue 9602 - segfault in app_macro ........ - -2007-05-08 16:53 +0000 [r63403-63448] Russell Bryant <russell@digium.com> - - * res/res_features.c: I mixed up the use of the find_feature() - function, so I renamed it find_dynamic_feature, and changed the - code to use the correct lock when using it. - - * res/res_features.c: Use a read/write lock when accessing the - built-in features. - - * contrib/scripts/realtime_pgsql.sql (added), - contrib/realtime_pgsql.sql (removed): Move realtime_pgsql.sql to - contrib/scripts to be with the rest of the sql examples. (issue - #9676, suretec) - -2007-05-08 06:22 +0000 [r63360] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c, /: Merged revisions 63359 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r63359 | tilghman | 2007-05-08 01:20:16 -0500 (Tue, 08 May 2007) - | 2 lines Issue 9527 - upon entering a folder, no message is - selected (curmsg == -1), so deleting causes memory corruption - (beyond bounds) ........ - -2007-05-07 22:28 +0000 [r63329] Russell Bryant <russell@digium.com> - - * configs/res_pgsql.conf.sample (added), - configs/extconfig.conf.sample, contrib/realtime_pgsql.sql - (added): Add a sample configuration file and example tables for - use with res_config_pgsql. (issue #9676, suretec) - -2007-05-07 21:45 +0000 [r63283-63286] Joshua Colp <jcolp@digium.com> - - * main/channel.c, include/asterisk/app.h, /, main/app.c: Merged - revisions 63285 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r63285 | file | 2007-05-07 17:39:52 -0400 (Mon, 07 May 2007) | 2 - lines Properly handle what happens during a masquerade in - relation to group counting. (issue #9657 reported by ramonpeek) - ........ - - * channels/chan_sip.c: Minor backport of revision 59083 in trunk. - Don't queue an unhold frame up if the call was never on hold to - begin with. - -2007-05-07 20:05 +0000 [r63196-63254] Olle Johansson <oej@edvina.net> - - * main/config.c: Don't remove configuration from memory just - because one section failed. - - * /: Guess svnmerge doesn't handle files that move around. Blocking - patch to ./config.c - -2007-05-06 12:28 +0000 [r63152] Olle Johansson <oej@edvina.net> - - * main/file.c: Stop the video stream when you stop playback of all - streams for a call - -2007-05-04 20:03 +0000 [r63099] Jason Parker <jparker@digium.com> - - * res/res_jabber.c: Fix a crash when checking version attribute in - an incoming XML caps element. Issue 9667, patch by phsultan. - -2007-05-04 16:45 +0000 [r63047] Pari Nannapaneni <paripurnachand@digium.com> - - * configs/manager.conf.sample: explanation for httptimeout in - manager.conf - -2007-05-03 16:44 +0000 [r62989] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 62987 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r62987 | file | 2007-05-03 13:42:19 -0300 (Thu, 03 May 2007) | 2 - lines When a peer is seeded or built tell the devicestate core to - update it's status. This is easier then having chan_sip load - before pbx_config. (issue #9658 reported by dlynes) ........ - -2007-05-03 16:38 +0000 [r62986] Kevin P. Fleming <kpfleming@digium.com> - - * main/loader.c: improve loader a bit, by avoiding trying to - initialize embedded modules twice and avoiding trying to load - modules from disk when they have been loaded already during the - 'preload' pass (reported by blitzrage on IRC, patch by me) - -2007-05-03 15:23 +0000 [r62942] Russell Bryant <russell@digium.com> - - * main/channel.c: Fix YADB (Yet Another DTMF Bug) ((C) Russell - Bryant, 2007, TM, Patent Pending). This set of changes came from - a debugging session I had with Dwayne Hubbard. When he called - into his home FXO, ran the Echo application, and pressed a digit, - the digit would be echoed back and would never end. This is - fixed, along with a couple other little improvements. * When - chan_zap is in the middle of playing a digit to a channel, it - feeds back null frames, not voice frames. So, I have modified - ast_read to check the timing on emulated DTMF when it receives - null frames, in addition to where it was doing this on voice - frames. * Make a tweak to setting the duration on emulated DTMF - digits. If there was no duration specified, it set it to be the - minimum, instead of the default. * Instead of timing the emulated - digits off of the number of samples in audio frames that pass - through, just use time values. Now there is no code in this - section that assumes 8kHz audio. - -2007-05-03 14:41 +0000 [r62913] Steve Murphy <murf@digium.com> - - * pbx/ael/ael-test/ref.ael-test18, pbx/ael/ael-test/ref.ael-test19 - (added), pbx/ael/ael-test/ael-test18/extensions.ael, - pbx/ael/ael-test/ael-test19/extensions.ael (added), - pbx/ael/ael-test/ael-test19 (added), - pbx/ael/ael-test/ref.ael-test20 (added), - pbx/ael/ael-test/ael-test20/extensions.ael (added), - pbx/ael/ael-test/ael-test20 (added): updated the ael regressions - to match what's in trunk - -2007-05-03 14:36 +0000 [r62912] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h, - channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, - channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c: Merged - revisions 61357,61770,62885 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61357 | crichter | 2007-04-11 14:05:57 +0200 (Mi, 11 Apr 2007) | - 1 line some fixes for PMP Hold/Retrieve, it should work now, when - briding=no ........ r61770 | crichter | 2007-04-24 15:50:05 +0200 - (Di, 24 Apr 2007) | 1 line added lock for sending messages to - avoid double sending. shuffled some empty_chans after the - cb_event calls, this avoids that a release_complete from a quite - different call releases a fresh created setup by accident. - ........ r62885 | crichter | 2007-05-03 15:59:00 +0200 (Do, 03 - Mai 2007) | 1 line fixed the problem that misdn_write did not - return -1 when called with 0 samples in a frame this resultet in - a deadlock in some circumstances, when the call ended because of - a busy extension. added encoding of keypad. ........ - -2007-05-03 13:54 +0000 [r62883] Steve Murphy <murf@digium.com> - - * pbx/ael/ael-test/ref.ael-test18 (added), - pbx/ael/ael-test/ref.ael-vtest13, - pbx/ael/ael-test/ael-test18/extensions.ael (added), - pbx/ael/ael-test/ael-test18 (added), - pbx/ael/ael-test/ref.ael-vtest17, pbx/ael/ael.tab.c, - pbx/ael/ael.y, pbx/ael/ael.tab.h, pbx/ael/ael-test/ref.ael-test7: - These mods fix bug 9623, where an '@' in the eswitch contents - causes a syntax error. I also updated the regressions. - -2007-05-03 00:23 +0000 [r62797-62842] Kevin P. Fleming <kpfleming@digium.com> - - * res/res_config_odbc.c, /: Merged revisions 62841 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r62841 | kpfleming | 2007-05-02 20:23:00 -0400 (Wed, 02 - May 2007) | 2 lines doh... initializing the pointer variable will - work just a bit better ........ - - * res/res_config_odbc.c, /: Merged revisions 62796 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r62796 | kpfleming | 2007-05-02 19:53:46 -0400 (Wed, 02 - May 2007) | 7 lines increase reliability and efficiency of static - Realtime config loading via ODBC: don't request fields we aren't - going to use don't request sorting on fields that are pointless - to sort on explicitly request the fields we want, because we - can't expect the database to always return them in the order they - were created (reported by blitzrage in person (!), patch by me) - ........ - - * res/res_config_pgsql.c: improve static Realtime config loading - from PostgreSQL: don't request sorting on fields that are - pointless to sort on use ast_build_string() instead of snprintf() - don't request the list of fieldnames that resulted from the query - when we both knew what they were before we ran the query _AND_ we - aren't going to do anything with them anyway (patch by me, - inspired by blitzrage's bug report about res_config_odbc) - -2007-05-02 22:59 +0000 [r62739-62789] Russell Bryant <russell@digium.com> - - * main/channel.c: Merge changes from team/russell/inband_dtmf ... - Fix some issues related to generating inband DTMF. There are two - changes here: 1) The list of DTMF tones in the senddigit_begin() - function explicitly specified 100ms of the tone followed by 100ms - of silence. This really broke things with the way that Asterisk - now wants complete control over when the digit begins and ends. - So, regardless of what Asterisk really wanted to do, this was - going to play out the tone at the length it wanted to. This - caused various problems like DTMF translation to inband to be - extremely unreliable. The list of tones has been changed so that - the correct DTMF tone is played indefinitely until Asterisk tells - it to stop. 2) ast_write() had to be modified to let a DTMF_END - frame get processed even when a generator is present. This is how - the tone will finally get stopped. (issues #8944, #9250, #9348, - maybe others. Thanks to mdu113 from #8944 for the testing and - feedback!) - - * main/manager.c: Backport the change that only went in to trunk - that fixes the command manager action over http. (reported - internally by pari and bkruse) - -2007-05-02 20:46 +0000 [r62738] Steve Murphy <murf@digium.com> - - * main/cdr.c, main/pbx.c, /: Merged revisions 62737 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r62737 | murf | 2007-05-02 14:10:32 -0600 (Wed, 02 May - 2007) | 1 line Some tweaks to satisfy CDR bug 8796, where being - in 'h' extension louses up the dst field ........ - -2007-05-02 17:43 +0000 [r62692] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, channels/chan_iax2.c: Merged revisions 62691 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r62691 | tilghman | 2007-05-02 12:38:16 -0500 (Wed, 02 May 2007) - | 4 lines Issue 9638 - if a text frame is sent with no - terminating NULL through a bridged IAX connection, the remote end - will receive garbage characters tacked onto the end. ........ - -2007-05-02 17:10 +0000 [r62689] Steve Murphy <murf@digium.com> - - * configs/extensions.conf.sample, main/channel.c, main/pbx.c, - channels/chan_zap.c, cdr/cdr_radius.c: a)In chan_zap, set the - clid, src fields in channel_alloc call. b)in the channel_alloc - func, set the cid_num and name fields from the arglist[blush]. c) - don't update the channel app & app data fields if you are in the - 'h' extension. d)the load_module func in cdr_radius needs to - return DECLINE, SUCCESS. - -2007-05-02 06:15 +0000 [r62624] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Don't unlock a channel that we already know - does not exist (propably isue 8228) - -2007-05-01 21:57 +0000 [r62548] Russell Bryant <russell@digium.com> - - * /, res/res_features.c: Merged revisions 62547 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r62547 | russell | 2007-05-01 16:55:19 -0500 (Tue, 01 May 2007) | - 4 lines Remove an unnecessary check that makes it so if you hang - up after doing an attended transfer before the target extension - answers the channel, the transfer is not successful. (issue - #9338, patch by svanlund) ........ - -2007-05-01 21:34 +0000 [r62545] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Bug 9590 - Memory leaks around find_user() - (found by rayjay, different fixes by me) - -2007-05-01 16:26 +0000 [r62497] Russell Bryant <russell@digium.com> - - * /, configs/indications.conf.sample: Merged revisions 62496 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r62496 | russell | 2007-05-01 11:26:23 -0500 (Tue, 01 May 2007) | - 3 lines Add indications.conf information for the Philippines. - (issue #9525, reported and patched by loloski) ........ - -2007-04-30 15:58 +0000 [r62414-62419] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c, /: Merged revisions 62417 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r62417 | russell | 2007-04-30 10:57:26 -0500 (Mon, 30 Apr 2007) | - 4 lines This patch fixes an issue where depending on the cause - code, when the network sends a PRI disconnect, the call may not - be properly hung up. (issue #9588, reported and patched by - softins) ........ - - * include/asterisk/http.h, main/http.c: When serving dynamic - content, include a Cache-Control header to instruct the browsers - to not store the resulting content. (issue #9621, reported by - Pari, patch by me) - -2007-04-30 14:52 +0000 [r62371] Jason Parker <jparker@digium.com> - - * configs/iax.conf.sample: Remove unused (and potentially - confusing) jitterbuffer options from sample config. - -2007-04-30 14:36 +0000 [r62369] Joshua Colp <jcolp@digium.com> - - * main/asterisk.c, /: Merged revisions 62368 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r62368 | file | 2007-04-30 11:34:07 -0300 (Mon, 30 Apr 2007) | 2 - lines Update copyright notice. It's now the year 2007! ........ - -2007-04-29 05:50 +0000 [r62299-62331] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c: Fix a bug that made the "language" setting - in zapata.conf not functional. (issue #9626, reported and fixed - by sergee) - - * apps/app_meetme.c: Note that the "talker optimization" option - will be enabled by default in 1.6 - -2007-04-27 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.4 released. - -2007-04-27 21:10 +0000 [r62218] Russell Bryant <russell@digium.com> - - * channels/chan_agent.c: Fix a weird problem where when a caller - talking to someone sitting behind an agent channel sent a digit, - the digit would be played to the agent for forever. This is - because chan_agent always returned -1 from its send_digit_begin - and _end callbacks. This non-zero return value indicates to the - Asterisk core that it would like an inband DTMF generator put on - the channel. However, this is the wrong thing to do. It should - *always* return 0, instead. When the digit begin and end - functions are called on the proxied channel, the underlying - channel will indicate whether inband DTMF is needed or not, and - the generator will be put on that one, and not the Agent channel. - (issue #9615, #9616, reported by jiddings and BigJimmy, and fixed - by me) - -2007-04-27 16:17 +0000 [r62174] Jason Parker <jparker@digium.com> - - * /, codecs/codec_zap.c: Merged revisions 62173 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r62173 | qwell | 2007-04-27 11:16:16 -0500 (Fri, 27 Apr 2007) | 3 - lines This transcoder message needn't be a NOTICE. I've seen it - cause confusion more than a few times. ........ - -2007-04-27 16:14 +0000 [r62171] Russell Bryant <russell@digium.com> - - * main/pbx.c: If no variables were passed into - pbx_substitute_variables_helper_full(), then don't even bother - creating a temporary bogus channel, since that is only for - allowing certain functions to operate on the variables as if they - were on a channel. Most importantly, this fixes a crash. (issue - #9613, reported by callguy, fixed by me) - -2007-04-27 14:04 +0000 [r62095-62137] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 62126 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r62126 | oej | 2007-04-27 15:57:45 +0200 (Fri, 27 Apr 2007) | 4 - lines Issue #7351 - SIP Cancel fails due to the wrong contact - uri. Reported by PPYY, failed to fix by OEJ final fix by wojtekka - - THANKS!!!! THis was a hard one to catch. ........ - - * channels/chan_zap.c, main/manager.c: Issue #9608 - fix some - annoying DEBUG messages not controlled by option_debug (DEA). - Thanks! - -2007-04-26 16:33 +0000 [r61959-62038] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 62037 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r62037 | file | 2007-04-26 12:30:57 -0400 (Thu, 26 Apr 2007) | 2 - lines Revert previous fix for when the IAX2 channel goes funky - (that's the technical term). This is causing legit calls to be - prematurely hung up. (issue #9600 reported by justdave) ........ - - * main/channel.c: Missed an ast_app_group_discard during merge. - Thanks blitzrage! - - * res/res_monitor.c: Don't always say that the channel is being - paused if it is actually being unpaused in the Manager ack - message. (reported by jsmith in #asterisk-bugs) - - * main/config.c, /: Merged revisions 61958 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61958 | file | 2007-04-25 21:25:03 -0400 (Wed, 25 Apr 2007) | 2 - lines Don't count failed include attempts against the - configuration include level. (issue #9593 reported by mostyn) - ........ - -2007-04-25 22:29 +0000 [r61914] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c, /: Merged revisions 61913 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61913 | kpfleming | 2007-04-25 17:24:59 -0500 (Wed, 25 Apr 2007) - | 2 lines handle a very bizarre race condition with channels - being redirected before a simple switch can be started on them - (issue #9286) ........ - -2007-04-25 21:59 +0000 [r61863-61870] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 61866 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61866 | russell | 2007-04-25 16:55:23 -0500 (Wed, 25 Apr 2007) | - 2 lines If the callerid= option is specified, but empty, clear - any previous data. ........ - - * /, channels/chan_iax2.c: Merged revisions 61862 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61862 | russell | 2007-04-25 16:06:22 -0500 (Wed, 25 Apr 2007) | - 2 lines Ensure that callerid settings are reset on a reload. - ........ - -2007-04-25 19:21 +0000 [r61805] Joshua Colp <jcolp@digium.com> - - * main/cli.c, main/channel.c, include/asterisk/app.h, - funcs/func_groupcount.c, /, main/app.c: Merged revisions 61804 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61804 | file | 2007-04-25 14:52:50 -0400 (Wed, 25 Apr 2007) | 2 - lines Merge rewritten group counting support. No more storing - data on the variable list of the channels. That was bad, mmmk? - (issue #7497 reported by sabbathbh) ........ - -2007-04-25 16:22 +0000 [r61799] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c, /: Merged revisions 61798 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61798 | russell | 2007-04-25 11:20:38 -0500 (Wed, 25 Apr 2007) | - 3 lines Fix a typo where cid_num got copied instead of cid_ani. - (issue #9587, reported and patched by xrg) ........ - -2007-04-24 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.3 released. - -2007-04-24 21:34 +0000 [r61781-61787] Russell Bryant <russell@digium.com> - - * main/manager.c, /: Merged revisions 61786 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61786 | russell | 2007-04-24 16:33:59 -0500 (Tue, 24 Apr 2007) | - 4 lines Don't crash if a manager connection provides a username - that exists in manager.conf but does not have a password, and - also requests MD5 authentication. (ASA-2007-012) ........ - - * main/channel.c, include/asterisk/channel.h: Improve DTMF handling - in ast_read() even more in response to a discussion on the - asterisk-dev mailing list. I changed the enforced minimum length - of a digit from 100ms to 80ms. Furthermore, I made it now enforce - a gap of 45ms in between digits. These values are not - configurable in a configuration file right now, but they can be - easily changed near the top of main/channel.c. - -2007-04-24 18:43 +0000 [r61779] Dwayne M. Hubbard <dhubbard@digium.com> - - * channels/chan_zap.c, /: Merged revisions 61777 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61777 | dhubbard | 2007-04-24 13:20:31 -0500 (Tue, 24 Apr 2007) - | 1 line removed #if 0 block from chan_phone, chan_zap, and - chan_modem restart_monitor() ........ - -2007-04-24 16:16 +0000 [r61774] Russell Bryant <russell@digium.com> - - * main/dial.c: Add a few more state changes in - handle_frame_ownerless() so that the SLA code will get notified - of these changes even when an owner channel is not provided. This - isn't from a specific bug report, it's just something I noticed - while poking around. - -2007-04-24 16:07 +0000 [r61772] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 61771 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61771 | file | 2007-04-24 12:05:06 -0400 (Tue, 24 Apr 2007) | 2 - lines Allow RFC2833 to be sent in the response SDP when an INVITE - comes in without SDP. (issue #9546 reported by mcrawford) - ........ - -2007-04-23 18:17 +0000 [r61763-61765] Russell Bryant <russell@digium.com> - - * main/pbx.c: Some dialplan functions, such as CUT(), expect to - operate on variables on a channel. So, this little hack lets them - work in places where a channel doesn't exist, such as within - DUNDi configuration. (issue #9465, reported and patched by - Corydon76, testing by blitzrage) - - * main/channel.c: Ensure that digits passing through Asterisk have - a reasonable minimum length. It is currently 100 ms. If someone - thinks this should be different, feel free to speak up. (related - to issues #8944, #9250, and #9348) - -2007-04-20 21:35 +0000 [r61705-61707] Jason Parker <jparker@digium.com> - - * main/rtp.c: Avoid invalid seqno cycling detection. Per comment - from Dave Troy: This adds back in some simple typecasting I had - in an earlier version which I realize now may be breaking things. - Issue #9554. - - * main/loader.c, /: Merged revisions 61704 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61704 | qwell | 2007-04-20 16:14:27 -0500 (Fri, 20 Apr 2007) | 4 - lines Fix an issue that I noticed while looking over issue 9571. - The reload timestamp was getting set after reloading the built-in - stuff, and before the modules. ........ - -2007-04-20 20:42 +0000 [r61697] Russell Bryant <russell@digium.com> - - * main/rtp.c: Remove a stray debug message introduced by a recent - commit. - -2007-04-20 19:51 +0000 [r61694] Jason Parker <jparker@digium.com> - - * /, apps/app_queue.c: Merged revisions 61692 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61692 | qwell | 2007-04-20 14:49:54 -0500 (Fri, 20 Apr 2007) | 5 - lines If the '* to hangup' option is not enabled, we don't need - to disable * as a valid exit key. If it was enabled, this - statement would've never been checked in the first place. Issue - #9552 ........ - -2007-04-20 18:19 +0000 [r61690] Russell Bryant <russell@digium.com> - - * main/config.c, apps/app_voicemail.c, main/manager.c, - include/asterisk/config.h: Fix the UpdateConfig manager action to - properly treat "variables" and "objects" differently (a=b versus - a=>b). (issue #9568, reported by pari, patch by me) - -2007-04-19 08:37 +0000 [r61686] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 61685 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61685 | oej | 2007-04-19 09:56:21 +0200 (Thu, 19 Apr 2007) | 3 - lines Send NOTIFY to Contact: in SUBSCRIBE - as reported by - Intertex and Citel. Fixed during SIPit 20 in Antwerp. ........ - -2007-04-19 04:36 +0000 [r61681-61683] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * main/manager.c: Bug 9557 - simple reason why reading a function - always returned NULL - - * funcs/func_callerid.c, funcs/func_language.c, funcs/func_moh.c, - funcs/func_groupcount.c, /, funcs/func_timeout.c, - funcs/func_cdr.c: Merged revisions 61680 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61680 | tilghman | 2007-04-18 21:30:18 -0500 (Wed, 18 Apr 2007) - | 5 lines Bug 9557 - Specifying the GetVar AMI action without a - Channel parameter can cause Asterisk to crash. The reason this - needs to be fixed in the functions instead of in AMI is because - Channel can legitimately be NULL, such as when retrieving global - variables. ........ - -2007-04-18 22:10 +0000 [r61678] Kevin P. Fleming <kpfleming@digium.com> - - * sounds/Makefile: allow external build systems to extract the - required sound file versions - -2007-04-18 20:46 +0000 [r61674-61676] Olle Johansson <oej@edvina.net> - - * main/rtp.c: Clean upp formatting, add some doxygen stuff while - we're in cleaning mode... Thanks Kevin! - - * main/rtp.c: Issue #9554 - Improve RTCP (Dave Troy) - -2007-04-16 14:47 +0000 [r61664-61666] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: #9483, half of patch by twilson to solve 302 - redirect issues - - * /: Blocking AstHoloPatch from 1.2 - -2007-04-13 21:17 +0000 [r61658] Steve Murphy <murf@digium.com> - - * main/cdr.c: This is a fix to the way CDR merge handles the data - that results from ForkCDR. - -2007-04-13 19:17 +0000 [r61648-61656] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c, /: Merged revisions 61655 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61655 | file | 2007-04-13 15:15:12 -0400 (Fri, 13 Apr 2007) | 2 - lines Add OUTBOUND_GROUP_ONCE variable to app_dial. This behaves - the same as OUTBOUND_GROUP except it will get unset after use so - it won't get accidentally inherited. (issue #BE-140) ........ - - * apps/app_speech_utils.c: Do not bother looking for a result if - none are present. - - * channels/chan_sip.c: For those very verbose SIP implementations - that attach tons of info to the Contact header... let's increase - our variable sizes. (issue #9535 reported by jeffg) - -2007-04-13 17:10 +0000 [r61645] Russell Bryant <russell@digium.com> - - * apps/app_voicemail.c: Eliminate a compiler warning with - ODBC_STORAGE enabled so that it will build under dev-mode. - -2007-04-13 17:01 +0000 [r61644] Steve Murphy <murf@digium.com> - - * channels/chan_oss.c: A fix for chan_oss that resulted from the - CDR changes; it helps to use the right info. - -2007-04-13 16:32 +0000 [r61641] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Don't assume the callid of a dialog will be - set, as in some circumstances it may not. (issue #9534 reported - by tecnoxarxa) - -2007-04-11 16:05 +0000 [r61477] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 61476 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61476 | russell | 2007-04-11 11:01:25 -0500 (Wed, 11 Apr 2007) | - 5 lines If someone sets the "useragent" option in sip.conf to be - empty, then don't add the User-Agent header at all. It is an - optional header, anyway. Also, the bug report says that some of - Japan's SIP providers don't allow it for some weird reason. - (issue #9488, reported by makoto, fixed by me) ........ - -2007-04-11 15:39 +0000 [r61443] Nadi Sarrar <ns@beronet.com> - - * channels/chan_misdn.c: Don't export AOCD variables on - misdn_hangup anymore, this was mainly a fix for trunk.. - -2007-04-11 15:09 +0000 [r61377-61427] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 61426 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61426 | russell | 2007-04-11 10:05:36 -0500 (Wed, 11 Apr 2007) | - 6 lines Fix a bug with switching between host=dynamic and using - specific hosts for peers. The code would only reset the peer's - address when it is dynamic if it was a new peer structure. Now, - it will also reset the address if it was already in the peer - list, but before the reload, it was not dynamic. (issue #9515, - reported by caio1982, fixed by me) ........ - - * main/http.c: Add "svgz" to the mimetypes table. (issue #9510, - bkruse) In passing, constify the elements of the mimetypes table. - - * /, channels/chan_sip.c: Merged revisions 61376 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61376 | russell | 2007-04-11 09:02:54 -0500 (Wed, 11 Apr 2007) | - 5 lines Remove the attempt at reporting configuration errors in - sip.conf. This can cause a bunch of improper messages when using - realtime. I give up. As oej tried to convince me when I put this - in, there is just no easy way to do it. (inspired by a message on - the -dev list) ........ - -2007-04-11 13:40 +0000 [r61342-61373] Nadi Sarrar <ns@beronet.com> - - * channels/chan_misdn.c: Export AOCD variables on misdn_hangup. - - * channels/chan_misdn.c: Ignore facility messages in case we don't - have a corresponding channel object. - - * channels/chan_misdn.c: AOCD's are now exported to asterisk - channel variables. - -2007-04-10 16:05 +0000 [r61220] Russell Bryant <russell@digium.com> - - * main/Makefile, main/http.c, main/minimime (removed): File upload - support was added to solve some needs for the Asterisk GUI. - However, after much discussion, it has been decided that adding - this to 1.4 is not in the best interests of the project. It has - been removed here, but will remain in trunk. - -2007-04-10 12:43 +0000 [r61183] Nadi Sarrar <ns@beronet.com> - - * channels/misdn_config.c, /: Merged revisions 61170 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r61170 | nadi | 2007-04-10 14:31:45 +0200 (Di, 10 Apr - 2007) | 2 lines msns config parameter defaults to '*' ........ - -2007-04-10 05:18 +0000 [r61136] Steve Murphy <murf@digium.com> - - * apps/app_cdr.c, main/cdr.c, res/res_features.c: Finished up a - previous fix to overcome a compiler warning; the app NoCDR() has - been updated to mark the channel CDR as POST_DISABLED instead of - destroying the CDR; this way its flags are propagated thru a - bridge and the CDR is actually dropped. The cases where only one - channel in a bridge has a CDR was cleaned up. - -2007-04-09 19:58 +0000 [r61072] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 61038 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r61038 | oej | 2007-04-09 21:38:59 +0200 (Mon, 09 Apr 2007) | 3 - lines - Don't send ActionID before Response: header. - Don't use - a blank in an AMI header ........ - -2007-04-09 19:55 +0000 [r61062-61070] Kevin P. Fleming <kpfleming@digium.com> - - * main/minimime/mm_envelope.c, res/res_features.c: fix up some - warnings found using --enable-dev-mode - - * main/minimime/Doxyfile (removed), - main/minimime/tests/messages/CVS (removed), - main/minimime/tests/CVS (removed): remove some more stuff we - don't need - -2007-04-09 19:41 +0000 [r61042-61044] Russell Bryant <russell@digium.com> - - * main/minimime/test (removed): Remove another directory that - should no longer be there - - * main/minimime/Make.conf (removed), main/minimime/mytest_files - (removed), main/minimime/.cvsignore (removed), main/minimime/sys - (removed), main/minimime/mm-docs (removed): Remove various files - that I thought I already removed. - -2007-04-09 19:05 +0000 [r61022] Jason Parker <jparker@digium.com> - - * apps/app_queue.c: Use the appropriate interface name with - COMPLETECALLER. Issue 9395. - -2007-04-09 18:32 +0000 [r60989] Steve Murphy <murf@digium.com> - - * channels/chan_oss.c, main/channel.c, main/cdr.c, - channels/chan_phone.c, channels/chan_misdn.c, - channels/chan_skinny.c, channels/chan_features.c, - channels/chan_h323.c, channels/chan_alsa.c, channels/chan_nbs.c, - channels/chan_mgcp.c, apps/app_voicemail.c, main/pbx.c, - channels/chan_vpb.cc, channels/chan_local.c, channels/chan_zap.c, - channels/chan_sip.c, res/res_features.c, channels/chan_agent.c, - include/asterisk/channel.h, channels/chan_gtalk.c, - channels/chan_iax2.c: This is a big improvement over the current - CDR fixes. It may still need refinement, but this won't have as - many folks bothered. - -2007-04-09 18:02 +0000 [r60984] Olle Johansson <oej@edvina.net> - - * res/res_jabber.c: Add final new line after JabberEvent - -2007-04-09 17:22 +0000 [r60936] Jason Parker <jparker@digium.com> - - * /, apps/app_directory.c: Merged revisions 60935 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r60935 | qwell | 2007-04-09 12:22:15 -0500 (Mon, 09 Apr 2007) | 5 - lines Allow matching on names shorter than 3 chars. This also - fixes the case where somebody wants to match on less then 3 - chars. Issue 9071 ........ - -2007-04-09 03:01 +0000 [r60847-60850] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * main/asterisk.c, include/asterisk.h, /: Merged revisions 60849 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r60849 | tilghman | 2007-04-08 21:49:06 -0500 (Sun, 08 Apr 2007) - | 2 lines Don't check for error when lowering priority (according - to the manpage, it should never happen anyway). It might could - happen, though, if another thread messed with the priority, so - safeguard against that (reported via -dev list). ........ - - * channels/chan_local.c, /: Merged revisions 60846 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r60846 | tilghman | 2007-04-08 21:37:18 -0500 (Sun, 08 - Apr 2007) | 2 lines Bug 9505 - If the return value for - local_queue_frame is set, then p->lock is no longer valid. - ........ - -2007-04-09 01:03 +0000 [r60762-60798] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c, /: Merged revisions 60797 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r60797 | file | 2007-04-08 20:59:29 -0400 (Sun, 08 Apr 2007) | 2 - lines When calling a device that then forwards us elsewhere... we - have to make our channels compatible if it is the only channel - being dialed. (issue #9445 reported by marcelbarbulescu) ........ - - * apps/app_queue.c: Allow app_queue to use MONITOR_EXEC even if - MONITOR_OPTIONS is not set. (issue #9495 reported by cduffy) - -2007-04-08 14:14 +0000 [r60661-60713] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, apps/app_macro.c: Merged revisions 60711 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r60711 | tilghman | 2007-04-08 09:00:22 -0500 (Sun, 08 Apr 2007) - | 2 lines Gosub called within a Macro resets the arguments - improperly and causes general weirdness. (Issue 8329) ........ - - * main/http.c: Fix --enable-dev-mode - - * channels/chan_oss.c: Off by one error, resulting in a crash - (Issue 9500) - - * /, main/file.c: Merged revisions 60660 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r60660 | tilghman | 2007-04-07 20:39:25 -0500 (Sat, 07 Apr 2007) - | 2 lines Bug 9486 - memory leak when opening a filestream - ........ - -2007-04-06 20:58 +0000 [r60603] Russell Bryant <russell@digium.com> - - * main/minimime/sys/mm_queue.h, main/minimime/Doxyfile, - main/minimime/mimeparser.yy.c, main/minimime/minimime.c, - main/manager.c, main/minimime/mm_mimepart.c, - main/minimime/test.sh, configure, include/asterisk/compat.h, - main/strcompat.c, main/minimime/mm_internal.h, main/http.c, - main/minimime/tests/parse.c, main/minimime/mm_base64.c, - main/minimime/mm_mimeutil.c, main/minimime/mm.h, - main/minimime/tests, main/minimime/mm_header.c, - main/minimime/mm_error.c, main/Makefile, - main/minimime/mm_codecs.c, main/minimime/mm_param.c, - configure.ac, main/minimime/Makefile, main/minimime/mm_init.c, - include/asterisk/manager.h, main/minimime/strlcpy.c, - configs/http.conf.sample, main/minimime/mm_parse.c, - main/minimime/tests/create.c, main/minimime/mm_contenttype.c, - main/minimime/mm_util.c, main/minimime/mm_envelope.c, - main/minimime/tests/messages/test1.txt, main/minimime/mm_mem.c, - main/minimime/tests/messages/test2.txt, - main/minimime/tests/messages/test3.txt, - main/minimime/mimeparser.h, main/minimime/mimeparser.tab.c, - main/minimime/tests/messages/test4.txt, - main/minimime/tests/messages/test5.txt, main/minimime/mm_util.h, - main/minimime/tests/messages/test6.txt, main/minimime/strlcat.c, - main/minimime/mm_mem.h, main/minimime/tests/messages/test7.txt, - main/minimime/mimeparser.l, main/minimime/mm_context.c, - main/minimime/mimeparser.tab.h, main/minimime (added), - main/minimime/mm_warnings.c, main/minimime/mm_queue.h, - main/minimime/tests/messages, include/asterisk/autoconfig.h.in, - main/minimime/mimeparser.y, Makefile.moddir_rules, - main/minimime/sys, main/minimime/tests/Makefile: To be able to - achieve the things that we would like to achieve with the - Asterisk GUI project, we need a fully functional HTTP interface - with access to the Asterisk manager interface. One of the things - that was intended to be a part of this system, but was never - actually implemented, was the ability for the GUI to be able to - upload files to Asterisk. So, this commit adds this in the most - minimally invasive way that we could come up with. A lot of work - on minimime was done by Steve Murphy. He fixed a lot of bugs in - the parser, and updated it to be thread-safe. The ability to - check permissions of active manager sessions was added by Dwayne - Hubbard. Then, hacking this all together and do doing the - modifications necessary to the HTTP interface was done by me. - -2007-04-06 20:32 +0000 [r60568-60572] Dwayne M. Hubbard <dhubbard@digium.com> - - * UPGRADE.txt: clarified a sentence in the format_wav section - - * UPGRADE.txt: updated UPGRADE.txt with format_wav GAIN change and - plan to remove GAIN code from trunk - -2007-04-06 19:50 +0000 [r60521-60565] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c: When a station picks up a trunk that was on - hold, make the hints reflect that nobody has the trunk on hold - anymore. - - * apps/app_meetme.c: Fix a few problems with SLA. (issue #9459, - reported by francesco_r, fixed by me) * The original behavior was - that if one station put a call on hold, another one picked it up, - and then hung up, the code would still consider the call on hold - by the first station, so the trunk would not be hung up. However, - to better comply with what most people seem to expect it to - behave, it will now hang up the trunk. * Fix a problem with - "barge=no". This was only intended to prevent people from joining - calls that are in progress. However, it also prevented other - people from picking up a call that was on hold. This has been - fixed. * When there are no active stations on a trunk and it is - on hold, the code now indicates the HOLD and UNHOLD conditions to - the trunk channel. This allows music on hold to be played to the - trunk when it is on hold. - -2007-04-06 18:21 +0000 [r60459-60485] Matt Frederickson <creslin@digium.com> - - * channels/chan_zap.c: Make sure we check the faxdetect option - before doing fax processing - - * channels/chan_zap.c, /: Merged revisions 60456 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r60456 | mattf | 2007-04-06 12:03:15 -0500 (Fri, 06 Apr 2007) | 2 - lines There should only be one code path for doing DTMF - conditionals on channels. This fixes it. ........ - -2007-04-06 14:49 +0000 [r60399] Kevin P. Fleming <kpfleming@digium.com> - - * /, codecs/codec_zap.c: Merged revisions 60398 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r60398 | kpfleming | 2007-04-06 09:41:37 -0500 (Fri, 06 Apr 2007) - | 2 lines remove undocumented 'cardsmode' parameter and stop - searching for transcoders during reload() ........ - -2007-04-06 01:14 +0000 [r60361] Joshua Colp <jcolp@digium.com> - - * res/res_speech.c, apps/app_speech_utils.c, - include/asterisk/speech.h: Add support for returning different - types of results (ie: NBest). - -2007-04-05 22:58 +0000 [r60325] Dwayne M. Hubbard <dhubbard@digium.com> - - * formats/format_wav.c: modified default GAIN for issue 5823, - thanks jrwalliker - -2007-04-05 22:35 +0000 [r60323] Steve Murphy <murf@digium.com> - - * configs/cdr_custom.conf.sample, configs/cdr.conf.sample: Added - some clarification to the example configs for CDRs, on how to - select a backend. Also, made cdr-csv the default if you 'make - samples', and no other changes. - -2007-04-05 16:10 +0000 [r60268] Jason Parker <jparker@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 60267 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r60267 | qwell | 2007-04-05 11:09:41 -0500 (Thu, 05 Apr 2007) | 5 - lines Just because we can't find the voicemail configuration - file, doesn't mean that the module failed to load. The user could - be using realtime. Issue #9473 ........ - -2007-04-05 15:47 +0000 [r60265] Russell Bryant <russell@digium.com> - - * main/http.c: Add the MIME type for gif by request from Pari - -2007-04-05 12:55 +0000 [r60214] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 60213 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r60213 | file | 2007-04-05 08:52:50 -0400 (Thu, 05 Apr 2007) | 2 - lines Only unlock our pvt and net locks if we are actually going - to try to lock the owner again. (issue #9472 reported by zoa) - ........ - -2007-04-04 17:40 +0000 [r60013-60137] Russell Bryant <russell@digium.com> - - * main/manager.c, /: Merged revisions 60134 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r60134 | russell | 2007-04-04 12:38:47 -0500 (Wed, 04 Apr 2007) | - 6 lines It is valid to redirect channels via the manager - interface that are not in the UP state. Instead of checking for - that to prevent to ensure a dead channel doesn't get redirected, - just use the ast_check_hangup() API call. (issue #9457, reported - by Callmewind, patch by me) (related to issue #8977) ........ - - * channels/chan_sip.c: Add a Content-Length of 0 to the response - built by transmit_response_with_unsupported(). (issue #9454, - reported by makoto, fixed by me) - - * /, channels/chan_sip.c: Merged revisions 60083 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r60083 | russell | 2007-04-04 11:37:04 -0500 (Wed, 04 Apr 2007) | - 4 lines Fix the return value of handle_common_options() so that - it always properly indicates whether it handled the option or - not. (issue #9455, reported by Netview, fixed by me) ........ - - * apps/app_meetme.c: Fix a problem where if a trunk was hung up - while it was on hold, all of the hints would reflect the line - still on hold, even though it should reflect that it is back to - not in use. (issue #9459, reported by francesco_r, fixed by me) - - * /: Blocked revisions 60016 via svnmerge ........ r60016 | russell - | 2007-04-03 18:23:23 -0500 (Tue, 03 Apr 2007) | 3 lines Add a - missing "\r\n" in the body of the NOTIFY that is sent to indicate - the status of a transfer. (issue #9388, reported by rarritt) - ........ - - * /: Blocked revisions 60014 via svnmerge ........ r60014 | russell - | 2007-04-03 18:00:10 -0500 (Tue, 03 Apr 2007) | 3 lines Use the - more generic check for "sed -r" support that was already present - in 1.4. (related to issue #9399) ........ - - * /: Blocked revisions 60012 via svnmerge ........ r60012 | russell - | 2007-04-03 17:54:49 -0500 (Tue, 03 Apr 2007) | 3 lines On - Darwin, the -r argument to sed is not valid. It has to be -E. - (issue #9399, reported by jcovert) ........ - -2007-04-03 19:40 +0000 [r59963] Joshua Colp <jcolp@digium.com> - - * apps/app_speech_utils.c: Don't clash when a person both speaks - and uses DTMF. - -2007-04-03 19:16 +0000 [r59853-59939] Russell Bryant <russell@digium.com> - - * /, channels/chan_sip.c: Merged revisions 59938 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59938 | russell | 2007-04-03 14:15:04 -0500 (Tue, 03 Apr 2007) | - 4 lines Don't attempt to report configuration errors in - build_user(). oej pointed out that for a "friend" entry, this - won't work, because all user options are valid for peers, but not - the other way around. ........ - - * /, channels/chan_sip.c: Merged revisions 59916 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59916 | russell | 2007-04-03 13:43:54 -0500 (Tue, 03 Apr 2007) | - 3 lines Make chan_sip report when it encounters an unknown - option. (issue #9440, reported by nightcrawler) ........ - - * /, main/app.c: Merged revisions 59886 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59886 | russell | 2007-04-03 12:58:19 -0500 (Tue, 03 Apr 2007) | - 5 lines When doing a built-in blind or attended transfer, restore - the ability to use '#' to terminate the number and immediately do - the transfer instead of having to dial the number and just wait - for the feature digit timeout. (issue #8366, xueliangliang) - ........ - - * Makefile: Ensure that menuselect gets executed in dependency - check mode every time you run make. - -2007-04-03 11:02 +0000 [r59804] Nadi Sarrar <ns@beronet.com> - - * channels/misdn_config.c, /, channels/misdn/chan_misdn_config.h: - Merged revisions 59788,59803 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59788 | nadi | 2007-04-03 11:37:00 +0200 (Di, 03 Apr 2007) | 2 - lines Use the new sysfs way of mISDN 1.2 to check if a port is NT - or not. ........ r59803 | nadi | 2007-04-03 12:40:58 +0200 (Di, - 03 Apr 2007) | 2 lines ptp is the 5th bit, not the 4th. ........ - -2007-04-03 07:20 +0000 [r59774] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/misdn_config.c, - channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h: - Merged revisions 59623-59624,59639 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59623 | crichter | 2007-04-02 09:12:24 +0200 (Mo, 02 Apr 2007) | - 1 line we can now make 30 channels on a PRI (before we forgot - chan 31..) ........ r59624 | crichter | 2007-04-02 09:25:54 +0200 - (Mo, 02 Apr 2007) | 1 line don't be verbose if no need ........ - r59639 | crichter | 2007-04-02 14:08:12 +0200 (Mo, 02 Apr 2007) | - 1 line added option which allows us to accept incoming SETUP - Messages without automatically sending Proceeding or Setup - Acknowledge, this is useful with some broken switches and if you - want to Release incoming calls without previously having - acknowledged them. The new option is - noautorespond_on_setup=yes|no default is no, so we don't break - the existing behaviour ........ - -2007-04-02 18:58 +0000 [r59724] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 59723 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59723 | file | 2007-04-02 14:55:25 -0400 (Mon, 02 Apr 2007) | 2 - lines Increase the maximum size for a string of mailboxes to - 1024. (issue #9270 reported by rtucker) ........ - -2007-04-02 17:31 +0000 [r59688] Steve Murphy <murf@digium.com> - - * pbx/pbx_ael.c: continue in for-loop should go to the incrementer, - not the test. As per 9435, thanks to marcelbarbulescu - -2007-04-02 15:39 +0000 [r59654] Russell Bryant <russell@digium.com> - - * main/netsock.c, /: Merged revisions 59608 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59608 | russell | 2007-04-01 17:35:25 -0500 (Sun, 01 Apr 2007) | - 6 lines Add the SO_REUSEADDR flag to sockets handled by netsock. - This is needed by the patch that went in for issue 7874. - chan_iax2 needs to be able to create socket that is lisetning on - INADDR_ANY, but also be able to bind sockets to specific - addresses. (Thanks to Stevenson on the asterisk-dev mailing list - for explaining why this flag was needed.) ........ - -2007-03-30 22:50 +0000 [r59573] Jason Parker <jparker@digium.com> - - * configure, main/Makefile, acinclude.m4: Add linux-uclibc host - arch..."thingy". Sorry, I don't know what it's called... - -2007-03-30 17:51 +0000 [r59452-59522] Steve Murphy <murf@digium.com> - - * main/cdr.c, main/channel.c, main/pbx.c, res/res_features.c, - include/asterisk/cdr.h: several changes via kpflemings review - - * main/cdr.c, main/channel.c, main/pbx.c, res/res_features.c, - include/asterisk/cdr.h: These mods fix CDR issues from 8221, - 8593, 8680, 8743, and perhaps others. Mainly with CDRs generated - from transfer situations. - - * configs/extensions.conf.sample: A small clarification to keep - bugs from being filed, and confusion from rising, if - clearglobalvars is set, and globals are set in the AEL file. - (9419) - -2007-03-29 17:43 +0000 [r59363] Russell Bryant <russell@digium.com> - - * res/res_jabber.c: When building a response to a subscription, the - "from" must be the full Jabber ID. This fixes some problems where - jabber users are not able to add their Asterisk account to their - user list, since they are unable to get Asterisk to approve their - subscription. (issue #8210, reported by caspy, and verified by - bradtem) - -2007-03-29 17:38 +0000 [r59361] Joshua Colp <jcolp@digium.com> - - * /, apps/app_meetme.c: Merged revisions 59360 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59360 | file | 2007-03-29 13:33:58 -0400 (Thu, 29 Mar 2007) | 2 - lines Keep a global array of variables indicating whether certain - conference rooms are in use. This ensures that two people going - into a new dynamic conference when the 'e' option is set don't go - into the same conference room. (issue #8835 reported by eliel) - ........ - -2007-03-29 17:17 +0000 [r59304-59358] Russell Bryant <russell@digium.com> - - * main/rtp.c, /: Merged revisions 59357 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59357 | russell | 2007-03-29 12:14:33 -0500 (Thu, 29 Mar 2007) | - 5 lines If an error occurs when reading from an RTP socket, and - the error code does not indicate that we should try again, then - return NULL instead of a "null frame". This will prevent Asterisk - from trying over and over again, and eventually causing the - system to crash. (issue #8285, john) ........ - - * /: Blocked revisions 59355 via svnmerge ........ r59355 | russell - | 2007-03-29 12:10:28 -0500 (Thu, 29 Mar 2007) | 3 lines Backport - the change to chan_iax2 to return NULL instead of a "null frame" - from its read callback. See revision 59341 to the 1.4 branch for - more info. ........ - - * channels/chan_iax2.c: When the IAX2 read callback gets called, - return NULL instead of a "null frame". This will cause Asterisk - to hangup the call instead of keep trying whatever it was doing. - Under normal conditions, this function would *never* be called. - However, the author of this patch says an error will occur that - will cause it to get called every 100 thousand calls or so. When - this does happen, it puts the channel in a loop that eventually - brings down the system. So, hangup up the call is certainly a - better alternative. (issue #8286, john) - - * Makefile: Export the GTK2 library and include information to sub - Makefiles. - -2007-03-29 16:07 +0000 [r59300-59302] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, cdr/cdr_odbc.c: Merged revisions 59301 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59301 | tilghman | 2007-03-29 11:04:46 -0500 (Thu, 29 Mar 2007) - | 3 lines Issue 9415 - No point to getting a diagnostic field if - we aren't doing anything with the information. (Plus, it tends to - crash the Postgres ODBC driver.) ........ - - * /: Blocked revisions 59299 via svnmerge ........ r59299 | - tilghman | 2007-03-29 10:33:10 -0500 (Thu, 29 Mar 2007) | 2 lines - Change ENV section to use setenv, instead of putenv (Alexandru - Pirvulescu <sigxcpu@gmail.com>, reported via -dev list) ........ - -2007-03-28 03:38 +0000 [r59281-59289] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * res/res_odbc.c: Another crash that I thought we had fixed already - - Issue 9396 - - * apps/app_voicemail.c, /: Merged revisions 59283 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59283 | tilghman | 2007-03-27 18:36:49 -0500 (Tue, 27 Mar 2007) - | 2 lines Oops ........ - - * apps/app_voicemail.c, /: Merged revisions 59280 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59280 | tilghman | 2007-03-27 18:31:20 -0500 (Tue, 27 Mar 2007) - | 2 lines Fix a few remaining bad mmap(2) return values ........ - -2007-03-27 23:20 +0000 [r59262-59278] Russell Bryant <russell@digium.com> - - * /, apps/app_directory.c: Merged revisions 59277 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59277 | russell | 2007-03-27 18:19:41 -0500 (Tue, 27 Mar 2007) | - 3 lines Fix the check of the return value from mmap(). Thanks to - Corydon for catching this one. ........ - - * apps/app_directory.c: Fix app_directory to actually compile with - ODBC_STORAGE, and update the code to the latest res_odbc API. - - * apps/Makefile: Fix app_directory when ODBC_STORAGE is being used. - The Makefile did not properly ensure that this information got - copied from what was selected for app_voicemail. (issue #9224) - - * channels/chan_sip.c: Fix the check that ensures that the CHANNEL - function's first argument is "rtpqos". Thanks, Corydon. :) - -2007-03-27 18:16 +0000 [r59261] Steve Murphy <murf@digium.com> - - * pbx/pbx_ael.c: via 9373 (duplicate context in AEL crashes - asterisk), kpfleming pointed on asterisk-dev, that DECLINE in - this case the proper thing to do. This change now has it doing - the proper thing. - -2007-03-27 18:05 +0000 [r59256-59259] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 59258 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59258 | russell | 2007-03-27 13:04:02 -0500 (Tue, 27 Mar 2007) | - 4 lines Fix the use of the "sourceaddress" option when "bindaddr" - is set to 0.0.0.0 instead of having each interface explicitly - listed. (issue #7874, patch by stevens) ........ - - * channels/chan_sip.c, funcs/func_channel.c: Convert the RTPQOS - function to just be additional parameter of the CHANNEL function. - This way, it will be possible for other RTP based channel drivers - to expose this information in the future. - -2007-03-27 15:00 +0000 [r59254] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /: Merged revisions 59252 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r59252 | crichter | 2007-03-27 15:56:15 +0200 (Di, 27 - Mär 2007) | 1 line fixed #9355 ........ - -2007-03-26 21:45 +0000 [r59230] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * channels/chan_sip.c: Oops, this should be case insensitive - -2007-03-26 21:41 +0000 [r59228] Steve Murphy <murf@digium.com> - - * pbx/pbx_ael.c: fix for 9373 (duplicate context in AEL crashes - asterisk). I turned a duplicate context from a WARNING to an - ERROR. Now you get a module load failure, and asterisk just - exits. That's better than a crash, right\? - -2007-03-26 21:37 +0000 [r59227] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * channels/chan_sip.c: Change this to a single dp function to make - oej happy. - -2007-03-26 20:06 +0000 [r59225] Steve Murphy <murf@digium.com> - - * main/config.c: Fix for 9257; by eliminating the globals in - main/config.c, we make it thread-safe, which is a minimum - requirement. - -2007-03-26 19:34 +0000 [r59223] Joshua Colp <jcolp@digium.com> - - * apps/app_speech_utils.c: Add ability to specify no timeout. This - means as soon as the prompt is done playing it moves on to the - next priority. - -2007-03-26 18:33 +0000 [r59215-59217] Russell Bryant <russell@digium.com> - - * apps/app_voicemail.c: Somehow the code for building the email for - voicemail got out of sync. This change makes a few tweaks to get - 1.4 in sync with trunk. (issue #9301) - - * apps/app_meetme.c: Fix some codec negotiation problems when - CallerID support is not enabled in SLA. (issue #9308, reported by - twilson) - -2007-03-26 18:13 +0000 [r59213] Joshua Colp <jcolp@digium.com> - - * apps/app_speech_utils.c: Make SpeechBackground obey the digit - timeout value. - -2007-03-26 17:53 +0000 [r59207-59209] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Rename the new dialplan functions to match - the variable name - - * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: The - AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in - some because they get set in sip_hangup. So, there are common - situations where the variables will not be available in the - dialplan at all. So, this patch provides an alternate method for - getting to this information by introducing AUDIORTPQOS and - VIDEORTPQOS dialplan functions. (issue #9370, patch by Corydon76, - with some testing by blitzrage) - -2007-03-26 17:38 +0000 [r59206] Steve Murphy <murf@digium.com> - - * main/ast_expr2.fl, main/ast_expr2f.c, pbx/ael/ael_lex.c, - pbx/ael/ael.flex: A fix for the flex input files, DONT_COMPILE, - and STANDALONE_AEL - -2007-03-26 15:25 +0000 [r59202] Nadi Sarrar <ns@beronet.com> - - * channels/misdn/isdn_lib.c, channels/misdn_config.c, - channels/misdn/isdn_lib.h, channels/chan_misdn.c, configure, - include/asterisk/autoconfig.h.in, channels/misdn/Makefile, - channels/misdn/chan_misdn_config.h, configure.ac: * mISDN >= 1.2 - provides a dsp pipeline for i.e. echo cancellation modules, make - chan_misdn use it. * add a check for linux/mISDNdsp.h to - configure.ac and update the autogenerated files: 'configure', - 'autoconfig.h.in' (the 'configure' script was not in sync with - the latest configure.ac, so the diff is a bit bigger than - expected). - -2007-03-26 15:16 +0000 [r59200] Joshua Colp <jcolp@digium.com> - - * pbx/ael/ael_lex.c: Have ast_copy_string magically appear in the - aelparse binary! DONT_OPTIMIZE should now work once again. - -2007-03-24 01:39 +0000 [r59195] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 59194 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59194 | file | 2007-03-23 21:35:49 -0400 (Fri, 23 Mar 2007) | 2 - lines Only try to handle a response if it has a response code. - (ASA-2007-011) ........ - -2007-03-23 16:11 +0000 [r59188-59189] Steve Murphy <murf@digium.com> - - * /: blocking out the fix in 59187... already incorporated here - - * /, apps/app_macro.c: Merged revisions 59186 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59186 | murf | 2007-03-23 09:57:26 -0600 (Fri, 23 Mar 2007) | 1 - line Added a few words in the Macro doc strings about the - behavior of macros with hangups (et al.), as per 9337 ........ - -2007-03-22 23:40 +0000 [r59180-59182] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: don't allow string input to overrun the - buffer to hold it (ASA-2007-010) - - * channels/chan_misdn.c: remove variables that are no longer used - (--enable-dev-mode is good, developers should be using it) - -2007-03-22 14:40 +0000 [r59145] Steve Murphy <murf@digium.com> - - * utils/Makefile: The stuff in utils was compiling with -O6 even if - DONT_OPTIMIZE is set in menuconfig. Added the include to fix that - -2007-03-21 18:08 +0000 [r59081-59089] Joshua Colp <jcolp@digium.com> - - * main/http.c: Add svg mimetype for pari. - - * res/res_monitor.c, /: Merged revisions 59086 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r59086 | file | 2007-03-21 14:03:20 -0400 (Wed, 21 Mar 2007) | 2 - lines Indicate the filename changed when it is changed. (issue - #9311 reported by jsmith) ........ - - * channels/chan_sip.c: Until we can do media level parsing for - sendrecv/etc just use the first value found. This crept up when a - phone was offered audio+video and returned an inactive video - stream. chan_sip thought the phone said to put the person on hold - but that was totally wrong. (issue #9319 reported by benbrown) - -2007-03-20 21:04 +0000 [r59078] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * main/logger.c: Fix defines for inline stack backtraces (only used - by developers anyway) - -2007-03-20 20:42 +0000 [r59076] Joshua Colp <jcolp@digium.com> - - * channels/iax2-parser.c: Copy len variable as well, should fix - remaining IAX2 DTMF issues. - -2007-03-20 17:48 +0000 [r59069-59070] Steve Murphy <murf@digium.com> - - * apps/app_stack.c: Ooops. Sorry, messed up app_stack. This should - return it to its previous, untouched, state. - - * apps/app_stack.c, pbx/pbx_ael.c, include/asterisk/ael_structs.h: - The fix for the AEL <<security hole>> (bug 9316) is here... - -2007-03-20 13:16 +0000 [r59064] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/misdn_config.c, - channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, - channels/misdn/chan_misdn_config.h: Merged revisions - 58849-58850,59062-59063 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r58849 | crichter | 2007-03-13 12:58:16 +0100 (Di, 13 Mär 2007) | - 1 line added method standard_dec for dialing out on groups, to - avoid conflicts, which caused issues with some ISDN providers - ........ r58850 | crichter | 2007-03-13 13:58:32 +0100 (Di, 13 - Mär 2007) | 1 line fixed the crypt_keys stuff ........ r59062 | - crichter | 2007-03-20 10:18:06 +0100 (Di, 20 Mär 2007) | 1 line - avoid sending a disconnect when we already received one. ........ - r59063 | crichter | 2007-03-20 10:23:22 +0100 (Di, 20 Mär 2007) | - 1 line modified a loglevel ........ - -2007-03-19 Jason Parker <jparker@digium.com> - - * Asterisk 1.4.2 released. - -2007-03-19 22:29 +0000 [r59049] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * funcs/func_strings.c: Oops, this should have been a %d all along - -2007-03-19 15:52 +0000 [r59042] Joshua Colp <jcolp@digium.com> - - * funcs/func_cdr.c: Fix typo in help for CDR function. (issue #9295 - reported by ajohnson) - -2007-03-19 15:42 +0000 [r59040] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * configs/sip_notify.conf.sample: Fix unescaped semicolon (reported - via -dev list) - -2007-03-18 20:37 +0000 [r59037] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #9313, Asterisk crash on SIP return - code 0 (reported by qwerty1979) - -2007-03-18 16:36 +0000 [r59035] BJ Weschke <bweschke@btwtech.com> - - * apps/app_followme.c: Don't return a non-zero return code if the - profile doesn't exist, to match what the documentation says it - already does. (#9307 Reported by kkiely) - -2007-03-16 16:12 +0000 [r58992] Joshua Colp <jcolp@digium.com> - - * apps/app_page.c: Wait for the async thread to exit when hanging - up all of the paged phones under all circumstances. (issue #9181 - reported by PhilSmith) - -2007-03-16 01:42 +0000 [r58947-58957] Russell Bryant <russell@digium.com> - - * configs/sla.conf.sample: fix a couple SLA documentation - references - - * doc/ajam.tex (removed), doc/manager.tex (removed), doc/misdn.tex - (removed), doc/freetds.txt (added), doc/odbcstorage.txt (added), - doc/sla.tex, doc/cygwin.txt (added), doc/model.txt (added), - doc/channelvariables.txt (added), doc/ael.txt (added), - doc/billing.tex (removed), build_tools/prep_tarball, - doc/callingpres.txt (added), doc/enum.txt (added), - doc/localchannel.tex (removed), doc/musiconhold-fpm.txt (added), - doc/cdrdriver.tex (removed), build_tools/make_buildopts_h, - doc/security.txt (added), doc/imapstorage.txt (added), - doc/PEERING, main/pbx.c, doc/odbcstorage.tex (removed), - doc/freetds.tex (removed), doc/privacy.txt (added), configure.ac, - doc/iax.txt (added), doc/ael.tex (removed), - doc/channelvariables.tex (removed), doc/enum.tex (removed), - doc/security.tex (removed), doc/math.txt (added), Makefile, - doc/imapstorage.tex (removed), doc/privacy.tex (removed), - doc/realtime.txt (added), doc/dundi.txt (added), doc/mysql.txt - (added), apps/app_voicemail.c, doc/cliprompt.txt (added), - doc/chaniax.txt (added), doc/app-sms.txt (added), - doc/ast_appdocs.tex (removed), doc/realtime.tex (removed), - doc/ices.txt (added), doc/dundi.tex (removed), - doc/linkedlists.txt (added), doc/queuelog.txt (added), - doc/extconfig.txt (added), doc/radius.txt (added), - doc/cliprompt.tex (removed), doc/chaniax.tex (removed), - doc/hardware.txt (added), doc/mp3.txt (added), doc/app-sms.tex - (removed), doc/ices.tex (removed), doc/asterisk.tex (removed), - doc/queuelog.tex (removed), doc/configuration.txt (added), - doc/asterisk-conf.txt (added), doc/sla.pdf (added), - doc/ip-tos.txt (added), doc/hardware.tex (removed), doc/h323.txt - (added), doc/mp3.tex (removed), doc/configuration.tex (removed), - doc/asterisk-conf.tex (removed), doc/jitterbuffer.txt (added), - doc/channels.txt (added), doc/ip-tos.tex (removed), - doc/extensions.txt (added), doc/queues-with-callback-members.txt - (added), doc/apps.txt (added), makeopts.in, doc/ajam.txt (added), - doc/misdn.txt (added), doc/manager.txt (added), - doc/jitterbuffer.tex (removed), doc/extensions.tex (removed), - doc/billing.txt (added), doc/localchannel.txt (added), - doc/queues-with-callback-members.tex (removed), doc/cdrdriver.txt - (added), doc/00README.1st (added): Making these documentation - changes in the 1.4 branch upset various people, so these chanes - will only be done in the trunk. - - * build_tools/prep_tarball: Add the --pdf option to the usage of - rubber in prep_tarball - - * Makefile, build_tools/menuselect-deps.in, configure, - include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add - configure script checking for GTK2 and some additional Makefile - targets to support gmenuselect - -2007-03-15 23:52 +0000 [r58946] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * main/pbx.c, doc/ast_appdocs.tex: Refashion dump command to match - common syntax and update the resulting appdocs TeX file - -2007-03-15 23:24 +0000 [r58941] Russell Bryant <russell@digium.com> - - * doc/asterisk.tex: add a link to the rubber homepage - -2007-03-15 23:11 +0000 [r58939] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_setcdruserfield.c, main/pbx.c, - apps/app_hasnewvoicemail.c, apps/app_settransfercapability.c: - Expand deprecation warnings from simply warning on use to the - builtin documentation. - -2007-03-15 22:51 +0000 [r58935-58937] Russell Bryant <russell@digium.com> - - * doc/asterisk.tex, Makefile: Add Asterisk version information to - the generated PDF - - * build_tools/prep_tarball: have prep_tarball attempt to build - asterisk.pdf - -2007-03-15 22:32 +0000 [r58933] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * funcs/func_realtime.c: Function works fine, but the documentation - is backwards. - -2007-03-15 22:25 +0000 [r58931] Russell Bryant <russell@digium.com> - - * doc/ajam.tex (added), doc/manager.tex (added), doc/misdn.tex - (added), doc/freetds.txt (removed), doc/odbcstorage.txt - (removed), configure, doc/sla.tex, doc/cygwin.txt (removed), - doc/model.txt (removed), doc/channelvariables.txt (removed), - doc/ael.txt (removed), doc/billing.tex (added), - doc/callingpres.txt (removed), doc/enum.txt (removed), - doc/localchannel.tex (added), doc/musiconhold-fpm.txt (removed), - doc/cdrdriver.tex (added), build_tools/make_buildopts_h, - doc/security.txt (removed), doc/imapstorage.txt (removed), - doc/PEERING, main/pbx.c, doc/odbcstorage.tex (added), - doc/freetds.tex (added), doc/privacy.txt (removed), configure.ac, - doc/iax.txt (removed), doc/ael.tex (added), - doc/channelvariables.tex (added), doc/enum.tex (added), - doc/security.tex (added), doc/math.txt (removed), Makefile, - doc/imapstorage.tex (added), doc/privacy.tex (added), - doc/realtime.txt (removed), doc/dundi.txt (removed), - doc/mysql.txt (removed), apps/app_voicemail.c, doc/cliprompt.txt - (removed), doc/chaniax.txt (removed), doc/app-sms.txt (removed), - doc/ast_appdocs.tex (added), doc/realtime.tex (added), - doc/ices.txt (removed), doc/dundi.tex (added), - doc/linkedlists.txt (removed), doc/queuelog.txt (removed), - doc/extconfig.txt (removed), doc/radius.txt (removed), - doc/cliprompt.tex (added), doc/chaniax.tex (added), - doc/hardware.txt (removed), doc/mp3.txt (removed), - doc/app-sms.tex (added), doc/ices.tex (added), doc/asterisk.tex - (added), doc/queuelog.tex (added), doc/configuration.txt - (removed), doc/asterisk-conf.txt (removed), doc/sla.pdf - (removed), doc/ip-tos.txt (removed), doc/hardware.tex (added), - doc/h323.txt (removed), doc/mp3.tex (added), - doc/configuration.tex (added), doc/asterisk-conf.tex (added), - doc/jitterbuffer.txt (removed), doc/channels.txt (removed), - doc/ip-tos.tex (added), doc/extensions.txt (removed), - doc/queues-with-callback-members.txt (removed), doc/apps.txt - (removed), makeopts.in, doc/ajam.txt (removed), doc/misdn.txt - (removed), doc/manager.txt (removed), doc/jitterbuffer.tex - (added), doc/extensions.tex (added), doc/billing.txt (removed), - doc/localchannel.txt (removed), - doc/queues-with-callback-members.tex (added), doc/cdrdriver.txt - (removed), doc/00README.1st (removed): Merge changes from - svn/asterisk/team/russell/LaTeX_docs. * Convert most of the doc - directory into a single LaTeX formatted document so that we can - generate a PDF, HTML, or other formats from this information. * - Add a CLI command to dump the application documentation into - LaTeX format which will only be include if the configure script - is run with --enable-dev-mode. * The PDF turned out to be close - to 1 MB, so it is not included. However, you can simply run "make - asterisk.pdf" to generate it yourself. We may include it in - release tarballs or have automatically generated ones on the web - site, but that has yet to be decided. - -2007-03-15 18:13 +0000 [r58923] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Don't assume that the pvt structure will - still exist after calling schedule_delivery as it may not. (issue - #9278 reported by fmachado) - -2007-03-14 19:18 +0000 [r58894-58906] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Some people like to put "limitonpeer" - instead of "limitonpeers" in their configuration. While we're at - it, support "limitonpeerz" and "limitonpeerssssss". (inspired by - issue #9172) - - * doc/sla.pdf, doc/sla.tex: Add a more basic example setup to the - examples section - - * doc/security.txt, /: Merged revisions 58896 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r58896 | russell | 2007-03-14 11:38:48 -0500 (Wed, 14 Mar 2007) | - 3 lines Add a note to the security file that the Asterisk CLI and - log files may contain sensitive information, and that people - should keep this in mind. ........ - - * configs/sla.conf.sample, apps/app_meetme.c: By default, don't - attempt to do any CallerID handling at all with SLA because it is - known to not work properly in some situations. However, add an - option to enable it for those that would like to use it anyway. - The short story behind this is that to properly handle CallerID - with SLA, we need the ability to change the CallerID on an - existing call, and we are not ready to handle that. - -2007-03-14 01:47 +0000 [r58880] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * funcs/func_strings.c: Issue 9162 - - pbx_substitute_variables_helper assumes the buffer is initialized - to all zeroes. This fixes a case where it wasn't. - -2007-03-13 23:19 +0000 [r58870-58872] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c: Ensure that the blinky lights show that the - trunk stopped ringing when the trunk hangs up before a station - has answered it. (issue #9234, reported by francesco_r) - - * configs/sla.conf.sample: fix the reference to the SLA - documentation - -2007-03-13 11:49 +0000 [r58843-58848] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 58847 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r58847 | oej | 2007-03-13 12:45:52 +0100 (Tue, 13 Mar 2007) | 2 - lines Issue #9229 - No port in request URI on register to non - default SIP ports (neelakantan) ........ - - * channels/chan_sip.c: Don't hangup the call on OK or errors on - MESSAGE and INFO inside of a dialog (like video update requests). - - * channels/chan_sip.c: Issue #9251 - Clear From URI from user - attributes (tgrman) - -2007-03-12 16:52 +0000 [r58833] Joshua Colp <jcolp@digium.com> - - * /: Blocked revisions 58832 via svnmerge ........ r58832 | file | - 2007-03-12 12:49:49 -0400 (Mon, 12 Mar 2007) | 2 lines We can't - use the assembler version of fetchadd_int under Intel Macs. - (issue #9254 reported by darrell budic) ........ - -2007-03-12 13:08 +0000 [r58825-58826] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged - revisions 57034,57523,57753,58558 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r57034 | crichter | 2007-02-28 17:09:27 +0100 (Mi, 28 Feb 2007) | - 1 line fixed bugs.digium.com bugs: #9157 and bugs.beronet.com - bugs: #302, #303, #304 ........ r57523 | crichter | 2007-03-02 - 19:32:51 +0100 (Fr, 02 Mar 2007) | 1 line fixed typo ........ - r57753 | crichter | 2007-03-04 11:39:50 +0100 (So, 04 Mar 2007) | - 1 line fixed another place where the out_cause was hardcoded to - 16 ........ r58558 | crichter | 2007-03-09 15:43:58 +0100 (Fr, 09 - Mar 2007) | 1 line we can free channel 31 as well, since we can - occupy it ........ - - * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, - channels/chan_misdn.c, channels/misdn/ie.c, - channels/misdn/isdn_msg_parser.c: added UU transceiving and - corect handling for rdnis - -2007-03-12 01:21 +0000 [r58779-58783] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Allow RFC2833 compensation to compensate for even - stupider implementations by queueing up the end frame at the - start, not the actual end. (issue #8963 reported by AndrewZ) - - * channels/chan_sip.c, configs/sip.conf.sample: Add - matchexterniplocally setting which only substitutes your - externip/externhost setting if it matches the localnet setting. I - know of at least two people who need opposite settings, so I made - it an option! (issue #8821 reported by kokoskarokoska) - -2007-03-10 18:11 +0000 [r58638-58705] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Fix a few more places in chan_iax2 where - the ast_frame used for receiving a frame was not properly - initialized. - Interpolating a frame when the jitterbuffer is in - use - decrypting a frame when IAX2 encryption is on - frames in - an IAX2 trunk - - * apps/app_meetme.c: Make the compiler happy and initialize a - variable. - - * doc/sla.pdf (added), doc/sla.txt (removed), doc/sla.tex (added): - Merge some updates to the SLA documentation. I plan to keep - working on this to explain all of the expected behavior with call - handling, configuration details for specific phones, and other - things. However, I got tired of doing it in plain text, so I - switched to using LaTeX. I have included the PDF version. I - haven't been able to get a nice looking plain text version out of - it yet, but I'm not terribly concerned since this is supposed to - be more of the manual, while the plain text sample configuration - file is the reference. - -2007-03-09 21:08 +0000 [r58584-58604] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Fix spelling of unavailable in voicemail - documentation. (issue #9248 reported by tensai) - - * /, channels/chan_sip.c: Merged revisions 58579 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r58579 | file | 2007-03-09 15:46:43 -0500 (Fri, 09 Mar 2007) | 2 - lines If we are unable to lookup the host in a c line we have to - abort, otherwise the previous data is gone and we will - (potentially) have no data when all is said and done. ........ - -2007-03-08 22:15 +0000 [r58510-58512] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c: Hang up the channel that put the call on hold - in the event processing thread to avoid a race condition. Also, - if the station originated the call that it is putting on hold, - don't hang up the trunk if it was the only station on the call - and it is hanging up due to hold and not a normal hangup. - - * channels/chan_zap.c: Add a missing break statement so that - handling the above event does not incorrectly destroy the - channel. (issue #9242, andrew) - -2007-03-08 21:33 +0000 [r58479] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * res/res_odbc.c: Fix segfault (Issue 9236) - -2007-03-08 20:54 +0000 [r58474] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c: Refactor hold handling a bit so that it does - not require keeping the call up when a call is put on hold. - -2007-03-08 18:01 +0000 [r58389-58436] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Make early SDP seeding even smarter! We have to check - codecs in the make_compatible function too. (issue #9221 reported - by marcelbarbulescu) - - * main/dsp.c, /: Merged revisions 58388 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r58388 | file | 2007-03-08 11:04:58 -0500 (Thu, 08 Mar 2007) | 2 - lines Only print out debug message if the definition that makes - the variables shows up was actually defined. (issue #9233 - reported by serginuez) ........ - -2007-03-08 13:23 +0000 [r58351-58354] Kevin P. Fleming <kpfleming@digium.com> - - * main/http.c: this change was not needed; fclose() handles closing - the file descriptor already - - * apps/app_meetme.c: fix a compiler warning, and overwriting 'res' - value - - * main/http.c: fix two cases where HTTP session file descriptors - would not be closed - -2007-03-08 01:01 +0000 [r58243-58320] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c, configure, configure.ac: If we receive - ZT_EVENT_REMOVED, destroy the specified channel. (issue #7256, - tzafrir) Also, update the configure script to make sure that we - don't try to build chan_zap if the installed version of zaptel - does not include ZT_EVENT_REMOVED. - - * /, channels/chan_iax2.c: (This bug was reported to me by Kinsey - Moore) Merged revisions 58242 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r58242 | russell | 2007-03-07 12:17:07 -0600 (Wed, 07 Mar 2007) | - 7 lines Fix a problem where the Asterisk channel name could be - that of the wrong IAX2 user for a call. This is because the first - step of choosing this name is to look for an IAX2 peer that - happens to have the same IP/port number that this call is coming - from and assuming that is it. However, this is not always - correct. So, I have made it change this name after authentication - happens since at that point, we have an exact match. ........ - -2007-03-07 17:52 +0000 [r58240] Joshua Colp <jcolp@digium.com> - - * main/rtp.c, channels/chan_sip.c: Ensure we have (or should have) - at least one matching codec before attempting early bridge SDP - seeding. (issue #9221 reported by marcelbarbulescu) - -2007-03-07 00:27 +0000 [r58165-58168] Russell Bryant <russell@digium.com> - - * /: Blocked revisions 58167 via svnmerge ........ r58167 | russell - | 2007-03-06 18:27:04 -0600 (Tue, 06 Mar 2007) | 2 lines Fix a - misplaced block of code in the 1.2 version of the patch to fix - issue #8977 ........ - - * main/manager.c, /: Merged revisions 58164 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r58164 | russell | 2007-03-06 18:20:13 -0600 (Tue, 06 Mar 2007) | - 4 lines If the channels acquired using the manager Redirect - action are not up, then don't attempt to do anything with them. - It could lead to weird behavior, including crashes. (issue #8977) - ........ - -2007-03-06 23:10 +0000 [r58121] Steve Murphy <murf@digium.com> - - * /, channels/chan_sip.c: Merged revisions 58115 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r58115 | murf | 2007-03-06 15:52:52 -0700 (Tue, 06 Mar 2007) | 1 - line Fix for 9220: Eyebeam cannot renew subscriptions for - presence info. Reason: re-SUBSCRIBE requests don't include Accept - headers, which the rfc says are optional (to put it tersely), (it - uses MAY), and luckily, the sip_pvt struct has the format info - stored, so we simply leave it if the format is set, and the - accept header null. ........ - -2007-03-06 23:00 +0000 [r58119] Russell Bryant <russell@digium.com> - - * configs/voicemail.conf.sample: Clarify the documentation of the - dialout and sendvoicemail options. (issue #9000, caio1982 and - serge-v) - -2007-03-06 20:37 +0000 [r58053] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Merged revisions 58052 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r58052 | oej | 2007-03-06 21:33:21 +0100 (Tue, 06 Mar 2007) | 2 - lines Change error message to proper message ........ - -2007-03-06 18:01 +0000 [r58023] Russell Bryant <russell@digium.com> - - * channels/chan_skinny.c: Return an error of transmit_response is - called without a session. (issue #9002) - -2007-03-05 19:19 +0000 [r57870-57914] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Since chan_iax2 does not support reception - of DTMF with duration ensure that it is set to 0 on the frame. - (issue #8521 reported by gdhgdh) - - * apps/app_meetme.c: Don't create a listen channel and record the - conference unless the option is turned on. (issue #9204 reported - by francesco_r) - - * apps/app_voicemail.c, /: Merged revisions 57869 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r57869 | file | 2007-03-05 12:49:18 -0500 (Mon, 05 Mar 2007) | 2 - lines Make create_dirpath use our standard for return values. -1 - is failure, 0 is success. (issue #9205 reported by ballares) - ........ - -2007-03-05 15:20 +0000 [r57826] Steve Murphy <murf@digium.com> - - * main/pbx.c, /: Merged revisions 57825 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r57825 | murf | 2007-03-05 07:53:57 -0700 (Mon, 05 Mar 2007) | 1 - line Fixed a typo introduced via 9156 (either the gotos or their - doc strings are wrong) ........ - -2007-03-05 04:19 +0000 [r57768-57798] Joshua Colp <jcolp@digium.com> - - * main/slinfactory.c: Don't allow a NULL pointer to reach - ast_frdup. (issue #9155 reported by cmaj) - - * res/res_jabber.c: Don't reference a potentially NULL pointer. - (issue #9199 reported by klolik) - - * main/rtp.c: Preserve marker bit when P2P bridging. (issue #9198 - reported by edgreenberg) - -2007-03-03 15:31 +0000 [r57707] Steve Murphy <murf@digium.com> - - * pbx/ael/ael-test/ref.ael-vtest13, pbx/ael/ael-test/ref.ael-test2, - pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test7: - Updated the regression tests - -2007-03-03 06:45 +0000 [r57649] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c, /: Merged revisions 57648 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r57648 | tilghman | 2007-03-03 00:36:55 -0600 (Sat, 03 Mar 2007) - | 2 lines Memory leak of a list, if call recording was abandoned - ........ - -2007-03-03 00:59 +0000 [r57620] Dwayne M. Hubbard <dhubbard@digium.com> - - * main/say.c: submitted patch for Georgian language, issue 9010, - submitted by Alexander Shaduri - -2007-03-03 00:02 +0000 [r57591] Russell Bryant <russell@digium.com> - - * configs/sla.conf.sample: add missing configuration template. - Thanks to Lacy Moore on asterisk-users for pointing this out\! - -2007-03-02 Russell Bryant <russell@digium.com> - - * Asterisk 1.4.1 released. - -2007-03-02 23:03 +0000 [r57556] Russell Bryant <russell@digium.com> - - * configure, configure.ac: Update the check that is used to - determine whether zaptel transcoder support is present. The - interface has changed. - -2007-03-02 17:06 +0000 [r57477] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 57475 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r57475 | file | 2007-03-02 12:02:46 -0500 (Fri, 02 Mar 2007) | 2 - lines If a SIP message comes in and goes to a method handler that - requires additional values that may not be present then send back - an error. ........ - -2007-03-02 16:55 +0000 [r57426-57473] Steve Murphy <murf@digium.com> - - * main/pbx.c, /: Merged revisions 57458 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r57458 | murf | 2007-03-02 09:39:33 -0700 (Fri, 02 Mar 2007) | 1 - line further refinement in wording of goto documentation, as per - 9156, goto not proceeding to next instruction ........ - - * pbx/pbx_ael.c, utils/ael_main.c: I almost had comma escapes - right, but 9184 points out the problem-- the escape is removed by - pbx_config, and pbx_ael should also, before sending it down into - the pbx engine. Also, you have to insert it back in, if you are - generating extensions.conf code from the AEL. - -2007-03-02 00:20 +0000 [r57364-57396] Russell Bryant <russell@digium.com> - - * main/file.c: Return the correct digit that interrupted the - stream. This fixes exiting the Background application when using - the m option. (issue #9176, mjagdis) - - * configs/sla.conf.sample, apps/app_meetme.c, doc/sla.txt, - include/asterisk/channel.h: Merge changes from - svn/asterisk/team/russell/sla_updates * Originally, I put in the - documentation that only Zap interfaces would be supported on the - trunk side. However, after a discussion with Qwell, we came up - with a way to make IP trunks work as well, using some things - already in Asterisk. So, here it is, this now officially supports - IP trunks. * Update the SLA documentation to reflect how to setup - IP trunks. * Add a section in sla.txt that describes how to set - up an SLA system with voicemail. * Simplify the way DTMF - passthrough is handled in MeetMe. * Fix a bug that exposed itself - when using a Local channel on the trunk side in SLA. The - station's channel needs to be passed to the dial API when dialing - the trunk. * Change a WARNING message to DEBUG in channel.h. This - message is of no use to users. - -2007-03-01 22:21 +0000 [r57318] Joshua Colp <jcolp@digium.com> - - * channels/chan_local.c, /: Merged revisions 57317 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r57317 | file | 2007-03-01 17:19:32 -0500 (Thu, 01 Mar - 2007) | 2 lines Don't even attempt to optimize things when a - proxy channel is involved. It will just explode in weird and - unexplaineable ways. (issue #9175 reported by - clegall_proformatique) ........ - -2007-03-01 03:02 +0000 [r57263] TransNexus OSP Development <support@transnexus.com> - - * doc/osp.txt: 1. Corrected a typo for www.etsi.org. Thank Patrick. - -2007-02-28 23:01 +0000 [r57144-57207] Russell Bryant <russell@digium.com> - - * configs/sla.conf.sample, doc/sla.txt: minor tweaks to the sla - docs - - * configs/sla.conf.sample, apps/app_meetme.c: Merge more changes - from svn/asterisk/team/russell/sla_updates * Add support for - private hold. By setting "hold=private" for a trunk, only the - station that put the call on hold will be able to retrieve it - from hold. Also, by setting "hold=private" for a station, any - call that station puts on hold can only be retrieved by that - station. - - * apps/app_meetme.c: Minor formatting change - - * configs/sla.conf.sample, apps/app_meetme.c: Merge changes from - svn/asterisk/team/russell/sla_updates * Add support for the - "barge=no" option for trunks. If this option is set, then - stations will not be able to join in on a call that is on - progress on this trunk. - -2007-02-28 19:23 +0000 [r57139] Steve Murphy <murf@digium.com> - - * main/pbx.c, /: Merged revisions 57118 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r57118 | murf | 2007-02-28 12:12:41 -0700 (Wed, 28 Feb 2007) | 1 - line a small documentation update, to reflect reality in the goto - doc strings, as per 9156, Goto does not proceed to next prio if - jump fails ........ - -2007-02-28 18:57 +0000 [r57093] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_agent.c: Merged revisions 57092 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r57092 | file | 2007-02-28 13:55:45 -0500 (Wed, 28 Feb - 2007) | 2 lines Fix a few more issues with the agent logoff CLI - command. (issue #9123 reported by arbrandes) ........ - -2007-02-28 18:20 +0000 [r57089] Russell Bryant <russell@digium.com> - - * configs/sla.conf.sample, apps/app_meetme.c: Merge current set of - changes from svn/asterisk/team/russell/sla_updates * Add support - for station ring delays. Ring delays can be set globally for a - station or for specific trunks on the station. * Fix a few bugs - in existing code. * Restructure and Reorganize code to improve - readability and maintainability. * Improve formatting of the "sla - show (trunks|stations)" CLI commands. - -2007-02-28 17:55 +0000 [r57053-57055] Joshua Colp <jcolp@digium.com> - - * apps/app_meetme.c: Picky compiler... - - * apps/app_speech_utils.c: Better handle timeouts when the - individual speaks after everything has been played but before the - timeout ends. - -2007-02-28 17:15 +0000 [r57049] Steve Murphy <murf@digium.com> - - * pbx/pbx_ael.c: I was surprised that I had not yet downgraded - missing goto targets and macro call defs to a warning, in case - they are in extensions.conf; I rectified this problem. Also, A - goto in a macro to a target in a catch block was not being found; - I fixed this too; the cause was that I needed to treat catch - statements like an extension in the find_match code. - -2007-02-27 17:36 +0000 [r56975] Russell Bryant <russell@digium.com> - - * apps/app_voicemail.c: Fix voicemail email attachments. I missed - the conversion of one of the line endings and there was an extra - one where it should not have been. (issue #9128) - -2007-02-26 22:01 +0000 [r56922] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_lookupcidname.c, apps/app_lookupblacklist.c: Picky, - picky... show deprecation warning in application help, too - (reported via list) - -2007-02-26 20:42 +0000 [r56888] Russell Bryant <russell@digium.com> - - * channels/chan_alsa.c: Restore the behavior of Asterisk 1.2 where - if a device was not specified in alsa.conf, then we just use the - system default, instead of creating our own default of hw:0,0. - (issue #9139) - -2007-02-26 20:07 +0000 [r56856] Joshua Colp <jcolp@digium.com> - - * /, pbx/pbx_config.c: Merged revisions 56850 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r56850 | file | 2007-02-26 15:05:02 -0500 (Mon, 26 Feb 2007) | 2 - lines Obey the clearglobalvars option in extensions reload (or - dialplan reload depending on your version). (issue #9146 reported - by ramonpeek) ........ - -2007-02-26 20:04 +0000 [r56847] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Fix a crash in my last change to - iax2_indicate(). (issue #9150) - -2007-02-26 19:33 +0000 [r56805-56839] Joshua Colp <jcolp@digium.com> - - * apps/app_record.c: Update app_record documentation to use new CLI - command, core show file formats. (issue #9151 reported by junky) - - * main/pbx.c: Use ast_strlen_zero to see if the language and/or - context argument is not present for Background instead of just - checking if it is NULL. (issue #9141 reported by mjagdis) - -2007-02-26 16:51 +0000 [r56785] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Do more complete locking of the - chan_iax2_pvt struct in the indicate callback. (Problem brought - up by Ben Smithurst on the asterisk-dev list) - -2007-02-26 16:36 +0000 [r56783] Joshua Colp <jcolp@digium.com> - - * main/asterisk.c: Allow both of the show version files and core - show file versions CLI commands to work. (issue #9135 reported by - mvanbaak) - -2007-02-26 01:04 +0000 [r56730-56740] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c: Move a comment to be in the correct struct. - - * /: Blocked revisions 56729 via svnmerge ........ r56729 | russell - | 2007-02-25 18:34:31 -0600 (Sun, 25 Feb 2007) | 4 lines Ensure - that lock.h is included in utils.c with AST_API_MODULE defined so - that the implementations will be properly included when the - AST_INLINE_API functions are not going to be inlined. (issue - #9124, festr) ........ - -2007-02-25 14:46 +0000 [r56685] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * main/channel.c, /: Merged revisions 56684 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r56684 | tilghman | 2007-02-25 08:38:03 -0600 (Sun, 25 Feb 2007) - | 3 lines Issue 9130 - If prev is the last item on the channel - list, then evaluating additional conditions (e.g. name prefix) - will cause a NULL dereference. ........ - -2007-02-24 02:02 +0000 [r56569] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Make sure to set a speeddials parent on - creation. Don't crash if hold is pressed when no call is active. - Don't return in places that we shouldn't.. - -2007-02-24 00:53 +0000 [r56548] Kevin P. Fleming <kpfleming@digium.com> - - * codecs/codec_zap.c: update to match zaptel 1.4 API change that - was committed a few minutes ago - -2007-02-23 23:24 +0000 [r56505] Russell Bryant <russell@digium.com> - - * main/asterisk.c, /: Merged revisions 56504 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r56504 | russell | 2007-02-23 17:20:55 -0600 (Fri, 23 Feb 2007) | - 8 lines Fix up a couple more signal handlers to not do bad things - that could cause various undesirable results. The other day, I - made Asterisk deadlock by hitting Control-C because of a bad - signal handler. Now, signal handlers just set a flag and write to - an alert pipe for the flag to be handled. Then, there is another - thread that is monitoring for these flags. If being run in - console mode, it is just the main thread. If Asterisk is in the - background, a thread is created to do it. ........ - -2007-02-23 21:53 +0000 [r56457] Joshua Colp <jcolp@digium.com> - - * main/sched.c: Change log notice to debug. It is possible for a - scheduled item to execute and be deleted at close to the same - time and unavoidable. If this happens this message creeps up. - -2007-02-23 20:20 +0000 [r56407] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 56406 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r56406 | russell | 2007-02-23 14:17:56 -0600 (Fri, 23 Feb 2007) | - 4 lines Don't destroy mutexes before unregistering all of the - entry points from the core. Also, fix a potential memory leak - from not destroying the locks for all of the possible call - numbers (about 32k of them). ........ - -2007-02-23 18:59 +0000 [r56372] Kevin P. Fleming <kpfleming@digium.com> - - * build_tools/make_version_h: build special version strings for - AADK/S800i builds - -2007-02-23 17:58 +0000 [r56341] Russell Bryant <russell@digium.com> - - * apps/app_voicemail.c: The IMAP storage code uses the same code to - build the email that is used when voicemail is sent via email - using something like sendmail. In the patch from bug 8033 to fix - various IMAP storage problems, the line endings in the email file - were changed in the code from "\n" to "\r\n". However, this - breaks sending regular voicemail to email. So, this change - conditionally sets line endings to "\r\n" only if IMAP_STORAGE is - enabled. (issue #9128, patch by jarjarbinks, modified by me to - not break IMAP storage) - -2007-02-22 23:25 +0000 [r56280] Joshua Colp <jcolp@digium.com> - - * /: Blocked revisions 56279 via svnmerge ........ r56279 | file | - 2007-02-22 18:19:25 -0500 (Thu, 22 Feb 2007) | 2 lines Always - defer Agent logoff if any channels are up until they hang up. - (issue #9123 reported by arbrandes) ........ - -2007-02-22 23:08 +0000 [r56277] Russell Bryant <russell@digium.com> - - * configs/sla.conf.sample, main/dial.c, apps/app_meetme.c, - doc/sla.txt: Merge changes from team/russell/sla_updates. This - batch of changes to the SLA code does a few different things. * I - made the SLA code event driven instead of having to act in a lot - of busy loops while dialing things to wait for state changes. - This makes the code more efficient and readable at the same time. - * I have implemented a couple of new features. The first is - inbound trunk ringing timeouts. This is an option that defines - how long to let an incoming call on a trunk to ring. * I have - also implemented ring timeouts for stations. They may be - specified for the entire station, meaning it is how long to let - the station ring before giving up. You can also specify a ring - timeout for a specific trunk on a station. So, you can say that - you only want a specific station to ring 5 seconds if it is line1 - ringing, but otherwise, there is no timeout. - -2007-02-22 18:49 +0000 [r56231] Joshua Colp <jcolp@digium.com> - - * main/channel.c, /, channels/chan_sip.c: Merged revisions 56230 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r56230 | file | 2007-02-22 13:44:24 -0500 (Thu, 22 Feb 2007) | 2 - lines Only change the original or clone channel if it's the - channel behind the proxy channel, not if it's just a regular - bridged channel. ........ - -2007-02-22 14:06 +0000 [r56169] TransNexus OSP Development <support@transnexus.com> - - * doc/osp.txt: Update OSP documentation for v1.4. - -2007-02-22 10:33 +0000 [r56125] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Move message from verbose to debug - -2007-02-22 02:39 +0000 [r56094] Steve Murphy <murf@digium.com> - - * sounds/Makefile: updated the sound tarball versions in Makefile - -2007-02-22 01:24 +0000 [r56011-56055] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Restructure a little bit of code to reduce - nesting. There is no functionality change here. - - * /, channels/chan_sip.c: Merged revisions 56010 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r56010 | russell | 2007-02-21 18:53:25 -0600 (Wed, 21 Feb 2007) | - 3 lines If we receive a frame that is not in any of the - negotiated formats, then drop it. (potentially issue #8781 and - SPD-12) ........ - -2007-02-22 00:35 +0000 [r56008] Joshua Colp <jcolp@digium.com> - - * main/cli.c: Print out deprecation notice on usage output of CLI - commands. (issue #8925 reported by blitzrage) - -2007-02-22 00:08 +0000 [r56006] Kevin P. Fleming <kpfleming@digium.com> - - * main/loader.c: disable unloading of embedded modules... there is - a fundamental problem with doing so that will not be fixed in - this version of Asterisk due to its invasiveness - -2007-02-21 20:35 +0000 [r55957] Joshua Colp <jcolp@digium.com> - - * /, apps/app_meetme.c: Merged revisions 55956 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r55956 | file | 2007-02-21 15:32:16 -0500 (Wed, 21 Feb 2007) | 2 - lines Change naughty warning message to provide useful - information. If a write now fails on a channel in meetme it will - tell you the channel name instead of spitting out the wrong error - message. ........ - -2007-02-21 20:27 +0000 [r55954] Jason Parker <jparker@digium.com> - - * channels/chan_gtalk.c: Fix locking issue, and accept - "transport-accept" as a valid accept message. This should solve - issues 8970 and 8503. - -2007-02-21 20:22 +0000 [r55951] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c: Simplify the last change to app_meetme, and - move the call to dispose_conf() up into the block where we know a - conf exists. - -2007-02-21 20:16 +0000 [r55914-55949] Joshua Colp <jcolp@digium.com> - - * apps/app_meetme.c: Only dispose of the conference if one was - created. - - * apps/app_speech_utils.c: Only start playing the next file if we - have not been quieted. - - * channels/chan_sip.c: Add a flag that indicates whether a SIP - dialog is an outgoing call or not. SIP_OUTGOING originally did it - but it was repurposed to the direction of the last transaction, - which can cause update_call_counter to falsely decrease the wrong - counters. (please don't hurt me oej) (issue #8943 reported by - mdu113) - -2007-02-21 14:06 +0000 [r55869] Kevin P. Fleming <kpfleming@digium.com> - - * /, build_tools/make_version: Merged revisions 55868 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r55868 | kpfleming | 2007-02-21 08:03:11 -0600 (Wed, 21 - Feb 2007) | 2 lines use new tag version script ........ - -2007-02-21 08:32 +0000 [r55834] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #8848 - Turn off lamp more quickly - after transfer (decrement inuse early on transferer's call leg) - -2007-02-21 02:01 +0000 [r55799] Jason Parker <jparker@digium.com> - - * channels/chan_gtalk.c: Fix segfault when buddy couldn't be found. - Issue 7764, patch by sailer - -2007-02-21 01:03 +0000 [r55751-55758] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c: Improve the reference counting to fix bugs - where people report seeing conferences listed that have no - members. (issue #9073) - - * /: Blocked revisions 55750 via svnmerge ........ r55750 | russell - | 2007-02-20 18:19:14 -0600 (Tue, 20 Feb 2007) | 9 lines Fix - random crashes when using the MeetMe application. This patch - converts list handling to use the linked list macros and most - importantly, implements reference counting on the ast_conference - objects. The reference counting was first backported from 1.4. - However, that code has some problems that caused the reference - count to never hit zero. Those problems are fixed in this patch - and will be resolved in 1.4 and trunk next, with a different - patch. (issues #7647, #9073, #9106, BE-115). ........ - -2007-02-21 00:11 +0000 [r55670-55741] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Better handle dropped IMAP connections. - (issue #9054 reported by bsmithurst) - - * channels/chan_sip.c: Return behavior I removed. I did not - remember that you could just add a localnet entry to make it - work. - - * channels/chan_sip.c: Don't test our own address against the - localnet settings. At least one person has had issues as a result - of this from #7051 so I'm reversing it. (issue #8821 reported by - kokoskarokoska) - - * /, channels/chan_agent.c: Merged revisions 55669 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r55669 | file | 2007-02-20 17:39:14 -0500 (Tue, 20 Feb - 2007) | 2 lines Defer clearing callback information if channels - are up until they are hung up. This ensures the hangup process - goes smoothly and no channels get hung in limbo. (issue #8088 - reported by kebl0155) ........ - -2007-02-20 20:26 +0000 [r55589-55634] Russell Bryant <russell@digium.com> - - * main/http.c: Add the Asterisk version information to the Server - header in HTTP responses. (requested by Pari) - - * include/asterisk/manager.h: Increase the maximum number of - manager headers to 128, at the request of Pari. - - * /: Blocked revisions 55588 via svnmerge ........ r55588 | russell - | 2007-02-20 13:49:50 -0600 (Tue, 20 Feb 2007) | 3 lines Convert - a tab to spaces so that the documentation is printed out properly - aligned. ........ - -2007-02-20 16:53 +0000 [r55555] Jason Parker <jparker@digium.com> - - * channels/chan_gtalk.c, res/res_jabber.c: No need to cast nor free - with strdupa (thanks file) 55555! - -2007-02-20 16:41 +0000 [r55553] Russell Bryant <russell@digium.com> - - * configs/sla.conf.sample: Change the formatting of sla.conf.sample - to make it more readable. (issue #9112, blitzrage) - -2007-02-19 21:12 +0000 [r55483] Olle Johansson <oej@edvina.net> - - * res/res_jabber.c: - Not sending arguments to an application is - not "out of memory" - Making error messages a bit more clear - -2007-02-19 18:11 +0000 [r55435] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c, /: Merged revisions 55434 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r55434 | tilghman | 2007-02-19 12:09:09 -0600 (Mon, 19 Feb 2007) - | 2 lines forcename and forcegreetings options should check to - see if the recording already exists ........ - -2007-02-19 14:52 +0000 [r55397] Doug Bailey <dbailey@digium.com> - - * channels/chan_iax2.c: Changed iax2 process thread to detached to - correct memory leak due to left over thread context on thread - exit. Modified module unload process to avoid deadlocks on - pthread cancels - -2007-02-18 12:35 +0000 [r55250-55278] Olle Johansson <oej@edvina.net> - - * /, apps/app_record.c: Merged revisions 55277 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r55277 | oej | 2007-02-18 13:32:13 +0100 (Sun, 18 Feb 2007) | 2 - lines Documentation update (#9053, jsmith) ........ - - * /: Block patch that was made only for 1.2 (already implemented in - 1.4 and trunk) - -2007-02-17 17:39 +0000 [r55219] Joshua Colp <jcolp@digium.com> - - * apps/app_queue.c: Add missing membername option to AddQueueMember - documentation. (issue #9088 reported by seanbright) - -2007-02-17 17:10 +0000 [r55217] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Fix an issue where callerid would not be - displayed on some phones. Issue 8995, initial patch and research - done by wedhorn - -2007-02-17 03:55 +0000 [r55086-55154] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c, /: Merged revisions 55153 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r55153 | file | 2007-02-16 22:53:45 -0500 (Fri, 16 Feb 2007) | 2 - lines Answer the channel before recording privacy information. - (issue #8926 reported by lmamane) ........ - - * apps/app_queue.c: Make the 'i' option of Queue actually work. - (issue #8986 reported by utis) - - * /, channels/chan_sip.c: Merged revisions 55073 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r55073 | file | 2007-02-16 20:09:50 -0500 (Fri, 16 Feb 2007) | 2 - lines Allow chan_sip to handle attended transfers from a SIP - phone that is sitting behind chan_agent. Yes folks, all it took - was one line of code. (issue #8784 reported by pzieba) ........ - -2007-02-17 00:40 +0000 [r55006-55052] Russell Bryant <russell@digium.com> - - * configure, include/asterisk/autoconfig.h.in, configure.ac: If the - pg_config application is found, but there is probably executing - it, then consider postgres unavailable. (issue #8637) - - * codecs/gsm/Makefile: Filter out yet another architecture that - does not work with the optimizations in the built-in libgsm. - (issue 8637, ovi) - - * /, apps/app_meetme.c, configs/meetme.conf.sample: Merged - revisions 55005 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r55005 | russell | 2007-02-16 16:48:22 -0600 (Fri, 16 Feb 2007) | - 9 lines Revert the change I did in revisions 54955, 54969, and - 54970, in 1.2, 1.4, and trunk. I decided that once a conference - is created from meetme.conf, it is acceptable behavior that the - pin can not be changed until the conference goes away. I also - added a note in meetme.conf to describe this behavior. We still - have another issue in 1.4 and trunk where some conferences with - no users don't go away. That is the real bug that needs to be - addressed here. ........ - -2007-02-16 22:18 +0000 [r55002] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_agent.c: Merged revisions 54999 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r54999 | file | 2007-02-16 17:13:45 -0500 (Fri, 16 Feb - 2007) | 2 lines Do not send indications through ast_indicate in - chan_agent but instead go directly to the technology. This way - when indications are emulated they happen on the Agent channel - and do not screw up formats on the channels. (issue #8439 - reported by punkgode) ........ - -2007-02-16 21:12 +0000 [r54969] Russell Bryant <russell@digium.com> - - * /, apps/app_meetme.c: Merged revisions 54955 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r54955 | russell | 2007-02-16 14:56:58 -0600 (Fri, 16 Feb 2007) | - 5 lines For conferences that are configured in meetme.conf, check - the configuration file every time someone joins the conference - instead of only when the conference is first created. This is to - ensure that changes to the pin numbers in the config file are - always honored. (issue #9073) ........ - -2007-02-16 18:51 +0000 [r54924] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c: Need to check macro extension as well as macro - context for directed pickup. - -2007-02-16 18:03 +0000 [r54888-54898] Russell Bryant <russell@digium.com> - - * pbx/pbx_config.c: Fix setting "autofallthrough" to yes by - default. It was set to enabled in pbx.c. However, if the option - was not present in extensions.conf, then pbx_config.c would set - it back to disabled. - - * res/res_features.c: Clean up a few coding guidelines issues - - spaces to tabs, use sizeof() to pass the size of a static buffer, - add spaces ... - -2007-02-16 17:25 +0000 [r54886] Jason Parker <jparker@digium.com> - - * main/asterisk.c: Clarify a restart message. It's silly, but the - reporter had a very valid point. Issue 9079 - -2007-02-16 17:02 +0000 [r54884] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c: Allow directed pickup to pick up the real - context instead of the macro context if a Macro is used. (issue - #8984 reported by jamesb63) - -2007-02-16 12:06 +0000 [r54772-54787] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #7541 - Handle multipart attachments - to SIP messages - even if boundary is quoted. - - * /, res/res_agi.c: Merged revisions 54771 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r54771 | oej | 2007-02-16 12:38:03 +0100 (Fri, 16 Feb 2007) | 2 - lines Issue #9069 - If we open with TH we should not close with - /TD. (seanbright) ........ - -2007-02-16 00:48 +0000 [r54481-54714] Joshua Colp <jcolp@digium.com> - - * apps/app_speech_utils.c: Don't let dtmf leak over into the engine - and let it skew the results... also give DTMF results priority. - (issue #9014 reported by surftek) - - * apps/app_dial.c, /: Merged revisions 54622 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r54622 | file | 2007-02-15 11:14:40 -0500 (Thu, 15 Feb 2007) | 2 - lines Use a separate variable to indicate execution should - continue instead of the return value. (issue #8842 reported by - pluto70) ........ - - * apps/app_dial.c: Forward begin DTMF frames as well as end. (issue - #9068 reported by mhardeman) - -2007-02-14 18:44 +0000 [r54439] Olle Johansson <oej@edvina.net> - - * /: Block patch only needed in 1.2 - -2007-02-14 16:56 +0000 [r54375] Matt Frederickson <creslin@digium.com> - - * channels/chan_zap.c, /: Merged revisions 54373 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r54373 | mattf | 2007-02-14 10:25:49 -0600 (Wed, 14 Feb 2007) | 2 - lines When handling glare on a PRI, move the requested channel - rather than hang up the old one. Fix for 8957 and 9011. ........ - -2007-02-14 01:09 +0000 [r54290] Joshua Colp <jcolp@digium.com> - - * main/channel.c: Add G722 to ast_best_codec. If anyone disagrees - with it's placement, feel free to change it. (issue #9045 - reported by gork) - -2007-02-13 21:31 +0000 [r54204-54235] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Remove a couple of leftover debug messages - - * include/asterisk/devicestate.h: Fix the documentation on the - return values from device state provider registration and - deletion. - - * channels/chan_sip.c: If we fail to create the SIP socket, then - return -1 from reload_config() so that load_module() will return - AST_MODULE_LOAD_DECLINE. Otherwise, the console will just get - spammed with error messages every time chan_sip tries to send a - message. - -2007-02-13 18:41 +0000 [r54180] Olle Johansson <oej@edvina.net> - - * /: Blocking patch for 1.2 only - -2007-02-12 19:17 +0000 [r54066-54103] Russell Bryant <russell@digium.com> - - * main/dial.c, include/asterisk/dial.h: Change - ast_set_state_callback() to ast_dial_set_state_callback() - - * main/dial.c, apps/app_meetme.c, apps/app_page.c, - include/asterisk/dial.h: - Add the ability to register a callback - to monitor state changes in an asynchronous dial operation. - - Rename the various references to "status" to "state" in the dial - API - -2007-02-12 16:34 +0000 [r54026] Joshua Colp <jcolp@digium.com> - - * configure, configure.ac: Make the --without-oss argument work. - (issue #9026 reported by puzzled) - -2007-02-12 15:38 +0000 [r54002] Russell Bryant <russell@digium.com> - - * configs/users.conf.sample: Fix a typo where "vmpassword" should - be "vmsecret" - -2007-02-10 09:09 +0000 [r53878-53881] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/chan_h323.c: Fix VLDTMF reception - - * apps/app_echo.c: Much simpler than previous one ;-) - - * main/channel.c: Provide correct DTMF duration - - * main/cli.c: Bring deprecated 'debug channel <x|all>' command back - -2007-02-10 06:06 +0000 [r53850] Kevin P. Fleming <kpfleming@digium.com> - - * configure, configure.ac, acinclude.m4: don't display the - --with-imap message unless --with-imap was specified without a - path use '-n' instead of '! -z' for tests - -2007-02-10 01:02 +0000 [r53783-53821] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c: Add some output for "show application - SLAStation/SLATrunk" - - * channels/chan_sip.c: Change some text to properly state "On - Hold", which was already done in trunk. - - * configs/sla.conf.sample, include/asterisk/app.h, - include/asterisk/utils.h, main/dial.c, apps/app_meetme.c, - channels/chan_sip.c, doc/sla.txt (added), - include/asterisk/linkedlists.h, include/asterisk/dial.h: Merge - team/russell/sla_rewrite This is a completely new implementation - of the SLA functionality introduced in Asterisk 1.4. It is now - functional and ready for testing. However, I will be adding some - additional features over the next week, as well. For information - on how to set this up, see configs/sla.conf.sample and - doc/sla.txt. In addition to the changes in app_meetme.c for the - SLA implementation itself, this merge brings in various other - changes: chan_sip: - Add the ability to indicate HOLD state in - NOTIFY messages. - Queue HOLD and UNHOLD control frames even if - the channel is not bridged to another channel. linkedlists.h: - - Add support for rwlock based linked lists. dial.c: - Add the - ability to run ast_dial_start() without a reference channel to - inherit information from. - - * apps/app_echo.c: When the Echo() application receives the digit - '#', echo that back as well. Since we already sent the BEGIN - frame for that digit, it makes sense to send the END as well. - -2007-02-09 23:52 +0000 [r53779-53781] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_gtalk.c: another dependency - - * apps/app_adsiprog.c, apps/app_voicemail.c, res/res_config_odbc.c, - funcs/func_odbc.c, res/res_adsi.c: add some inter-module - dependencies - - * build_tools/get_moduleinfo, build_tools/get_makeopts: fix awk - scripts to work when both MODULEINFO and MAKEOPTS are present in - a source file - -2007-02-09 19:33 +0000 [r53749] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c: Temporarily change musicclass on channel to one - specified in Dial so that the 'm' option functions properly. - (issue #8969 reported by christianbee) - -2007-02-09 16:42 +0000 [r53715] Kevin P. Fleming <kpfleming@digium.com> - - * doc/imapstorage.txt, configure, configure.ac: clarify the fact - that voicemail IMAP storage cannot be built against a distro's - binary c-client library package (at least not at this time) - -2007-02-08 23:18 +0000 [r53672] Olle Johansson <oej@edvina.net> - - * main/acl.c: Don't output debug unless we asked for it - -2007-02-08 17:54 +0000 [r53601] Joshua Colp <jcolp@digium.com> - - * apps/app_speech_utils.c: Fix timeout issue when utterance is - longer then timeout itself. - -2007-02-08 13:47 +0000 [r53530-53532] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * main/loader.c: Issue 9007 - Mutex not released on early return - - * apps/app_voicemail.c, /: Merged revisions 53529 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r53529 | tilghman | 2007-02-08 07:36:10 -0600 (Thu, 08 Feb 2007) - | 2 lines Issue 9003 - If fullname is empty, quote() passes back - "\"" ........ - -2007-02-07 23:52 +0000 [r53464-53497] Russell Bryant <russell@digium.com> - - * main/db1-ast/Makefile: When building libdb1.a, put the additional - flags needed at the beginning of ASTCFLAGS, instead of at the - end. This way, we ensure that we find the local headers first - before accidentally trying to use headers that exist in locations - specified in the ASTCFLAGS passed from the main Makefile. (issue - #8637, ovi) - - * main/Makefile: The clean target actually needs to run "distclean" - on editline. This is because we need to make sure that its - configure script gets executed again, because the CFLAGS we want - to pass to editline may have changed. - -2007-02-07 17:53 +0000 [r53434] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: We can not reliably do P2P bridging with DTMF passing - back with compensation if we need to listen for DTMF frames. - (issue #8962 reported by caio1982) - -2007-02-07 17:39 +0000 [r53429] Russell Bryant <russell@digium.com> - - * main/rtp.c: When parsing the NTP timestamp in a sender report - message, you are supposed to take the low 16 bits of the integer - part, and the high 16 bits of the fractional part. However, the - code here was erroneously taking the low 16 bits of the - fractional part. It then shifted the result 16 bits down, so the - result was always zero. This fix makes it grab the appropriate - high 16 bits, instead. (issue #8991, pointed out by - andre_abrantes) - -2007-02-07 17:04 +0000 [r53358-53399] Joshua Colp <jcolp@digium.com> - - * apps/app_playback.c: Directly load say.conf in load_module - instead of calling the reload function. (issue #8946 reported by - junky) - - * /, channels/chan_iax2.c: Merged revisions 53357 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r53357 | file | 2007-02-07 10:38:48 -0500 (Wed, 07 Feb 2007) | 2 - lines Fix a few potential memory leaks with realtime users and - peers. (issue #8999 reported by bsmithurst) ........ - -2007-02-07 15:33 +0000 [r53355] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, apps/app_macro.c: Merged revisions 53354 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r53354 | tilghman | 2007-02-07 09:30:02 -0600 (Wed, 07 Feb 2007) - | 2 lines Issue 7440 - Macro called from Macro from the h - extension exits prematurely ........ - -2007-02-07 09:22 +0000 [r53324] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /: Merged - revisions 52843 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r52843 | crichter | 2007-01-30 15:38:08 +0100 (Di, 30 Jan 2007) | - 1 line fixed some possible segfaults. also fixed an very - important bug which occurs on high load (when calls are very fast - generated) ........ - -2007-02-07 05:24 +0000 [r53246-53294] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * res/res_jabber.c: Text fix for jabber reload command (reported by - bkruse via IRC) - - * main/manager.c, /: Merged revisions 53245 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r53245 | tilghman | 2007-02-06 00:58:28 -0600 (Tue, 06 Feb 2007) - | 2 lines Issue 8987 - Status could return two responses - (mnicholson) ........ - -2007-02-05 23:43 +0000 [r53222] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Formatting - -2007-02-05 17:06 +0000 [r53150-53152] Joshua Colp <jcolp@digium.com> - - * apps/app_playback.c: Ensure say_cfg is NULL when the module is - loaded. (issue #8946 reported by junky) - - * apps/app_playback.c: Unregister Playback CLI commands as well as - dialplan application. (issue #8946 reported by junky) - -2007-02-05 00:18 +0000 [r53143] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Add some comments on queue system behaviour - and how it affects the SIP channel - -2007-02-03 21:05 +0000 [r53138] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Make SIPDtmfMode application work with - recent capability changes, and also fix an RTP stack issue when - the auto option was used. (issue #8972 reported by mdu113) - -2007-02-03 20:44 +0000 [r53135-53136] Russell Bryant <russell@digium.com> - - * apps/app_dial.c, /: Merged revisions 53133 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r53133 | russell | 2007-02-03 14:38:13 -0600 (Sat, 03 Feb 2007) | - 4 lines set the DIALSTATUS variable to contain "INVALIDARGS" when - the dial application exits early because of invalid arguments - instead of just leaving it empty. (issue #8975) ........ - - * /: Blocked revisions 53134 via svnmerge ........ r53134 | russell - | 2007-02-03 14:39:45 -0600 (Sat, 03 Feb 2007) | 2 lines Revert - some changes that accidentally got committed as a part of another - fix. ........ - -2007-02-03 10:02 +0000 [r53131] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/h323/ast_h323.cxx: Remove quote from H.323 vendor string - because due to compatibilities with CS1000 reported at - www.voip-info.org - -2007-02-02 21:26 +0000 [r53129] BJ Weschke <bweschke@btwtech.com> - - * UPGRADE.txt, apps/app_queue.c: I'm baaaaaaaaaack. :) Post a - warning to the console that things might possibly be - misconfigured when queue member's states are still 'Not in Use' - when we're about to bridge them with a caller from queue. Also, - put some documentation quoted from oej's queues.txt efforts - started in /trunk today. This commit puts #7433 into feedback - state for 1.4, and pending no further negative feedback, it will - finally be closed. - -2007-02-02 17:15 +0000 [r53114-53120] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Correct a copy/pasted error message line for RTCP. - - * main/config.c, /: Merged revisions 53117 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r53117 | file | 2007-02-02 10:58:09 -0600 (Fri, 02 Feb 2007) | 2 - lines Pass the glob expanded filename to process_text_line so - that error messages contain the actual filename, not the original - include one. (issue #8959 reported by tzafrir) ........ - - * Makefile: Add systemname to asterisk.conf generation per recent - discussions about it. (issue #8968 reported by blitzrage) - -2007-02-02 00:24 +0000 [r53109] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c, configs/sip.conf.sample: Disable the direct - p2p RTP call setup in SIP. You can enable it in sip.conf, but it - is now considered experimental until we solve the - AST_CONTROL_ANSWER with payload and videocaps stuff. - -2007-02-01 23:16 +0000 [r53108] Jason Parker <jparker@digium.com> - - * /: Blocked revisions 53107 via svnmerge ........ r53107 | qwell | - 2007-02-01 17:14:09 -0600 (Thu, 01 Feb 2007) | 2 lines Fix a - small typo. Synopsis lines shouldn't have a newline ........ - -2007-02-01 22:24 +0000 [r53097-53104] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 53103 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r53103 | file | 2007-02-01 16:21:56 -0600 (Thu, 01 Feb 2007) | 2 - lines Copy noncodeccapability over to the joint variable so that - telephone-event will get transmitted in the sent INVITE. ........ - - * main/db1-ast/hash/hash.c: Huh... fix the berkeley DB to compile - here as well, but it apparently required both dev mode and no - optimizations to creep up. - - * /, channels/chan_sip.c: Merged revisions 53095 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r53095 | file | 2007-02-01 15:47:11 -0600 (Thu, 01 Feb 2007) | 2 - lines Don't negotiate RFC2833 when not configured to do so. - (issue #8799 reported by mdu113) ........ - -2007-02-01 21:24 +0000 [r53093] Russell Bryant <russell@digium.com> - - * funcs/func_strings.c: Fix the FIELDQTY function to not crash. - (reported by blitzrage and Corydon on IRC) - -2007-02-01 21:15 +0000 [r53091] Olle Johansson <oej@edvina.net> - - * /: Going backwards, blame file. - -2007-02-01 21:11 +0000 [r53086-53088] Joshua Colp <jcolp@digium.com> - - * /, res/res_musiconhold.c: Merged revisions 53084 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r53084 | file | 2007-02-01 15:03:10 -0600 (Thu, 01 Feb - 2007) | 2 lines Return previous behavior of having MOH pick up - where it was left off. (issue #8672 reported by sinistermidget) - ........ - - * funcs/func_strings.c: Make func_strings build under dev mode. - Didn't I do this today already in the berkeley DB? - -2007-02-01 21:05 +0000 [r53079-53085] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: - Clean INC_COUNT flag when we decrement - call counter - If it's still set at time of dialog destruction, - make sure we decrement the device call counter properly before we - destroy the dialog - - * apps/app_queue.c: Change debug level for state change message - that is not really informative when debugging app_queue - - * channels/chan_sip.c: Cleaning up the devicestate callback - function - -2007-02-01 20:13 +0000 [r53075-53077] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * funcs/func_strings.c: Oops. - - * /, funcs/func_strings.c: Merged revisions 53074 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r53074 | tilghman | 2007-02-01 14:07:35 -0600 (Thu, 01 Feb 2007) - | 2 lines Bug 8965 ........ - -2007-02-01 19:33 +0000 [r53072] Joshua Colp <jcolp@digium.com> - - * main/asterisk.c: Add missing 'F' letter to getopt so it magically - becomes a valid option. (issue #8960 reported by tzafrir) - -2007-02-01 19:21 +0000 [r53070] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * main/pbx.c, /, funcs/func_strings.c: Merged revisions 53069 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r53069 | tilghman | 2007-02-01 13:13:53 -0600 (Thu, 01 Feb 2007) - | 2 lines No wonder FIELDQTY doesn't work with functions... the - documentation in pbx.c was wrong ........ - -2007-02-01 17:37 +0000 [r53064] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Fix silly logic. We really want to write - UDPTL frames out when the call is up. - -2007-02-01 16:35 +0000 [r53062] Olle Johansson <oej@edvina.net> - - * configs/sip.conf.sample: Add explanation of port= in combination - with defaultip= (thanks jsmith) - -2007-02-01 13:17 +0000 [r53060] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: we update the name on any first reply of - our setup - -2007-02-01 11:07 +0000 [r53057] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/chan_h323.c: chan_h323 is very stable, so let it built - by default - -2007-02-01 00:24 +0000 [r53050-53052] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: When going on hold have the side that was put on hold - reinvite back to Asterisk. When going off hold have the side that - was taken off hold reinvited back to the other party. - - * main/rtp.c: Add more frame types to forward in the RTP bridge - loops. - -2007-01-31 21:32 +0000 [r52859-53046] Russell Bryant <russell@digium.com> - - * main/cdr.c, main/manager.c, pbx/pbx_spool.c, - channels/chan_skinny.c, channels/chan_h323.c, main/http.c, - pbx/pbx_dundi.c, apps/app_rpt.c, channels/chan_mgcp.c, - main/pbx.c, channels/chan_zap.c, /, apps/app_meetme.c, - channels/chan_sip.c, apps/app_queue.c, channels/chan_iax2.c: - Merged revisions 53045 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r53045 | russell | 2007-01-31 15:25:11 -0600 (Wed, 31 Jan 2007) | - 3 lines Fix a bunch of places where pthread_attr_init() was - called, but pthread_attr_destroy() was not. ........ - - * apps/app_userevent.c: Remove an extra \r\n from manager user - events. (issue #8955, mnicholson) - - * main/rtp.c, /: Merged revisions 53039 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r53039 | russell | 2007-01-31 11:41:51 -0600 (Wed, 31 Jan 2007) | - 3 lines Use the proper format string to print unsigned values in - the rtp debug output. (issue #8954, wmis) ........ - - * apps/app_queue.c: Only changed the paused status in an existing - queue member if the paused column exists. - - * apps/app_queue.c: Instead of always creating a realtime queue - member as unpaused, read the "paused" column and use that value - for the paused status of the member. (issue #8949, jmls) - - * contrib/init.d/rc.suse.asterisk: Update init script for SuSE 10. - (issue #8363, johnlange) - - * doc/cdrdriver.txt: Add documentation for using cdr_pgsql. (issue - #8942, lters) - - * configure, include/asterisk/autoconfig.h.in, configure.ac, - codecs/codec_gsm.c: When we are checking for a system installed - version of libgsm, we need to check for gsm.h as well. - Furthermore, when checking for this header, it may be located in - a gsm/ sub directory, so check for that, as well. (issue #8773) - - * /: Blocked revisions 52954 via svnmerge ........ r52954 | russell - | 2007-01-30 13:41:52 -0600 (Tue, 30 Jan 2007) | 4 lines Don't - print a message indicating that we don't know what to do with a - proceeding control frame in ast_request_and_dial(). We just need - to ignore it. (reported by JerJer on #asterisk-dev) ........ - - * channels/chan_sip.c: Only set the DTMF flag on the rtp structure - if the DTMF mode is actually RFC2833, not just that it is not - INFO. This makes it get set for inband DTMF as well, which is not - valid. (issue #8936) - - * main/asterisk.c, /: Merged revisions 52903 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r52903 | russell | 2007-01-30 11:12:04 -0600 (Tue, 30 Jan 2007) | - 9 lines The SIGHUP handler was implemented to allow admins to - send SIGHUP to a running Asterisk process to reload the - configuration. However, doing the actual reload in the signal - handler itself is a very bad thing to do, because the reload - process includes calling non-reentrant functions such as - malloc/calloc/etc. If Asterisk is running in the background, then - the reload will happen immediately. However, if running in - console mode, the reload doesn't work until something is typed at - the console. That sort of defeats the purpose, but I don't see an - easy way to get around it at this point. ........ - - * /: Blocked revisions 52857 via svnmerge ........ r52857 | russell - | 2007-01-30 09:35:23 -0600 (Tue, 30 Jan 2007) | 5 lines Comment - out the parts in the Makefile that make codec_zap get built. It - will not yet build against zaptel 1.2, so I am disabling it to - prevent further bug reports until it gets merged. (issue #8940) - ........ - -2007-01-30 15:29 +0000 [r52856] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Drop the deprecated show commands since the - original ones were changed back. (issue #8937 reported by - PCadach) - -2007-01-30 08:46 +0000 [r52807-52809] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/chan_h323.c: Revert reprecation of h.323 gk cycle - command from pre-1.4 version instead of duplicated h323 cycle gk - - * res/res_odbc.c: Don't play with free()'d pointers - - * configure, acinclude.m4: Handle non-standard OpenH323/PWLib - library names - -2007-01-30 00:15 +0000 [r52763] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 52762 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r52762 | russell | 2007-01-29 18:15:06 -0600 (Mon, 29 Jan 2007) | - 5 lines Fix the extraction of the timestamp from video frames. It - was using the mapping for a mini-frame instead of a video-frame, - which caused it to get invalid data. (issue #8795, mihai) - ........ - -2007-01-29 23:43 +0000 [r52717] Joshua Colp <jcolp@digium.com> - - * apps/app_mixmonitor.c, /: Merged revisions 52716 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r52716 | file | 2007-01-29 18:39:39 -0500 (Mon, 29 Jan - 2007) | 2 lines Now that filename is part of the structure and - since it comes before postprocess... we have to add it to our - postprocess line. (reported on asterisk-dev by Boris Bakchiev) - ........ - -2007-01-29 22:58 +0000 [r52688-52695] Russell Bryant <russell@digium.com> - - * main/Makefile: Add a missing quotation mark. This was pointed out - by jcmoore on #asterisk-dev. - - * main/manager.c: Remove a recursive lock of the manager session. - This was pointed out by zandbelt in issue #8711. - -2007-01-29 22:12 +0000 [r52679] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * pbx/pbx_config.c: Argument number correction - -2007-01-29 21:36 +0000 [r52611-52647] Russell Bryant <russell@digium.com> - - * main/Makefile: ASTLDFLAGS needs to be passed to the editline - configure script as LDFLAGS. (issue #8928, zandbelt) - - * main/rtp.c: Fix a problem with packet-to-packet bridging and DTMF - mode translation. P2P bridging can only be used when the DTMF - modes don't match if the core is monitoring DTMF in both - directions. Then, the core will handle the translation. - Otherwise, this bridging method can not be used. (issue #8936) - - * main/manager.c: The session lock can not be held while calling - action callbacks. If so, then when the WaitEvent callback gets - called, then no event can happen because the session can't be - locked by another thread. Also, the session needs to be locked in - the HTTP callback when it reads out the output string. This fixes - the deadlock reported in both 8711 and 8934. Regarding issue - 8711, there still may be an issue. If there is a second action - requested before the processing of the first action is finished, - there could still be some corruption of the output string buffer - used to build the result. (issue #8711, #8934) - -2007-01-29 18:59 +0000 [r52572] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Use ast_calloc instead of malloc. - -2007-01-29 17:57 +0000 [r52535] Steve Murphy <murf@digium.com> - - * apps/app_voicemail.c, main/say.c: this is for 8778 (pt_BR - backport to 1.4). It was committed to trunk via 7663. But it - wasn't so much an enhancement as a fix for the bad language - output for portuguese in Brazil, so, after a lot of prodding from - patient Brazilians, here is the same fix for 1.4 - -2007-01-29 17:33 +0000 [r52523] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Set quota information to 0 when creating a - vm_state. (issue #8924 reported by neutrino88) - -2007-01-29 16:54 +0000 [r52506] Russell Bryant <russell@digium.com> - - * main/jitterbuf.c, include/jitterbuf.h: Clean up a few things in - the last commit to the adaptive jitterbuffer code. - Specifically - indicate to the compiler that the "dropem" variable only needs - one but. - Change formatting to conform to coding guidelines. - -2007-01-29 04:18 +0000 [r52494] Jim Dixon <telesistant@hotmail.com> - - * main/jitterbuf.c, include/jitterbuf.h: Fixed problem with - jitterbuf, whereas it would not complain about, and would allow - itself to be overfilled (per the max_jitterbuf parameter). Now it - rejects any data over and above that size, and complains about - it. - -2007-01-28 05:15 +0000 [r52462] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * configure, configure.ac: Suggested change to fix normal usage of - --with-tds=/usr/local (Sean Bright, via asterisk-dev mailing - list) - -2007-01-27 02:13 +0000 [r52335-52416] Joshua Colp <jcolp@digium.com> - - * /, apps/app_queue.c: Merged revisions 52415 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r52415 | file | 2007-01-26 21:09:10 -0500 (Fri, 26 Jan 2007) | 2 - lines Make COMPLETECALLER and COMPLETEAGENT output to queue_log - follow documentation. (issue #7677 reported by amilcar) ........ - - * main/manager.c: Have the manager interface send back an "Already - logged in" message instead of "Invalid/Unknown Command" when the - client authenticates for a second time. (issue #8509 reported by - pari) - - * /, channels/chan_iax2.c: Merged revisions 52360 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r52360 | file | 2007-01-26 19:03:23 -0500 (Fri, 26 Jan 2007) | 2 - lines Make the last context entry read in the dominant one. - (issue #8918 reported by pj) ........ - - * main/file.c: Fix core show file formats CLI command. - -2007-01-25 19:18 +0000 [r52163-52265] Joshua Colp <jcolp@digium.com> - - * /, main/jitterbuf.c: Merged revisions 52264 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r52264 | file | 2007-01-25 14:15:29 -0500 (Thu, 25 Jan 2007) | 2 - lines Allow dequeueing of frames with negative timestamp by - moving jitterbuffer frames check to jb_next. (issue #8546 - reported by harmen) ........ - - * channels/chan_sip.c: Drop out variables I accidentally put in. - - * channels/chan_sip.c: Decrement onHold count if we are hung up on - and still on hold. (issue #8909 reported by alexh42) - - * apps/app_mixmonitor.c, /: Merged revisions 52162 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r52162 | file | 2007-01-24 20:48:52 -0500 (Wed, 24 Jan - 2007) | 2 lines Add another note about audio files being played - back to each bridged party. (issue #8718 reported by ppyy) - ........ - -2007-01-25 01:37 +0000 [r52107-52160] Russell Bryant <russell@digium.com> - - * apps/app_voicemail.c, configs/users.conf.sample: By suggestion - from kpfleming last week, change "vmpassword" to "vmsecret". - - * configure, configure.ac: Remove libnsl as a required lib for - libiksemel to work. This change was already made in the trunk. - (issue #8762) - - * /: Blocked revisions 52137 via svnmerge ........ r52137 | russell - | 2007-01-24 18:39:50 -0600 (Wed, 24 Jan 2007) | 3 lines Fix a - seg fault when running this application with no arguments from - AGI. (issue #8905, junky) ........ - - * include/asterisk/dial.h: Fix the formatting of doxygen comments - to properly indicate that the comment documents the previous - entity, as opposed to the next one. - -2007-01-24 18:26 +0000 [r52052] Steve Murphy <murf@digium.com> - - * utils/check_expr.c, utils/Makefile, /: Merged revisions 52002 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r52002 | murf | 2007-01-24 10:43:50 -0700 (Wed, 24 Jan 2007) | 1 - line updated check_expr via 8322 (refactoring of expression - checking impl); elfring contributed a nice code reorg, I - contributed some time to get it working again, better messages - ........ - -2007-01-24 18:20 +0000 [r52016-52049] Joshua Colp <jcolp@digium.com> - - * main/dial.c (added), apps/app_page.c, main/Makefile, - include/asterisk/dial.h (added): Merge in dialing API and the - app_page that uses it. (issue #BE-118) - - * channels/chan_sip.c: Fix changing channel formats when joint - capability changes and there are no audio formats... I didn't - break it originally! (issue #8535 reported by ivoc) - -2007-01-24 17:14 +0000 [r52000] Russell Bryant <russell@digium.com> - - * configure: rebuild configure script to reflect last chan_h323 - related changes. - -2007-01-24 12:57 +0000 [r51979-51989] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: added fix from #8899 - - * channels/chan_misdn.c, /: Merged revisions 51966 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r51966 | crichter | 2007-01-24 11:48:09 +0100 (Mi, 24 - Jan 2007) | 1 line fixed the busy problem (dialstatus was not - busy when we called a busy extension) ........ - -2007-01-24 09:30 +0000 [r51931] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Show capabilities *and* preference in - general settings in "sip show settings" (reported by Clona/Telio - - Thanks!) - -2007-01-24 08:04 +0000 [r51895] Paul Cadach <paul@odt.east.telecom.kz> - - * acinclude.m4: Allow x64 builds of H.323 (please, rebuild - configure) - -2007-01-24 00:59 +0000 [r51829-51848] Russell Bryant <russell@digium.com> - - * main/channel.c, /: Merged revisions 51843 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r51843 | russell | 2007-01-23 18:57:28 -0600 (Tue, 23 Jan 2007) | - 6 lines Fix an issue related to synchronization of recordings - when using Monitor(). The bug is a miscalculation of the amount - to seek the stream for writing to disk when the number of samples - coming in and out of a channel do not match up. (issue #8298, - #8887, report and patch by guillecabeza, patch files created and - testing done by whoiswes) ........ - - * apps/app_while.c, /: Merged revisions 51828 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r51828 | russell | 2007-01-23 18:17:50 -0600 (Tue, 23 Jan 2007) | - 4 lines Don't set a new value for the END_ variable on the - channel before using the old value. If you do, it will lead to - accessing a memory address that has been free()'d. (issue #8895, - arkadia) ........ - -2007-01-23 22:46 +0000 [r51788] Joshua Colp <jcolp@digium.com> - - * channels/chan_oss.c, channels/chan_phone.c, channels/chan_zap.c, - channels/chan_sip.c, channels/chan_skinny.c, - channels/chan_features.c, channels/chan_alsa.c, - channels/chan_gtalk.c, channels/chan_iax2.c: Update channel - drivers to use module referencing so that unloading them while in - use will not result in crashes. (issue #8897 reported by junky) - -2007-01-23 22:04 +0000 [r51750-51781] Russell Bryant <russell@digium.com> - - * main/manager.c: Fix some bugs in process_message(). The manager - session lock needs to be held when sending some sort of response, - or calling one of the manager action callbacks. This resolves an - issue where people using the GUI would get random crashes when - they start clicking around a lot. (issue #8711, reported and - debugged by zandbelt) - - * main/http.c: Fix setting the default port of 8088 on 64-bit or - big-endian machines. - - * main/manager.c: When traversing the list of manager actions, the - iterator needs to be initialized to the list head *after* locking - the list. Also, lock the actions list in one place it is being - accessed where it was not being done. - -2007-01-23 20:32 +0000 [r51683-51716] Steve Murphy <murf@digium.com> - - * res/res_features.c: this mod from 8593 (dstchannel in cdr is - empty when transfer call). - - * main/callerid.c: via 8748 (callerid.c loses name when returning - PRIVATE_NUMBER flag), the user suggested this mod, saying it - would allow 'WITHHELD' to appear in the name field, which would - be useful - -2007-01-23 10:28 +0000 [r51648-51649] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/chan_misdn.c, /, - channels/misdn/isdn_msg_parser.c: Merged revisions 50495,50506 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r50495 | crichter | 2007-01-11 14:27:52 +0100 (Do, 11 Jan 2007) | - 6 lines * more additions to make the RESTART message work * added - fix for misdn_call to allow SETUPs with empty extensions, - replaced the strtok_r functions with strsep for that (inspired by - Sandro Cappellazzo, thanks) ........ r50506 | crichter | - 2007-01-11 15:45:38 +0100 (Do, 11 Jan 2007) | 1 line when we get - L2 UP, the L1 is UP definitely too, so we set the L1 state up as - well. ........ - - * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, - channels/chan_misdn.c: manually merged r49922 and r50335, because - of conflicts. this commint includes addition of the ISDN RESTART - Message - -2007-01-23 06:51 +0000 [r51615] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/chan_h323.c, channels/Makefile: Do not abort Asterisk - startup if h323 configuration file not found (reported by - mithraen) - -2007-01-23 03:00 +0000 [r51513-51558] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Only change audio formats on the channel if - we have an audio format to change to. (issue #8535 reported by - ivoc) - - * /, res/res_musiconhold.c: Merged revisions 51512 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r51512 | file | 2007-01-22 20:41:35 -0500 (Mon, 22 Jan - 2007) | 2 lines Yield before reading from zaptel timing source - under Solaris so that other threads get a chance to do things. - (issue #7875 reported by bob) ........ - -2007-01-22 19:41 +0000 [r51411] Russell Bryant <russell@digium.com> - - * /: Blocked revisions 51410 via svnmerge ........ r51410 | russell - | 2007-01-22 13:39:30 -0600 (Mon, 22 Jan 2007) | 3 lines Merge - codec_zap support for the transcoder card. This is a standalone - codec module so it will not affect anything else. ........ - -2007-01-22 19:28 +0000 [r51409] Steve Murphy <murf@digium.com> - - * pbx/pbx_ael.c: This fixes 8836, according to dnatural - -2007-01-22 19:13 +0000 [r51360-51407] Joshua Colp <jcolp@digium.com> - - * apps/app_mixmonitor.c, /: Merged revisions 51406 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r51406 | file | 2007-01-22 14:08:52 -0500 (Mon, 22 Jan - 2007) | 2 lines Move filestream creation to Mixmonitor loop. This - will prevent a blank file from being created if no frames ever - pass through to be recorded. (issue #7589 reported by - steve_mcneil) ........ - - * /: Blocked revisions 51359 via svnmerge ........ r51359 | file | - 2007-01-22 11:23:03 -0500 (Mon, 22 Jan 2007) | 2 lines Explicitly - declare what codecs are supported by default globally since using - a bitmask for all may include ones we don't need. (issue #8357 - reported by gknispel_proformatique) ........ - -2007-01-20 06:53 +0000 [r51348-51350] Jason Parker <jparker@digium.com> - - * configs/say.conf.sample: Fix Italian numeral support in say.conf - for "_[2-9]00" case. "2131" would've translated to something - along the lines of (pardon my..Italian {or lack thereof}) - "duecentocentotrentuno", which makes no sense at all. - - * configs/say.conf.sample: Fix German language support in say.conf - Properly support 21, 31, 41, 51, 61, 71, 81, and 91. - einundzwanzig has the same format as zweiundzwanzig (as do all - other "_ZX" spoken numerals) Fix support for numbers in the - 10,000,000 to 99,999,999 range. Add support for numbers in the - 100,000,000 to 999,999,999 range. - -2007-01-20 00:13 +0000 [r51302-51343] Russell Bryant <russell@digium.com> - - * apps/app_meetme.c: Remove an unused instance of an unnamed enum. - - * apps/app_meetme.c: Remove another duplicated definition - - * apps/app_meetme.c: Remove a variable that was declared twice. - - * codecs/gsm/Makefile: Add a couple more processors that need - optimizations excluded. (issue #8637) - - * channels/chan_gtalk.c: Fix VLDTMF support in chan_gtalk. - AST_FRAME_DTMF and AST_FRAME_DTMF_END are actually the same - thing. So, a digit would have been interpreted incorrectly here. - Since the channel driver will always have the begin and end - callbacks called for a digit, only support the button-down and - button-up messages. - - * .cleancount: Bump the cleancount since my last commit changed the - channel structure. - - * channels/chan_oss.c, main/rtp.c, main/channel.c, - channels/chan_phone.c, channels/chan_misdn.c, - channels/chan_skinny.c, channels/chan_features.c, - channels/chan_h323.c, channels/chan_alsa.c, channels/chan_mgcp.c, - channels/chan_zap.c, channels/chan_local.c, main/frame.c, - channels/chan_sip.c, channels/chan_agent.c, - include/asterisk/channel.h, channels/chan_gtalk.c, - channels/chan_iax2.c: Merge the changes from the - /team/group/vldtmf_fixup branch. The main bug being addressed - here is a problem introduced when two SIP channels using SIP INFO - dtmf have their media directly bridged. So, when a DTMF END frame - comes into Asterisk from an incoming INFO message, Asterisk would - try to emulate a digit of some length by first sending a DTMF - BEGIN frame and sending a DTMF END later timed off of incoming - audio. However, since there was no audio coming in, the DTMF_END - was never generated. This caused DTMF based features to no longer - work. To fix this, the core now knows when a channel doesn't care - about DTMF BEGIN frames (such as a SIP channel sending INFO - dtmf). If this is the case, then Asterisk will not emulate a - digit of some length, and will instead just pass through the - single DTMF END event. Channel drivers also now get passed the - length of the digit to their digit_end callback. This improves - SIP INFO support even further by enabling us to put the real - digit duration in the INFO message instead of a hard coded 250ms. - Also, for an incoming INFO message, the duration is read from the - frame and passed into the core instead of just getting ignored. - (issue #8597, maybe others...) - - * main/asterisk.c: Merged revisions 51300 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r51300 | russell | 2007-01-19 10:44:09 -0600 (Fri, 19 Jan 2007) | - 4 lines Fix a memory leak on command line tab completion. The - container for the matches was freed, but the individual matches - themselves were not. (issue #8851, arkadia) ........ - -2007-01-19 00:17 +0000 [r51272-51274] Dwayne M. Hubbard <dhubbard@digium.com> - - * channels/chan_zap.c: chan_zap compiles without libpri after - committing 7877 patch - - * channels/chan_zap.c, /: Merged revisions 51271 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r51271 | dhubbard | 2007-01-18 17:47:10 -0600 (Thu, 18 Jan 2007) - | 3 lines issue 7877: chan_zap module reload does not use - default/initialized values on subsequent loads. Reset - configuration variables to default values prior to parsing - configuration file. ........ - -2007-01-18 23:36 +0000 [r51270] Kevin P. Fleming <kpfleming@digium.com> - - * /: block this patch since it is already here - -2007-01-18 22:50 +0000 [r51265] Jason Parker <jparker@digium.com> - - * apps/app_voicemail.c, main/channel.c, main/pbx.c, - funcs/func_strings.c, main/app.c: Add some more checks for - option_debug before ast_log(LOG_DEBUG, ...) calls. Issue 8832, - patch(es) by tgrman - -2007-01-18 21:54 +0000 [r51262] Russell Bryant <russell@digium.com> - - * Makefile, configure, main/Makefile, acinclude.m4, makeopts.in: - Ensure that the locations given to the Asterisk configure script - for ncurses, curses, termcap, or tinfo are further passed along - to the editline configure script. This fixes some - cross-compilation environments. (issue #8637, reported by ovi, - patch by me) - -2007-01-18 21:14 +0000 [r51256] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, main/stdtime/localtime.c: Merged revisions 51255 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r51255 | tilghman | 2007-01-18 15:11:34 -0600 (Thu, 18 - Jan 2007) | 2 lines If a timezone is not specified, assume - localtime (instead of gmtime) (Issue #7748) ........ - -2007-01-18 19:17 +0000 [r51251] Joshua Colp <jcolp@digium.com> - - * apps/app_speech_utils.c: Only start timeout once we reach the end - of the files to play back. - -2007-01-18 18:42 +0000 [r51245] Jason Parker <jparker@digium.com> - - * main/cli.c: Fix an issue with file name completion in "module - load" and "load". Issue 8846 - -2007-01-18 18:36 +0000 [r51243] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Copy MOH settings when calling a peer so - that if they put someone on hold or get put on hold themselves - they get the right music class. (issue #8840 reported by mdu113) - -2007-01-18 18:28 +0000 [r51241] Jason Parker <jparker@digium.com> - - * main/channel.c: Fix an issue with deprecated commands - -2007-01-18 17:49 +0000 [r51236] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * contrib/scripts/vmdb.sql, /: Merged revisions 51235 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r51235 | tilghman | 2007-01-18 11:42:17 -0600 (Thu, 18 - Jan 2007) | 2 lines Document all the fields, including the - indication that "uniqueid" should not be renamed. ........ - -2007-01-18 17:18 +0000 [r51233] Russell Bryant <russell@digium.com> - - * main/manager.c: Make the "hasmanager" option in users.conf - actually have an effect. (issue #8740, LnxPrgr3) - -2007-01-18 00:48 +0000 [r51211-51213] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Build the IMAP remote directory string - better and properly. Fix an issue with encoding the GSM voicemail - when attaching to the voicemail. (issue #8808 reported by - akohlsmith) - - * main/rtp.c: Pass data as well for hold/unhold/vidupdate frames. - (issue #8840 reported by mdu113) - -2007-01-17 23:31 +0000 [r51198-51205] Russell Bryant <russell@digium.com> - - * funcs/func_odbc.c: Fix some instances where when loading - func_odbc, a double-free could occur. Also, remove an unneeded - error message. If the failure condition is actually a memory - allocation failure, a log message will already be generated - automatically. - - * channels/chan_zap.c: Instead of dividing the offset by 2 - directly, make it more clear that the offset is being scaled by - the size of the elements in the buffer. (Inspired by a discussing - on the asterisk-dev list about this code) - - * /, channels/chan_sip.c: Merged revisions 51197 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r51197 | russell | 2007-01-17 15:17:21 -0600 (Wed, 17 Jan 2007) | - 3 lines Move the check for a failure of ast_channel_alloc() to - before locking the pvt structure again. Otherwise, on a failure, - this will cause a deadlock. ........ - -2007-01-17 20:56 +0000 [r51195] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, main/utils.c: Merged revisions 51194 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r51194 | tilghman | 2007-01-17 14:52:21 -0600 (Wed, 17 Jan 2007) - | 4 lines When ast_strip_quoted was called with a zero-length - string, it would treat a NULL as if it were the quoting character - (and would thus return the string in memory immediately following - the passed-in string). ........ - -2007-01-17 17:36 +0000 [r51186] Jason Parker <jparker@digium.com> - - * apps/app_voicemail.c: re-add "password" for realtime voicemail - -2007-01-17 06:36 +0000 [r51182] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Return the correct result when directly writing out a - packet so that the core doesn't then decide to handle it the - regular way again. (issue #8833 reported by rcourtna) - -2007-01-17 01:29 +0000 [r51176] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_voicemail.c: a few more coding style cleanups and one - bug fix (from AnthonyL) - -2007-01-17 00:46 +0000 [r51172] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Move rescheduling of lagrq/pings into the - scheduler callback. - -2007-01-17 00:20 +0000 [r51165-51170] Jason Parker <jparker@digium.com> - - * main/rtp.c: Fix issue with dtmf continuation packets when the - dtmf digit is 0... Issue 8831 - - * apps/app_voicemail.c, contrib/scripts/vmdb.sql: Fix an issue with - IMAP storage and realtime voicemail. Also update the vmdb sql - script for IMAP specific options. Issue 8819, initial patches by - bsmithurst (slightly modified by me) - - * doc/voicemail_odbc_postgresql.txt: change documentation to - reflect new procedure in 1.4/trunk - -2007-01-16 21:51 +0000 [r51159-51162] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, doc/voicemail_odbc_postgresql.txt (added): Merged revisions - 51161 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r51161 | tilghman | 2007-01-16 15:50:04 -0600 (Tue, 16 Jan 2007) - | 2 lines Add documentation walkthrough on getting Postgres to - work with voicemail (from Issue 8513) ........ - - * apps/app_voicemail.c, /: Merged revisions 51158 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r51158 | tilghman | 2007-01-16 15:26:06 -0600 (Tue, 16 Jan 2007) - | 2 lines Postgres driver doesn't like a NULL pointer when - retrieving the length (Bug 8513) ........ - -2007-01-16 17:46 +0000 [r51150] Matt O'Gorman <mogorman@digium.com> - - * apps/app_voicemail.c: minor things i missed before i get jumped - on - -2007-01-16 17:39 +0000 [r51148] Joshua Colp <jcolp@digium.com> - - * /, res/res_features.c: Merged revisions 51145 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r51145 | file | 2007-01-16 12:36:50 -0500 (Tue, 16 Jan 2007) | 2 - lines Return previous behavior. ParkedCalls will be able to do - DTMF based transfers again. trunk however will get an option to - allow this to be set on/off. (issue #8804 reported by nortex) - ........ - -2007-01-16 17:36 +0000 [r51146] Jason Parker <jparker@digium.com> - - * main/file.c: Display more useful output when streaming files. - Include the channel name to which the file is being played. Issue - 8828, patch by junky. - -2007-01-16 05:55 +0000 [r51087] Joshua Colp <jcolp@digium.com> - - * channels/chan_zap.c, /: Merged revisions 51085 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r51085 | file | 2007-01-16 00:53:31 -0500 (Tue, 16 Jan 2007) | 2 - lines Add none as a valid callgroup/pickupgroup option. I - consider it a bug that it would inherit it all the way down and - not have any way to reset it to nothing - so that's why it is in - 1.2. (issue #8296 reported by gkloepfer) ........ - -2007-01-16 01:15 +0000 [r51057] Russell Bryant <russell@digium.com> - - * main/config.c: It is possible for the config pointer to be NULL - here, so it needs to be checked before dereferencing it. - -2007-01-16 00:22 +0000 [r51030] Matt O'Gorman <mogorman@digium.com> - - * apps/app_voicemail.c, configs/users.conf.sample: Patch allows for - changing voicemail password in users.conf from voicemail main, - written by AnthonyL bug #8436 - -2007-01-15 23:49 +0000 [r50994] Russell Bryant <russell@digium.com> - - * Makefile.rules: Filter out a few CFLAGS that are not valid - CXXFLAGS. - -2007-01-15 23:10 +0000 [r50988] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /: Blocked revisions 50987 via svnmerge ........ r50987 | - tilghman | 2007-01-15 17:09:02 -0600 (Mon, 15 Jan 2007) | 2 lines - Check return value before dereferencing (Bug 8822) ........ - -2007-01-15 21:08 +0000 [r50957] Matt O'Gorman <mogorman@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 50946 via svnmerge from - https://svn.digium.com/svn/asterisk/branches/1.2 ........ r50946 - | mogorman | 2007-01-15 14:44:53 -0600 (Mon, 15 Jan 2007) | 4 - lines Solves issue with forwarding voicemails from folders other - than inbox. patch by anthonyl. ........ - -2007-01-15 18:23 +0000 [r50921] Jason Parker <jparker@digium.com> - - * main/asterisk.c: re-add deprecated "show version" CLI command. - -2007-01-15 16:36 +0000 [r50895] Joshua Colp <jcolp@digium.com> - - * main/manager.c: Move event processing into do_message so that it - gets executed again when events are tripped. - -2007-01-15 15:03 +0000 [r50867] Kevin P. Fleming <kpfleming@digium.com> - - * configure, include/asterisk/autoconfig.h.in, main/Makefile, - configure.ac, Makefile.rules, acinclude.m4, makeopts.in: use the - ACX_PTHREAD macro from the Autoconf macro archive for setting up - compiler pthreads support... should improve portability to - platforms with unusual pthreads requirements - -2007-01-14 21:59 +0000 [r50820] Joshua Colp <jcolp@digium.com> - - * main/astmm.c: Add missing newlines for two memory CLI commands. - -2007-01-14 05:13 +0000 [r50782] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * main/db1-ast/db/db.c, main/db1-ast/recno/rec_get.c, - main/db1-ast/btree/bt_seq.c, main/db1-ast/hash/hash_func.c, - main/db1-ast/btree/bt_utils.c, main/db1-ast/recno/rec_seq.c, - main/db1-ast/btree/bt_overflow.c, main/db1-ast/btree/bt_search.c, - main/db1-ast/btree/bt_conv.c, main/db1-ast/btree/bt_close.c, - main/db1-ast/btree/bt_put.c, main/db1-ast/recno/rec_utils.c, - main/db1-ast/recno/rec_open.c, main/db1-ast/hash/hash_bigkey.c, - main/db1-ast/recno/rec_delete.c, main/db1-ast/hash/hash_buf.c, - main/db1-ast/hash/hash_page.c, main/db1-ast/recno/rec_close.c, - main/db1-ast/recno/rec_put.c, main/db1-ast/include/ndbm.h, - main/db1-ast/btree/bt_debug.c, main/db1-ast/mpool/mpool.c, - main/db1-ast/btree/bt_split.c, main/db1-ast/btree/bt_open.c, - main/db1-ast/btree/bt_delete.c, main/db1-ast/hash/hash_log2.c, - main/db1-ast/hash/hsearch.c, /, main/db1-ast/btree/bt_page.c, - main/db1-ast/recno/rec_search.c, main/db1-ast/btree/bt_get.c, - main/db1-ast/hash/hash.c: Merged revisions 50781 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r50781 | tilghman | 2007-01-13 23:01:16 -0600 (Sat, 13 - Jan 2007) | 2 lines Bug 8814 - db should look for its header - using a relative path, instead of the system path (Fixes FreeWRT) - ........ - -2007-01-13 16:45 +0000 [r50754] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile, build_tools/make_sample_voicemail (added): when - building the sample greetings for maibox 1234@default during - 'make samples', build a greeting for each language and file - format the user selected to install with menuselect (reported by - Brian Capouch on asterisk-dev) - -2007-01-13 06:00 +0000 [r50674-50727] Joshua Colp <jcolp@digium.com> - - * main/channel.c: Only write a frame out to the channel if one - exists. There are cases where one may not and would therefore - cause the channel driver to segfault. (issue #8434 reported by - slimey) - - * res/res_snmp.c: Only join the snmp thread on an unload if the - thread is actually running. (issue #8810 reported by junky) - -2007-01-12 19:24 +0000 [r50647] Jason Parker <jparker@digium.com> - - * configs/voicemail.conf.sample: Update documentation to state that - you shouldn't use realtime static with voicemail.conf - -2007-01-12 16:42 +0000 [r50602] Joshua Colp <jcolp@digium.com> - - * main/manager.c: We need to check for res being 0 in do_message - itself, otherwise our headers will get lost. - -2007-01-12 14:42 +0000 [r50562] Kevin P. Fleming <kpfleming@digium.com> - - * main/pbx.c, /: Merged revisions 50561 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r50561 | kpfleming | 2007-01-12 08:34:15 -0600 (Fri, 12 Jan 2007) - | 2 lines minor documentation clarification ........ - -2007-01-11 05:53 +0000 [r50377-50468] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Remove check for channel state as it can - definitely be something other then ring, and also clean up the - code a bit. This should solve the parking issues and maybe some - attended transfer issues people have been seeing. - - * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: Add - support to see whether NAT was detected (yay symmetric RTP) and - also add a check in chan_sip so that if NAT has been detected and - the reinvite behind nat option has been turned off, then just do - partial bridge. (issue #8655 reported by mnicholson) - - * apps/app_speech_utils.c: Merge speech-multi branch which adds - support for joining multiple sound files together to be played - one after another in SpeechBackground. - - * main/config.c: Fix parsing when using something like ldap - settings. (done by anthonyl) - - * channels/chan_sip.c: Fix chan_sip not working issue. Let's not - prematurely return 0. (issue #8783 reported by st41ker) - -2007-01-10 16:45 +0000 [r50346] Jason Parker <jparker@digium.com> - - * cdr/cdr_manager.c: Reverse some logic in cdr_manager, which made - it fail to load if the config file existed. Issue 8777 - -2007-01-10 04:55 +0000 [r50266-50298] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c, /: Merged revisions 50295 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r50295 | file | 2007-01-09 23:51:06 -0500 (Tue, 09 Jan 2007) | 2 - lines Add another return value to dial_exec_full that indicates - execution is going to continuing at a new - extension/context/priority and to just let it slide. (issue #8598 - reported by jon) ........ - - * main/pbx.c: Ensure data's existence before trying to access it. - (issue #8774 reported by rcourtna) - -2007-01-10 02:17 +0000 [r50228] Russell Bryant <russell@digium.com> - - * Makefile, /: Merged revisions 50227 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r50227 | russell | 2007-01-09 21:16:45 -0500 (Tue, 09 Jan 2007) | - 6 lines Make the number that represents the major version number - a single digit instead of 2. Using two digits makes it an octal - number when put into version.h, which breaks the compilation of - any out of tree module that checks the version for any version - after 1.2.7 (reported by Matteo Brancaleoni on the asterisk-dev - mailing list, who gave credit to vihai for pointing it out) - ........ - -2007-01-09 17:11 +0000 [r50186] Jason Parker <jparker@digium.com> - - * main/cli.c: Re-add CLI command that should have only been - deprecated in 1.4. Thanks kshumard! (reported in person, so no - associated issue #) - -2007-01-09 13:40 +0000 [r50151] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c, /: Merged revisions 50150 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r50150 | tilghman | 2007-01-09 07:30:04 -0600 (Tue, 09 Jan 2007) - | 4 lines The advent of realtime has enabled people to use commas - in the fullname field. This could cause an issue with sending - voicemails, when the field is unquoted. (Issue 8595) ........ - -2007-01-09 11:25 +0000 [r50124] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: - handle re-invites properly in sip_hangup() - - Add some invitestate status changes just to be sure - -2007-01-08 23:39 +0000 [r50098] Jason Parker <jparker@digium.com> - - * apps/app_voicemail.c: Fix an issue with voicemail and users.conf, - where it wouldn't ever parse a password, since it was using - "secret" instead of "password" Issue 8761, reported by and patch - suggestion from ssokol. - -2007-01-08 21:11 +0000 [r50073] Matt O'Gorman <mogorman@digium.com> - - * apps/app_senddtmf.c: we can't unlock a channel if we cant find - it. - AnthonyL bug #8741 - -2007-01-08 18:21 +0000 [r50032] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Disable the more intense packet2packet bridging until - the bugs can be worked out. - -2007-01-08 14:26 +0000 [r49925-50006] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #8677 - Handle failure of T.38 - re-invite This is not a fix, but adding an error message to tell - the admin that we have a bad configuration. We should not send - T.38 re-invites to devices that can't handle it (with the current - architecture where you have to hard-code t.38 support per - device). To really fix this, we need to figure out a way to tell - the incoming call that the re-invite failed, so we can signal - failure on that end and go back to the original call. - - * channels/chan_sip.c: Issue #8524, support multiple via header - values (tardieu) Thanks! - - * channels/chan_sip.c: We only need one forward declaration - - * channels/chan_sip.c: Issue 8735: Terminate state when extension - is unavailable for subscription - -2007-01-08 05:11 +0000 [r49890] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 49889 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r49889 | file | 2007-01-08 00:10:07 -0500 (Mon, 08 Jan 2007) | 2 - lines Ensure we use the default refresh value of 60 if the remote - server does not send one. (issue #8746 reported by maethor) - ........ - -2007-01-08 03:53 +0000 [r49866] Kevin P. Fleming <kpfleming@digium.com> - - * configure, configure.ac: since we use AC_PATH_TOOL to find tools, - we should use the results it provides for us (reported by Brian - Capouch on the asterisk-dev list) - -2007-01-07 21:44 +0000 [r49831-49834] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, apps/app_dictate.c: Merged revisions 49833 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r49833 | tilghman | 2007-01-07 15:43:10 -0600 (Sun, 07 Jan 2007) - | 2 lines If openstream fails, then we crash (Issue 8564) - ........ - - * channels/chan_sip.c: Second condition was a subset of the first, - so hold was never decremented, thus hint stayed stuck (Issue - 8747) - -2007-01-06 00:24 +0000 [r49742] Jason Parker <jparker@digium.com> - - * main/pbx.c, res/res_features.c, pbx/pbx_config.c: Save 1 whopping - byte of allocated memory! This looks like it may have been a - chicken/egg scenario.. You had to call a cleanup func, because - everything was allocated. Then since you had to call a cleanup - func, you were forced to allocate - ie; strdup(""). - -2007-01-05 23:51 +0000 [r49710-49715] Kevin P. Fleming <kpfleming@digium.com> - - * configure, acinclude.m4: one more time... - - * configure, acinclude.m4: proper fix for r49712 - - * configure, acinclude.m4: if --with-foo=<path> is specific for a - configure option, ensure that it is used for header file checking - as well - - * main/manager.c: ast_func_read() needs a writable copy of the - function name to be passed - -2007-01-05 23:16 +0000 [r49705] Jason Parker <jparker@digium.com> - - * channels/chan_zap.c, codecs/codec_zap.c: Make codec_zap and - chan_zap also depend on zaptel. This fixes an issue (8727) with - zaptel being in a different directory, using --with-zaptel. - -2007-01-05 22:52 +0000 [r49676-49680] Kevin P. Fleming <kpfleming@digium.com> - - * main/manager.c: don't 'consume' the params list before we try to - use it again - - * res/res_monitor.c, main/config.c, apps/app_setcdruserfield.c, - main/manager.c, include/asterisk/jabber.h, apps/app_senddtmf.c, - main/db.c, channels/chan_zap.c, channels/chan_sip.c, - apps/app_meetme.c, res/res_features.c, channels/chan_agent.c, - utils/astman.c, include/asterisk/manager.h, channels/chan_iax2.c, - apps/app_queue.c, res/res_jabber.c: reduce stack consumption for - AMI and AMI/HTTP requests by nearly 20K in most cases - -2007-01-05 22:14 +0000 [r49675] Joshua Colp <jcolp@digium.com> - - * main/channel.c: Don't keep repeating the warning over and over - when the end of the call is reached. (issue #8724 reported by - xrg) - -2007-01-05 17:09 +0000 [r49581-49636] Kevin P. Fleming <kpfleming@digium.com> - - * /, channels/chan_sip.c, channels/chan_skinny.c, - channels/chan_iax2.c: Merged revisions 49635 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r49635 | kpfleming | 2007-01-05 10:56:40 -0600 (Fri, 05 Jan 2007) - | 2 lines ensure that threads which are supposed to be detached - (because we aren't going to wait on them) are created properly - ........ - - * channels/chan_iax2.c: revert the dynamic_list insertion change... - that was not the right thing to do - - * channels/chan_iax2.c: create the IAX2 processing threads as - background threads so they will use smaller stacks when we create - a dynamic thread, put it on the dynamic_list right away so we - don't lose track of it - -2007-01-04 23:00 +0000 [r49568] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: It's possible for the iax2 pvt to - disappear, so if it has... don't bother looking for dpentries. - -2007-01-04 22:51 +0000 [r49553] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/threadstorage.h, main/asterisk.c, - build_tools/cflags.xml, include/asterisk.h, main/Makefile, - main/threadstorage.c (added), main/utils.c: add support for - tracking thread-local-storage objects that exist via - 'threadstorage' CLI commands - -2007-01-04 22:28 +0000 [r49551] Joshua Colp <jcolp@digium.com> - - * main/config.c: Only free comments and line buffer once we reach - the first level. (issue #8678 reported by ssokol, fixed by - anthonyl) - -2007-01-04 21:58 +0000 [r49460-49536] Kevin P. Fleming <kpfleming@digium.com> - - * channels/iax2-parser.c, main/frame.c: don't mark these - allocations as 'cache' allocations when caching has been disabled - - * channels/iax2-parser.c: if we're going to decrement the frame - count when we free a frame, we should inrement it when we create - one :-) - - * channels/iax2-parser.c, channels/iax2-parser.h, - channels/chan_iax2.c: only do IAX2 frame caching for voice and - video frames - - * main/frame.c: don't do frame header caching in the core if - LOW_MEMORY is defined - - * channels/iax2-parser.c: don't define this type either if - LOW_MEMORY is enabled - -2007-01-04 18:11 +0000 [r49459] Matt O'Gorman <mogorman@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 49447 via svnmerge from - https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49447 - | mogorman | 2007-01-04 11:45:16 -0600 (Thu, 04 Jan 2007) | 2 - lines converted a lot of 256 to PATH_MAX and some white space - fixes. ........ - -2007-01-04 18:06 +0000 [r49457-49458] Kevin P. Fleming <kpfleming@digium.com> - - * channels/iax2-parser.c: don't do frame caching in LOW_MEMORY mode - - * codecs/Makefile: make building of codec_gsm against the system - GSM library actually work - -2007-01-04 16:50 +0000 [r49413] Matt O'Gorman <mogorman@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 49412 via svnmerge from - https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49412 - | mogorman | 2007-01-04 10:48:43 -0600 (Thu, 04 Jan 2007) | 3 - lines good catch russell sorry i missed that. fix magic number - with proper sizeof ........ - -2007-01-04 04:33 +0000 [r49388] Russell Bryant <russell@digium.com> - - * funcs/func_realtime.c: Fix the REALTIME() dialplan function. - ast_build_string() advances the string pointer to the position to - begin the next write into the buffer. So, this pointer can not be - used to copy the contents of the string later. The beginning of - the buffer must be saved. Interestingly enough, this code could - not have ever worked. (Pointed out by Sebb on IRC, thanks!) - -2007-01-03 23:32 +0000 [r49355] Matt O'Gorman <mogorman@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 49354 via svnmerge from - https://svn.digium.com/svn/asterisk/branches/1.2 ........ r49354 - | mogorman | 2007-01-03 17:22:47 -0600 (Wed, 03 Jan 2007) | 6 - lines When using ODBC_STORAGE VoicemailMain doesn't create the - subdirectories for a mailbox such as the INBOX directory. this - patch solves that problem, was written by anthony be-125 ........ - -2007-01-03 09:06 +0000 [r49313] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/misdn_config.c, - doc/misdn.txt, channels/misdn/isdn_lib.h, channels/chan_misdn.c, - /, channels/misdn/ie.c, channels/misdn/isdn_msg_parser.c, - configs/misdn.conf.sample: Merged revisions - 48319,48321,48467,48552,48576,49135,49303 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) | - 1 line changed a few debugs to higher debug levels ........ - r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) | - 1 line added the export and import of the MISDN_ADDRESS_COMPLETE - Variable to inidcate wether the extension is already completely - dialed or if there might come additional digits by information - elements. also added some docs for that. ........ r48467 | - crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line - removed FIXUP state. added check for channel allocation conflict - when we create a setup while the other site creates a setup on - the same channel, besides the check we resolve this conflict. - ........ r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18 - Dez 2006) | 1 line when our PTP Partner sends us a SETUP with a - preselected channel we just accept it, even when we're NT. added - some checks for segfaults. ........ r48576 | crichter | - 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line when we - reject a channel, because it's in use already, we shouldn't - process the setup anymore. made the channel allocation a bit - easier and more understandable, removed a few unused lines - ........ r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02 - Jan 2007) | 1 line added check for channel ranges in the - set/empty channel functions. set pmp_l1_check default to no. - added misdn restart pid cli command. added cleaning of channel - when we send a RELEASE_COMPLETE. ........ r49303 | crichter | - 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines * Added - check for bridging in misdn_call to avoid setting - echocancellation when 2 mISDN channels are involved and when - bridging is set. That lead to a kernel panic before under - different situations, because we switched about 2 times between - hardware bridging and echocancelation * readded MISDN_URATE - variable which got lost before, this should make app_v110 work - again * fixed typo ........ - -2007-01-03 03:21 +0000 [r49282] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile, Makefile.rules: various Makefile improvements to get - chan_vpb (and any other C++ modules) to build properly - -2007-01-03 01:19 +0000 [r49259] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Check pvt structure presence before passing - to send_command. This gets rid of the irritating message about a - packet without pvt structure. This happens because the scheduled - item is getting cancelled at almost the exact moment it is - getting executed. - -2007-01-02 22:30 +0000 [r49237] Steve Murphy <murf@digium.com> - - * main/ast_expr2.fl, main/ast_expr2f.c, pbx/ael/ael_lex.c, - pbx/ael/ael.flex: This is a slight modification to Josh's edits - for #8579; both files edited were the produced by flex; so the - source files need to be changed instead, and the generated files - regenerated. - -2007-01-02 19:58 +0000 [r49212] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Small cleanup of add_t38sdp - it's always - enabled at that point in the code - -2007-01-02 17:33 +0000 [r49189] Jason Parker <jparker@digium.com> - - * main/pbx.c: Allow fractions of a second in the Wait() - application, like it says it allows. - -2007-01-02 13:59 +0000 [r49165] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c: remove comment that is unrelated to this - function - -2007-01-02 12:08 +0000 [r49145] Olle Johansson <oej@edvina.net> - - * configs/features.conf.sample: Adding note on effect of - applicationmap features on re-invites - -2007-01-01 23:34 +0000 [r49098-49102] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c, build_tools/menuselect-deps.in, configure, - configure.ac, codecs/codec_zap.c: check specifically for VLDTMF - and transcoding support in the system's Zaptel installation, and - make only the modules that need those features dependent on them - (this will allow building the other Zaptel-using parts of - Asterisk against older versions of Zaptel or those on other - platforms that haven't caught up yet to the Linux version) - - * Makefile: use a simpler (and portable) method to ensure that - menuselect is built as a host binary - - * Makefile: revert this change until a better solution can be - found... 'env -i' was not being used properly, but even when - changed to do so, this process fails during cross-compilation - because the menuselect build still sees 'CC' as set to the - cross-compiler - -2007-01-01 20:14 +0000 [r49096] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: remove incomplete implementation of dnsmgr. - Let's fix this in trunk. - -2006-12-30 18:31 +0000 [r49063-49073] Joshua Colp <jcolp@digium.com> - - * pbx/pbx_config.c: IAX has been deprecated for quite some time so - we had better use IAX2 when creating the dial string for users. - (issue #8697 reported by ssokol) - - * channels/chan_zap.c: Use asprintf to build the channel names - instead of custom function. I believe the custom function is - doing some things that are not portable across all - implementations. (issue #8570 reported by hterag & issue #8692 - reported by nicolasg) - - * main/rtp.c: If the Packet2Packet bridge is being broken because - of a masquerade then attempt to read a frame in so the masquerade - actually happens. Otherwise weirdness will occur. (issue #8696 - reported by kjotte) - - * channels/chan_iax2.c: Initialize the packet queue in load_module - instead of just declaring the list with the default value. (issue - #8695 reported by ssokol) - -2006-12-30 00:40 +0000 [r49061] Steve Murphy <murf@digium.com> - - * pbx/pbx_ael.c: A fix for 8661, where the CUT func needed to have - comma args converted to vertical bars. I hope this change does - little harm. - -2006-12-29 00:50 +0000 [r49042-49048] Kevin P. Fleming <kpfleming@digium.com> - - * /: put this value into the correct property - - * /, BUGS: Merged revisions 49045 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r49045 | kpfleming | 2006-12-28 18:32:32 -0600 (Thu, 28 Dec 2006) - | 2 lines location of the bug posting guidelines has changed - ........ - - * sample.call: simple commit to test CIA integration - -2006-12-28 21:26 +0000 [r49032-49035] Jason Parker <jparker@digium.com> - - * main/cli.c: Fix some deprecated commands. Issue 8682, patch by me - - * main/http.c: saw this in passing... fix a small typo - -2006-12-28 20:08 +0000 [r49028] Kevin P. Fleming <kpfleming@digium.com> - - * sounds/Makefile: new versions of sounds - -2006-12-28 19:52 +0000 [r49024] Jason Parker <jparker@digium.com> - - * main/http.c: make the uris_lock a rwlock instead of a mutex lock - - needs to be forward ported to trunk - -2006-12-28 19:43 +0000 [r49022] Joshua Colp <jcolp@digium.com> - - * configure, include/asterisk/autoconfig.h.in, configure.ac, - include/asterisk/lock.h: Backport support for read/write locks. - -2006-12-28 19:21 +0000 [r49020] Steve Murphy <murf@digium.com> - - * main/ast_expr2.fl, main/ast_expr2.c, main/frame.c, - pbx/ael/ael.tab.c, main/ast_expr2.y, main/ast_expr2f.c, - pbx/ael/ael_lex.c, include/asterisk/ael_structs.h, - pbx/ael/ael.tab.h, utils/ael_main.c: removed <err.h> as in trunk - from the ael stuff. Also, threw in a minor fix to frame.c to - avoid build-killing compiler warnings. - -2006-12-27 22:28 +0000 [r49009] Joshua Colp <jcolp@digium.com> - - * main/ast_expr2f.c, pbx/ael/ael_lex.c: ast_copy_string is not - available when LOW_MEMORY is used and things are being built in - the utils directory, so we need to resort to the old method of - strncpy. (issue #8579 reported by mottano) - -2006-12-27 22:06 +0000 [r48998-49006] Kevin P. Fleming <kpfleming@digium.com> - - * main/enum.c, main/asterisk.c, main/rtp.c, main/term.c, - main/cdr.c, main/channel.c, main/udptl.c, main/pbx.c, - main/dnsmgr.c, main/frame.c, main/manager.c, main/file.c, - main/http.c, main/logger.c: since these variables all have static - duration, none of them need initializers (they default to zero - anyway) - - * include/asterisk/options.h, main/asterisk.c, main/file.c: move - extern declaration for this option to a header file where it - belongs provide an initial value for 'languageprefix' option, - instead of relying on randomness to provide a useful value - -2006-12-27 21:06 +0000 [r48993-48997] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Only include acl.h and lock.h once - - * channels/chan_sip.c: Only set rfc2833compensate flag once - (handle_common_options) - - * channels/chan_sip.c: - Remove checking for T38 options twice. - Keeping them in handle_common_options - -2006-12-27 18:33 +0000 [r48987-48988] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: make the option actually match the - documentation - - * channels/iax2-parser.c, include/asterisk/utils.h, - include/asterisk/astmm.h, main/frame.c, main/astmm.c: allow 'show - memory' and 'show memory summary' to distinguish memory - allocations that were done for caching purposes, so they don't - look like memory leaks - -2006-12-27 17:59 +0000 [r48975-48985] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c, configs/sip.conf.sample: Be a bit more - politically correct - - * channels/chan_sip.c, configs/sip.conf.sample: Issue #8575 - Buggy - cisco MWI support. Normally we try not to change our software for - bugs in other devices. But in this case, the Cisco phones are so - widespread so we try to implement a fix while waiting for a - bugfix from Cisco. - - * channels/chan_sip.c: - Make sure handle_common_options return 1 - when we found a common option - Move uncommon (only global) - option away from handle_common_options Reported by rizzo. Thanks! - - * channels/chan_sip.c: Issue 8599 (rizzo) Change invitestate before - re-sending invite with auth. - - * /, channels/chan_sip.c: Fix bogus content-length in t38 sdp. - (rizzo, #8600) - -2006-12-26 05:20 +0000 [r48960-48966] Joshua Colp <jcolp@digium.com> - - * apps/app_meetme.c: Get rid of a needless memory allocation and - only create a conference structure in find_conf_realtime if data - was read from realtime. (issue #8669 reported by robl) - - * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: Add an - API call that initializes an RTP structure. We need this because - chan_sip is cheeky and uses a temporary RTP structure for codec - purposes, and the API calls that are used rely on the lock. - (Pointed out on asterisk-dev by Andy Wang) - - * configure, configure.ac: Clean up autoconf file (gets rid of - warnings seen when rebuilding configure) and rebuild configure. - -2006-12-25 05:21 +0000 [r48931-48956] Russell Bryant <russell@digium.com> - - * /, funcs/func_math.c: Merged revisions 48955 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48955 | russell | 2006-12-25 00:19:48 -0500 (Mon, 25 Dec 2006) | - 6 lines Fix an error introduced by copying and pasting the - handling of the >= operator for the MATH function. If a single - equal sign was used as an operator, the function would treat it - is as if it were the >= operator. Now, it properly handles it as - an invalid operator. (issue #8665, patch by tempest1) ........ - - * channels/chan_oss.c: Fix a typo in an error message that - indicated that the MGCP channel type could not be registered, - instead of the correct type, OSS. - - * /, channels/chan_iax2.c: Merged revisions 48943 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48943 | russell | 2006-12-24 02:23:07 -0500 (Sun, 24 Dec 2006) | - 3 lines Check for the proper return value on an error in a call - to mmap(). This was reported by Andy Wang on the asterisk-dev - list. Thanks! ........ - - * /, channels/chan_sip.c: Merged revisions 48939 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48939 | russell | 2006-12-24 01:47:29 -0500 (Sun, 24 Dec 2006) | - 3 lines Remove a couple of misplaced dots in log messages. This - was reported by Andrea Spadaccini on the asterisk-dev mailing - list. ........ - - * main/http.c: Implement locking for the list of URI handlers to - make it thread-safe. - -2006-12-23 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.4.0 released. - -2006-12-22 22:33 +0000 [r48870-48906] Jason Parker <jparker@digium.com> - - * Makefile, main/stdtime/localtime.c: Minor fixes for Solaris. - - * channels/chan_skinny.c: Fix for issue 7774 - patch by alamantia - -2006-12-21 20:26 +0000 [r48783] Joshua Colp <jcolp@digium.com> - - * /, redhat/asterisk.spec: Merged revisions 48782 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48782 | file | 2006-12-21 15:25:01 -0500 (Thu, 21 Dec 2006) | 2 - lines Add new silence sound files to the spec for Redhat. (issue - #8652 reported by alvaro_palma_aste) ........ - -2006-12-20 02:56 +0000 [r48592-48637] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: vms doesn't exist on non-IMAP storage - builds. - - * apps/app_voicemail.c: Pass 'vms' pointer to record_and_review so - it is then passed to the IMAP store file function. (issue #8614 - reported by punknow) - - * doc/snmp.txt: find is not the same as bind when it comes to - documentation. (issue #8626 reported by johann8384) - -2006-12-19 21:28 +0000 [r48586] Kevin P. Fleming <kpfleming@digium.com> - - * channels/Makefile: suppress compiler warnings in this module - until it can be improved - -2006-12-19 21:12 +0000 [r48585] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c, /: Merged revisions 48584 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48584 | file | 2006-12-19 16:10:26 -0500 (Tue, 19 Dec 2006) | 2 - lines Free localuser structure when we fail to dial (issue #8612 - reported by rizzo) ........ - -2006-12-19 21:03 +0000 [r48583] Luigi Rizzo <rizzo@icir.org> - - * apps/app_sms.c: fix a bogus datalen in the frames generated by - app_sms (causing noisy output if you listen to the output!) This - affects trunk as well, whereas 1.2 is ok. - -2006-12-19 14:57 +0000 [r48577] Kevin P. Fleming <kpfleming@digium.com> - - * res/res_config_odbc.c, funcs/func_odbc.c: use the proper variable - type for these unixODBC API calls, eliminating warnings on 64-bit - platforms that use the 'new' 64-bit types for ODBC API calls - -2006-12-19 03:46 +0000 [r48571] Joshua Colp <jcolp@digium.com> - - * Makefile: Use env -i to start a fresh environment when going to - build menuselect. This is more portable then using unset. (issue - #8543 reported by jtodd) - -2006-12-18 17:23 +0000 [r48566] Luigi Rizzo <rizzo@icir.org> - - * include/asterisk/channel.h: unbreak the macro used for - incrementing the frame counters. I don't know when the bug was - introduced, but with the typical usage c->fin = - FRAMECOUNT_INC(c->fin) the frame counters stay to 0. affects - trunk as well (fix coming). - -2006-12-18 17:15 +0000 [r48564] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Put thread into proper list if we abort - handling due to an error, and also hold the lock while putting it - back into the proper idle list so we don't prematurely get a - signal. (issue #8604 reported by arkadia) - -2006-12-18 11:59 +0000 [r48513-48554] Kevin P. Fleming <kpfleming@digium.com> - - * codecs/lpc10/Makefile, main/Makefile, codecs/gsm/Makefile, - utils/astman.c, utils/smsq.c, codecs/ilbc/Makefile, - utils/ael_main.c: remove some now-unnecessary explicit includes - of autoconfig.h clean up per-file dependencies during 'make - clean' - - * build_tools/prep_tarball: need an additional argument here to - make the downloads actually occur - - * configure, include/asterisk/autoconfig.h.in, configure.ac, - acinclude.m4: use m4 quoting for AC_MSG_NOTICE calls, to keep - these calls from thinking they have multiple arguments - - * codecs/ilbc, formats, utils/Makefile, agi/Makefile, Makefile, - funcs, build_tools/mkdep (removed), codecs/lpc10, main/db1-ast, - main, codecs/gsm, pbx, res, channels, codecs, utils, agi, - main/Makefile, apps, Makefile.moddir_rules, Makefile.rules, cdr: - simplify dependency tracking system, using the compiler's - built-in method for generating them, and only doing dependency - tracking if developer mode is enabled via the configure script - - * Makefile, include/asterisk.h, main/stdtime/localtime.c: since we - really, really have to have autoconfig.h included before all - other headers (especially system headers), the Makefile will now - force it to happen (this will fix build problems with files like - ast_expr2f.c, where we can't control the inclusion order in the - file itself) - - * funcs/func_curl.c: instead of initializing the curl library every - time the CURL() function is invoked, do it only once per thread - (this allows multiple calls to CURL() in the dialplan for a - channel to run much more quickly, and also to re-use connections - to the server) (thanks to JerJer for frequently complaining about - this performance problem) - -2006-12-15 19:55 +0000 [r48502-48506] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Turn payload_lock into bridge_lock and make it - encompass all RTP structure contents that may relate to bridge - information, including who we are bridged to. - - * channels/chan_iax2.c: Hold call structure lock in places where a - qualify or peer action can destroy it. - - * channels/chan_iax2.c: Lock network retransmission queue in all - places that it is used. - -2006-12-15 10:55 +0000 [r48481-48487] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Issue #8592 - treat 504 as 503 (imported - from 1.2) - - * channels/chan_sip.c: Update to latest IANA spec - -2006-12-15 06:28 +0000 [r48461-48478] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Use a wakeup variable so that we don't wait - on IO indefinitely if packets need to be retransmitted. - - * main/rtp.c, include/asterisk/rtp.h: Payload values on the RTP - structure can change AFTER a bridge has started. This comes from - the packet handling of the SIP response when indication that it - was answered has been sent. Therefore we need to protect this - data with a lock when we read/write. (issue #8232 reported by - tgrman) - - * main/rtp.c: Remove direct RTCP bridging. I've come to the - conclusion that we should handle this through the core and not - just forward it on. Should solve a few bugs. - -2006-12-12 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.4.0-beta4 released. - -2006-12-12 04:13 +0000 [r48401] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Use S_OR in my previous app_voicemail. This - is the way it should have been done. - -2006-12-11 23:02 +0000 [r48396-48399] Matt O'Gorman <mogorman@digium.com> - - * sounds/Makefile: new sounds package with 100% more silence - - * /, apps/app_externalivr.c: Merged revisions 48394 via svnmerge - from https://svn.digium.com/svn/asterisk/branches/1.2 ........ - r48394 | mogorman | 2006-12-11 15:55:43 -0600 (Mon, 11 Dec 2006) - | 4 lines app_externalivr needs a real silence file, and - additional changes to add silence files into core instead of - extra patch provided by bug 8177 with minor additions. ........ - -2006-12-11 21:31 +0000 [r48391] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Return non-existant callerid handling to - that which it was before. In 1.4 and trunk callerid can be - allocated but not have any contents so we have to use - ast_strlen_zero before passing it to the relevant functions. - (issue #8567 reported by pabelanger) - -2006-12-11 05:37 +0000 [r48382] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * funcs/func_strings.c: STRFTIME() does not actually require an - argument (issue 8540) - -2006-12-11 05:36 +0000 [r48377-48381] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Merge in my latest RTP changes. Break out RTP and - RTCP callback functions so they no longer share a common one. - - * apps/app_meetme.c: Use the correct API call to say a device state - changed. (Yes, I'm a nub.) - - * apps/app_meetme.c: Don't access the conference structure after it - has been freed. - -2006-12-11 00:47 +0000 [r48375] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_nbscat.c, /, apps/app_festival.c, apps/app_mp3.c, - res/res_agi.c, apps/app_zapras.c, apps/app_externalivr.c, - apps/app_ices.c, res/res_musiconhold.c: Merged revisions 48374 - via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48374 | tilghman | 2006-12-10 18:33:59 -0600 (Sun, 10 Dec 2006) - | 5 lines When doing a fork() and exec(), two problems existed - (Issue 8086): 1) Ignored signals stayed ignored after the exec(). - 2) Signals could possibly fire between the fork() and exec(), - causing Asterisk signal handlers within the child to execute, - which caused nasty race conditions. ........ - -2006-12-10 03:04 +0000 [r48372] Steve Murphy <murf@digium.com> - - * channels/chan_zap.c, /: Merged revisions 48371 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48371 | murf | 2006-12-09 19:14:13 -0700 (Sat, 09 Dec 2006) | 1 - line This version applies the patch suggested by stevens in bug - 7836 (make inbound channel RINGING state consistent with other - channels). ........ - -2006-12-09 15:59 +0000 [r48362-48363] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Use locking when accessing the - registrations list. This list is not actually used very often, so - the likelihood of there being a problem is pretty small, but - still possible. For example, if the CLI command to list the - registrations was called at the same time that a reload was - occurring and the registrations list was getting destroyed and - rebuilt, a crash could occur. In passing, go ahead and convert - this list to use the linked list macros. - - * /: Blocked revisions 48361 via svnmerge ........ r48361 | russell - | 2006-12-09 10:45:37 -0500 (Sat, 09 Dec 2006) | 6 lines Use - locking when accessing the registrations list. This list is not - actually used very often, so the likelihood of there being a - problem is pretty small, but still possible. For example, if the - CLI command to list the registrations was called at the same time - that a reload was occurring and the registrations list was - getting destroyed and rebuilt, a crash could occur. ........ - -2006-12-07 18:17 +0000 [r48357] Russell Bryant <russell@digium.com> - - * /, res/res_musiconhold.c: Merged revisions 48356 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r48356 | russell | 2006-12-07 13:14:13 -0500 (Thu, 07 - Dec 2006) | 3 lines Ensure that the file position is not - incremented beyond the total number of files available for - playback. (issue #8539, ulogic) ........ - -2006-12-07 15:33 +0000 [r48349] Steve Murphy <murf@digium.com> - - * main/manager.c, UPGRADE.txt, CHANGES: Here lies the fixes that - killed bug 8423 -- OriginateSuccess and OriginateError incomplete - channel name. May it rest in peace. - -2006-12-06 16:25 +0000 [r48326] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Issue #8258 - fix handling of 487 being - retransmitted to Asterisk - -2006-12-06 16:15 +0000 [r48323] Russell Bryant <russell@digium.com> - - * configs/iax.conf.sample, /: Merged revisions 48322 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r48322 | russell | 2006-12-06 11:05:54 -0500 (Wed, 06 - Dec 2006) | 3 lines Fix the name of the rtignoreregexpire option - in the sample configuration file. (issue #8526, arkadia) ........ - -2006-12-06 12:27 +0000 [r48316-48317] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Don't send Contact on MESSAGE - -2006-12-05 20:42 +0000 [r48279] Jason Parker <jparker@digium.com> - - * configure.ac: Fix curl version number testing to be much more - friendly to non-bash shells. Issue 8508, patch by me. This - *SHOULD* be POSIX compliant now.. - -2006-12-05 17:29 +0000 [r48264-48270] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Merging the invitestate-1.4 branch after - successful testing. Will check if I can solve this with less - changes in 1.2. - - * configs/sip.conf.sample: Add missing s from another repository. - (thanks jcmoore!) - - * configs/sip.conf.sample: Updating sip.conf.sample with - information about T38 not working when chan_local or chan_agent - is involved in the call. I don't know how big a fix that would be - to solve, but this is the current state of affairs. (Chan_sip - currently checks if the other side of the bridge has a SIP tech. - We could/should implement another check, possibly for udptl_write - or some flag in the ast_channel structure). - -2006-12-05 01:41 +0000 [r48252-48254] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Oops, forgot to release the odbc handle - - * apps/app_voicemail.c, /: Merged revisions 48251 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48251 | tilghman | 2006-12-04 19:26:08 -0600 (Mon, 04 Dec 2006) - | 6 lines If the recording in the database is too large, it will - fail to retrieve with an mmap error. Not too sure why this - doesn't happen when we put it in the database, also, but since - that doesn't seem to be broken, I'm not going to fix it (at least - until someone reports it). Solution is to ask for the file in - smaller chunks. (Bug 8385) ........ - -2006-12-04 21:48 +0000 [r48237-48248] Jason Parker <jparker@digium.com> - - * apps/app_voicemail.c: Fix an issue which didn't allow - unavail/greet/busy/etc messages from being saved into ODBC (and - probably IMAP). - - * /: Blocked revisions 48246 via svnmerge ........ r48246 | qwell | - 2006-12-04 15:20:34 -0600 (Mon, 04 Dec 2006) | 7 lines Revert - change from 8016 - this breaks other stuff... Needs further - review. Tip: When you've reported a bug about something and - somebody has put up a patch for it.. It's not a good idea to open - a completely new bug and say that something is broken because of - the patch in the other bug - PLEASE mention something in the bug - where the patch was actually created. ........ - - * /: Blocked revisions 48236 via svnmerge ........ r48236 | qwell | - 2006-12-04 13:06:26 -0600 (Mon, 04 Dec 2006) | 4 lines Fix an - issue where a message isn't saved correctly when using ODBC - storage and reviewing a message. Issue 8016 - patch by sokhapkin. - ........ - -2006-12-04 18:16 +0000 [r48234] Joshua Colp <jcolp@digium.com> - - * /: Blocked revisions 48233 via svnmerge ........ r48233 | file | - 2006-12-04 13:14:46 -0500 (Mon, 04 Dec 2006) | 2 lines If the - generic bridge tells us not to retry, and we have a frame to spit - out then break the bridge. Props to markit in #asterisk-bugs for - bringing this up. ........ - -2006-12-04 17:54 +0000 [r48228-48230] Jason Parker <jparker@digium.com> - - * configs/voicemail.conf.sample: Add documentation to - voicemail.conf.sample for ODBC storage. Issue 8499 - patch by - blitzrage. - - * doc/snmp.txt: Attempt to document some of the dependencies that - are needed for net-snmp Issue 8499 - initial patch by blitzrage. - -2006-12-03 06:34 +0000 [r48223] Russell Bryant <russell@digium.com> - - * sounds/Makefile: When "fetch" is in use, instead of "wget", - --continue is not a valid option. (issue #8451) - -2006-12-02 21:45 +0000 [r48199-48219] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: - Removing one of two pieces of code to - handle 481 response on INVITE - Move handling of REFER response - to handle_response_refer() - - * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h, - configs/sip.conf.sample: - Disable RTP hold timers while T.38 fax - transmission happens - Encapsulate RTP timers in the rtp - structure so we have one for video and one for audio The video - one is not used in 1.4, really. Will be used for RTP keepalives - when we can send something that video phones support in the RTP - stream. I now this is a big architectual change at this stage for - 1.4, but decided it was needed to avoid future bug reports. - - Document the RTP NAT keepalive option in sip.conf.sample Issue - 7679 in the bug tracker. Please test. - -2006-12-02 03:50 +0000 [r48195] Russell Bryant <russell@digium.com> - - * include/asterisk/utils.h: Backport the comment containing the - warning regarding the limitations on the usage of this function. - It is thread safe, but not technically reentrant. - -2006-12-01 23:37 +0000 [r48193] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_dial.c, /: Merged revisions 48192 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48192 | kpfleming | 2006-12-01 17:30:59 -0600 (Fri, 01 Dec 2006) - | 2 lines if Dial() is going to send music-on-hold to the calling - party, it has to send PROGRESS first to ensure that the reverse - audio path has been setup first (BE-106) ........ - -2006-12-01 23:16 +0000 [r48190] Russell Bryant <russell@digium.com> - - * Makefile, configure, configure.ac, makeopts.in, sounds/Makefile: - FreeBSD 6.1 does not include wget by default. However, it has - fetch which will work just fine for our purposes of downloading - the sounds packages. So, check for both wget and fetch and the - configure script and use what was found to download them. If - neither one was found, and sound packages are selected that must - be downloaded, the install process will print out an informative - error message indicating the situation. Also, fix a couple places - where "make" was hard coded into some output messages by - replacing them with the $(MAKE) variable. (issue #8451, initial - patch by pabelanger, with additional modifications by me) - -2006-12-01 20:25 +0000 [r48184-48186] Jason Parker <jparker@digium.com> - - * configs/extensions.conf.sample, /: Merged revisions 48183 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48183 | qwell | 2006-12-01 14:19:10 -0600 (Fri, 01 Dec 2006) | 2 - lines Fix a small typo - issue 8848, reported by pabelanger - ........ - -2006-12-01 19:38 +0000 [r48179] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * main/cli.c: Double-unlock error (reported by blitzrage on IRC) - -2006-12-01 17:41 +0000 [r48177] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c, configs/sip.conf.sample: - Backport of the - "limitonpeers" patch from trunk, to fix a lot of issues with - queues and SIP device states - Remove support for T.38 early - media, since it's impossible. (Two patches in one - extra friday - evening offer due to being off line from svn today... :-) - -2006-11-30 21:18 +0000 [r48168] Joshua Colp <jcolp@digium.com> - - * main/rtp.c, include/asterisk/rtp.h, channels/chan_gtalk.c: Do not - do a partial bridge for Google Talk since we need to handle STUN. - (issue #8448 reported by phsultan) - -2006-11-30 20:51 +0000 [r48166] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Issue 8319 - change noncecount before - using it. - -2006-11-30 20:28 +0000 [r48143-48162] Joshua Colp <jcolp@digium.com> - - * /: Blocked revisions 48161 via svnmerge ........ r48161 | file | - 2006-11-30 15:27:29 -0500 (Thu, 30 Nov 2006) | 2 lines Don't - write AST_FRAME_NULL or AST_FRAME_IAX frames out to the channel - driver. (issue #8390 reported by hselasky) ........ - - * /, channels/chan_iax2.c: Merged revisions 48157 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48157 | file | 2006-11-30 15:06:43 -0500 (Thu, 30 Nov 2006) | 2 - lines Only print out debug message if bridged channel is not - NULL. (issue #8412 reported by jubilex) ........ - - * /, res/res_features.c: Merged revisions 48154 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48154 | file | 2006-11-30 14:04:11 -0500 (Thu, 30 Nov 2006) | 2 - lines Do not listen for DTMF on the bridge that comes into - existence when ParkedCall is executed. This means native bridging - can now occur for this. (issue #8406 reported by kebl0155) - ........ - - * main/cdr.c, /: Merged revisions 48151 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48151 | file | 2006-11-30 13:42:45 -0500 (Thu, 30 Nov 2006) | 2 - lines Print certain CDR messages out at the NOTICE level versus - WARNING since they can occur when used with the CDR applications - and are perfectly fine. (issue #8367 reported by dartvader) - ........ - - * /: Blocked revisions 48146 via svnmerge ........ r48146 | file | - 2006-11-30 13:17:54 -0500 (Thu, 30 Nov 2006) | 2 lines Remember - the pointer to the allocated block of memory so that we can free - it and not cause a memory leak. (issue #8449 reported by arkadia) - ........ - - * /, configs/sip.conf.sample: Merged revisions 48142 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r48142 | file | 2006-11-30 12:55:23 -0500 (Thu, 30 Nov - 2006) | 2 lines Document 'port' for SIP peers, came up because of - the current mailing list thread. (issue #8450 reported by - blitzrage) ........ - -2006-11-30 14:29 +0000 [r48129-48135] Olle Johansson <oej@edvina.net> - - * doc/manager.txt: Explain status reports and make codefreeze more - happy :-) - - * /, channels/chan_sip.c: Clean up bad dialogs properly. Caused by - GS 487 adapter without CSEQ on separate line in the REGISTER - request. Imported from 1.2. - -2006-11-29 21:05 +0000 [r48115] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Use MAILTMPLEN instead of sizeof in - mm_login. (issue #8420 reported by slimey) - -2006-11-29 19:56 +0000 [r48113] Olle Johansson <oej@edvina.net> - - * configs/sip.conf.sample: Explain the use device status system - implemented in SIP for subscriptions, queues and manager a bit - better. Like in 1.2, you will get more detailed information if - you set a call limit for a device. When the call limit is - reached, the status system will report a device as busy. For - queues, setting a call limit per SIP device is propably a - requirement. In most cases, it will work much better if you only - use type=peer and not type=friend. We might decide to backport - the new setting from trunk to apply all call limits to the peer - part of a friend only. - -2006-11-29 16:50 +0000 [r48107] Joshua Colp <jcolp@digium.com> - - * main/rtp.c, /: Merged revisions 48106 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48106 | file | 2006-11-29 11:47:10 -0500 (Wed, 29 Nov 2006) | 2 - lines If the frame was duplicated before writing out then we need - to free it. (issue #8429 reported by edguy3) ........ - -2006-11-29 08:03 +0000 [r48105] Olle Johansson <oej@edvina.net> - - * configs/sip.conf.sample: Clarify RTP timers. Sorry, grandma. - -2006-11-29 04:26 +0000 [r48101] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Don't crash if the mailstream was not - created. - -2006-11-28 18:26 +0000 [r48095] Jason Parker <jparker@digium.com> - - * Makefile: Export several more variables in top level Makefile. - Inspired by issue 8438. - -2006-11-28 16:57 +0000 [r48054-48088] Joshua Colp <jcolp@digium.com> - - * channels/chan_phone.c, /: Merged revisions 48087 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r48087 | file | 2006-11-28 11:56:01 -0500 (Tue, 28 Nov - 2006) | 2 lines According to the research I have done we never - needed to include compiler.h in the first place so let's not! - (issue #8430 reported by edguy3) ........ - - * apps/app_voicemail.c, /: Merged revisions 48053 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48053 | file | 2006-11-27 13:03:57 -0500 (Mon, 27 Nov 2006) | 2 - lines Use the proper function to get the new message count - instead of always using the filesystem. (issue #8421 reported by - slimey) ........ - -2006-11-27 17:20 +0000 [r48049] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, res/res_musiconhold.c: Merged revisions 48045 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r48045 | tilghman | 2006-11-27 11:15:54 -0600 (Mon, 27 - Nov 2006) | 2 lines Random MOH wasn't really random (bug 8381) - ........ - -2006-11-27 17:17 +0000 [r48046] Russell Bryant <russell@digium.com> - - * main/manager.c: Remove a couple of unused variables (issue #8380, - casper) - -2006-11-27 15:32 +0000 [r48038] Joshua Colp <jcolp@digium.com> - - * pbx/pbx_spool.c, /: Merged revisions 48037 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r48037 | file | 2006-11-27 10:30:37 -0500 (Mon, 27 Nov 2006) | 2 - lines Do not reference the freed outgoing structure in the debug - message. (issue #8425 reported by arkadia) ........ - -2006-11-27 06:41 +0000 [r48031] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Change logging message - -2006-11-26 00:26 +0000 [r48015-48017] Steve Murphy <murf@digium.com> - - * funcs/func_cdr.c: might as well also document the raw values of - the flag vars - - * /, funcs/func_cdr.c: A little bit of func_cdr documentation - upgrade-- no bug# involved, although 8221 may have inspired it. - -2006-11-25 09:28 +0000 [r48002] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Not having a HINT is not an ERROR. In 1.4 - and future releases, you can disable subscription support totally - or per peer in sip.conf with allowsubscribe = yes | no - -2006-11-24 17:17 +0000 [r47992] Steve Murphy <murf@digium.com> - - * main/translate.c: bug 8189 posted this fix for main/translate.c - for PLC - -2006-11-24 15:46 +0000 [r47989] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/misdn_config.c, - channels/chan_misdn.c, /: Merged revisions 47968 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r47968 | crichter | 2006-11-23 17:10:23 +0100 (Do, 23 - Nov 2006) | 1 line fixed a litle bug regarding HOLD/RETRIEVE. - beatufied some logs, changed some loglevels. changed the default - value of block_on_alarm ........ - -2006-11-23 11:01 +0000 [r47959] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Don't allocate unused variable. - -2006-11-22 21:47 +0000 [r47944] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Video will never reach Packet2Packet bridging and can - do more harm then good. - -2006-11-21 17:32 +0000 [r47897] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: If we have the non standard G726-32 setting turned on - we want to return G726-32 to the SDP, not our AAL2 string. (issue - #8330 reported by voipgate) - -2006-11-21 15:20 +0000 [r47892] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Apparently Exosip sends a 101 after a 100 - provisional response. Let's not treat that as early media. - (discovered at the AVTF meeting in Paris). - -2006-11-20 20:01 +0000 [r47863-47864] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Oops, merge missed release of odbc object - - * apps/app_voicemail.c, /: Merged revisions 47862 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r47862 | tilghman | 2006-11-20 13:59:07 -0600 (Mon, 20 Nov 2006) - | 2 lines Failing to trap -1 error from mmap causes segfault - (Issue 8385) ........ - -2006-11-20 19:51 +0000 [r47850-47860] Joshua Colp <jcolp@digium.com> - - * main/frame.c, /: Merged revisions 47859 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r47859 | file | 2006-11-20 14:50:21 -0500 (Mon, 20 Nov 2006) | 2 - lines Don't forget to byte swap if we are exiting the smoother - feed early. (issue #8287 reported by arturs) ........ - - * /: Blocked revisions 47855 via svnmerge ........ r47855 | file | - 2006-11-20 11:16:22 -0500 (Mon, 20 Nov 2006) | 2 lines Free - history items at the end of use of the temporary SIP pvt - structure. (issue #8383 reported by benh) ........ - - * main/rtp.c: Only remove/destroy the RTCP I/O item if it exists. - - * .cleancount, apps/app_dial.c, apps/app_directed_pickup.c, - include/asterisk/channel.h: Use a separate variable in the - channel structure to store the context that the channel was - dialed from. (issue #8382 reported by jiddings) - -2006-11-20 11:45 +0000 [r47843-47845] Olle Johansson <oej@edvina.net> - - * configs/sip.conf.sample: Explain properly how videosupport works. - Committ from Asterisk Video Task Force meeting in Paris! - - * /, channels/chan_sip.c: Make sure we destroy scheduled items and - not use them ever again after destruction (rizzo) - -2006-11-18 17:59 +0000 [r47823] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: fix bug 7450 - Parsing fails if From header - contains angle brackets (the bug was only in a corner case where - the < was right after the opening quote, and the fix is trivial). - -2006-11-16 23:19 +0000 [r47781-47782] Jason Parker <jparker@digium.com> - - * apps/app_db.c, apps/app_dial.c: Fix a couple of typos. Initially - pointed out by mrobinson. - - * /: Blocked revisions 47780 via svnmerge ........ r47780 | qwell | - 2006-11-16 17:16:35 -0600 (Thu, 16 Nov 2006) | 2 lines Fix a - couple of typos in applications.. Initially spotted by mrobinson. - ........ - -2006-11-16 23:00 +0000 [r47777] Kevin P. Fleming <kpfleming@digium.com> - - * /, doc/billing.txt: update documentation regarding IAX2 transfers - and CDRs Merged revisions 47776 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r47776 | kpfleming | 2006-11-16 16:57:31 -0600 (Thu, 16 Nov 2006) - | 2 lines update clearly wrong documentation regarding cdr_custom - ........ - -2006-11-16 21:11 +0000 [r47762-47764] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Compare technology using the pointers - instead of a straight comparison based on name. (issue #8228 - reported by dean bath) - - * /: Blocked revisions 47761 via svnmerge ........ r47761 | file | - 2006-11-16 15:29:28 -0500 (Thu, 16 Nov 2006) | 2 lines Look for - the header file specifically in all cases, not just the existence - of the directory. (issue #8358 reported by mrness) ........ - -2006-11-16 20:09 +0000 [r47758] Kevin P. Fleming <kpfleming@digium.com> - - * configure, configure.ac: check for pre-1.4 versions of Zaptel and - abort the configure script if found with an appropriate error - message - -2006-11-16 19:24 +0000 [r47755] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c, configs/sip.conf.sample: Make the HOLD - notification optional, in order to avoid a lot of extra database - lookups for all those realtime users out there. - -2006-11-16 18:29 +0000 [r47748-47751] Joshua Colp <jcolp@digium.com> - - * channels/chan_local.c, /: Merged revisions 47750 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r47750 | file | 2006-11-16 13:26:50 -0500 (Thu, 16 Nov - 2006) | 2 lines Because of the way chan_local is written we - should be extra careful and make sure our callback functions have - a tech_pvt. (issue #8275 reported by mflorell) ........ - - * apps/app_meetme.c: Don't unreference the SLA object if there is - no SLA object in the devicestate callback. (issue #8354 reported - by loloski) - -2006-11-16 16:51 +0000 [r47733-47744] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Don't fixup if there's nothing to fixup - - * UPGRADE.txt: Warn users about change in canreinvite - - * channels/chan_sip.c, configs/sip.conf.sample: - CANCEL is never - authenticated (according to the RFC) - Update docs on - canreinvite. "nonat" is the recommended setting for most users - with phones behind a NAT. - -2006-11-15 22:31 +0000 [r47712] Joshua Colp <jcolp@digium.com> - - * channels/chan_local.c, /: Merged revisions 47711 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r47711 | file | 2006-11-15 17:29:30 -0500 (Wed, 15 Nov - 2006) | 2 lines Make sure that the pvt structure exists before - trying to do fixup on Local channels. (issue #7937 reported by - mada123, fix by alamantia with mods by me) ........ - -2006-11-15 21:56 +0000 [r47709] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Fix ODBC_STORAGE for when context is NULL - -2006-11-15 21:33 +0000 [r47707] Joshua Colp <jcolp@digium.com> - - * main/channel.c: We need to ensure timelimit stuff is included as - well so warnings get played. (issue #8050 reported by KNK) - -2006-11-15 20:50 +0000 [r47701] Kevin P. Fleming <kpfleming@digium.com> - - * main/file.c: don't try to call fclose() if fopen() failed - -2006-11-15 20:31 +0000 [r47698] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: - Improve SIP history - Never send reply to - ACK (again...) - -2006-11-15 20:31 +0000 [r47684-47697] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 47677 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r47677 | kpfleming | 2006-11-15 11:56:42 -0600 (Wed, 15 Nov 2006) - | 4 lines ensure that message duration is included in email - notifications for forwarded messages (BE-96, fix by me after - corydon used his clue-bat on me) ensure that duration in the - message metadata is updated if prepending is done during - forwarding (related to BE-96) remove prototype for API call that - does not exist ........ - - * main/config.c, /: Merged revisions 47686,47688-47689 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r47686 | kpfleming | 2006-11-15 13:42:05 -0600 (Wed, 15 - Nov 2006) | 2 lines clear the category's variable tail pointer as - well when variables are detached from it ........ r47688 | - kpfleming | 2006-11-15 13:47:43 -0600 (Wed, 15 Nov 2006) | 2 - lines when appending a list of variable to a category, ensure the - tail pointer points to the last variable in the list ........ - r47689 | kpfleming | 2006-11-15 13:58:46 -0600 (Wed, 15 Nov 2006) - | 2 lines when re-writing the config file, don't repeat the path - if it hasn't changed ........ - - * main/config.c, /: Merged revisions 47682 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r47682 | kpfleming | 2006-11-15 12:39:47 -0600 (Wed, 15 Nov 2006) - | 2 lines ouch... don't use printf, use ast_log/ast_verbose - ........ - -2006-11-15 17:46 +0000 [r47672] Luigi Rizzo <rizzo@icir.org> - - * main/cli.c: fix longest match search in find_cli. Trunk already - fixed. 1.2 not affected (well, i have no idea, the code is - totally different there). - -2006-11-15 15:25 +0000 [r47649-47656] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Send error message when we can't allocate - SIP dialog, possibly due to limitation of file descriptors. - (imported from 1.2) - -2006-11-15 04:45 +0000 [r47645] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: If NAT detection is turned on or already detected - then say NAT is active when setting the remote RTP peer when - doing early bridging. (issue #8365 reported by marcelbarbulescu) - -2006-11-15 00:19 +0000 [r47641] Kevin P. Fleming <kpfleming@digium.com> - - * main/term.c: more formatting cleanup, and avoid running off the - end of the string - -2006-11-15 00:14 +0000 [r47639] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Turn notice about unknown RTCP packet type into a - debug message instead. - -2006-11-15 00:05 +0000 [r47635] Kevin P. Fleming <kpfleming@digium.com> - - * channels/misdn/isdn_lib.c: silence compiler warning on 64-bit - platforms (this variable is an 'int' anyway, comparing it to - 'signed long' is not useful) - -2006-11-14 22:17 +0000 [r47625-47632] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c, /: Merged revisions 47631 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r47631 | file | 2006-11-14 17:15:10 -0500 (Tue, 14 Nov 2006) | 2 - lines Update copyright information in the ADSI logo blob. - ........ - - * channels/chan_sip.c: Only keep the video RTP structure around if - 1. Video support is enabled and 2. A video codec is enabled on - the dialog - - * funcs/func_uri.c: Small documentation clarification for - URIENCODE. (issue #8294 reported by salaud) - -2006-11-14 18:54 +0000 [r47621] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Conversion of res_odbc API to include ast_ - prefix did not completely transition app_voicemail when - ODBC_STORAGE is used (reported on IRC by caio1982, not in - bugtracker) - -2006-11-14 16:45 +0000 [r47617] Joshua Colp <jcolp@digium.com> - - * apps/app_amd.c: Use LOG_DEBUG to print out the indication that - app_amd is using default settings instead of using LOG_NOTICE. - This stops needless logging of this information under normal - circumstances. (issue #8361 reported by Seb7) - -2006-11-14 16:22 +0000 [r47597-47613] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Update documentation to fit the - implementation... - - * /, channels/chan_sip.c: Issue #8272 - Don't destroy dialog in - retransmission system if it's an OPTION packet from peerpoke - -2006-11-13 21:28 +0000 [r47584] Joshua Colp <jcolp@digium.com> - - * /, cdr/cdr_pgsql.c: Merged revisions 47583 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r47583 | file | 2006-11-13 16:26:36 -0500 (Mon, 13 Nov 2006) | 2 - lines Initialize global pointers for connection and result to - NULL. (issue #8356 reported by james) ........ - -2006-11-13 20:20 +0000 [r47581] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /, channels/chan_sip.c: Merged revisions 47580 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r47580 | tilghman | 2006-11-13 14:18:30 -0600 (Mon, 13 Nov 2006) - | 2 lines Having more than 255 old messages caused corruption in - the new/old count ........ - -2006-11-13 19:15 +0000 [r47576] Steve Murphy <murf@digium.com> - - * main/config.c: This solves bug 8342, whereby a crash occurs under - certain circumstances while reading a config file with comments-- - a call to CB_ADD shouldn't happen if withcomments is zero - -2006-11-13 19:11 +0000 [r47573] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * main/cli.c, channels/chan_sip.c: Re-enable old deprecated - commands - -2006-11-13 19:10 +0000 [r47572] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: - Don't reply to INVITE already replied - to when we get BYE - Declare errmsg as int. Oops. - -2006-11-13 18:18 +0000 [r47564] Steve Murphy <murf@digium.com> - - * pbx/ael/ael-test/ref.ael-test3: Eager people beat me to fixing - the messed if, but we all forgot to update the regressions. Until - now. - -2006-11-13 17:13 +0000 [r47553] Steve Murphy <murf@digium.com> - - * pbx/pbx_ael.c: AEL need not complain about parkedcalls not being - found... just confuses users - -2006-11-13 17:08 +0000 [r47542-47551] Joshua Colp <jcolp@digium.com> - - * /, apps/app_sms.c: Merged revisions 47549 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r47549 | file | 2006-11-13 12:05:32 -0500 (Mon, 13 Nov 2006) | 2 - lines When sending an SMS with a user data header properly set - the UDH flag in the first byte. (issue #8347 reported by - hoffmeis) ........ - - * main/cli.c: Free full command string upon unregistering of CLI - command. Backported from revision 47536 from rizzo. - -2006-11-13 16:00 +0000 [r47540] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Only produce error message about sip history - once - -2006-11-13 05:48 +0000 [r47527] Russell Bryant <russell@digium.com> - - * configure, acinclude.m4: AC_PROG_SED is included in autoconf - 2.60, but apparently it is not included in 2.59. So, to maintain - compatability with 2.59 since it is a small change, copy this - macro into acinclude.m4 and rename it to AST_PROG_SED. (issue - #8345) - -2006-11-13 05:46 +0000 [r47523-47526] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * res/res_odbc.c, /: Merged revisions 47525 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r47525 | tilghman | 2006-11-12 23:45:11 -0600 (Sun, 12 Nov 2006) - | 2 lines If the execute fails a second time, make sure that we - don't pass back a stale handle ........ - - * channels/chan_zap.c, /: Merged revisions 47522 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r47522 | tilghman | 2006-11-12 18:34:44 -0600 (Sun, 12 Nov 2006) - | 2 lines Don't play dialtone if the seizing the channel fails - (Bug 7754) ........ - -2006-11-12 16:12 +0000 [r47507-47513] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue 8314 - Restore auto-framing (Thanks - DEA!!!) - - * channels/chan_sip.c: Part of issue 8078 - parse even if udptl is - UDPTL in sdp... - - * channels/chan_sip.c: - Don't destroy SIP dialog because of a - failed T.38 re-invite. Wait for a bye. Final response to a - re-invite does not mean that the session dies, only that the - re-invite fails. - Keep RTP active during processing of T.38 - re-invite. If the re-invite fails, RTP needs to remain as before - the re-invite. Issue 8338 - darren1713. Please test. - - * channels/chan_sip.c: -Remove blocking of ptime: parsing in sdp - -Add some comments to t.38 code - -2006-11-12 06:23 +0000 [r47492-47497] Russell Bryant <russell@digium.com> - - * /, channels/chan_iax2.c: Merged revisions 47496 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r47496 | russell | 2006-11-12 01:09:03 -0500 (Sun, 12 Nov 2006) | - 4 lines Only do the check to determine whether the channel - calling this function is an IAX2 channel when getting the IP - address using the special argument, CURRENTCHANNEL. (issue #8341, - jcovert) ........ - - * Makefile: Add the target "menuconfig" as an alias for the - "menuselect" target. This is just a favor to users so that if you - accidentally type "make menuconfig" instead of "make menuselect", - it still works. (inspired by a comment on IRC from wangster - calling me an "especially devious asterisk developer" for having - it be menuselect instead of menuconfig. :) ) - - * main/term.c: Tweak the formatting of this new function to better - conform to coding guidelines. - -2006-11-11 02:04 +0000 [r47490] Matt O'Gorman <mogorman@digium.com> - - * main/term.c, /, main/logger.c, include/asterisk/term.h: woohoo - safe output! - -2006-11-10 22:23 +0000 [r47480] Matt Frederickson <creslin@digium.com> - - * channels/chan_zap.c: Make sure we don't use 32 bits when we only - need one bit. - -2006-11-10 21:42 +0000 [r47463-47476] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: ...and make sure that the dialog is - destroyed, even if we don't get any answer on the bye... This is - the channel that remains dead after the SIP transfer - - * channels/chan_sip.c: Add debug output while trying to trace bug - in bug report - - * channels/chan_sip.c: Make sure we destroy dialog... - - * /, channels/chan_sip.c: Small cleanup of handle_request_invite() - - imported from 1.2 with changes - -2006-11-10 19:47 +0000 [r47462] Matt Frederickson <creslin@digium.com> - - * channels/chan_zap.c: Fix for #7321. Be able to explicitly hide - callerid name for switches that bork on it. - -2006-11-10 18:56 +0000 [r47454] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Issue 8010 - Fix support for multipart - SDP (alphaque) - -2006-11-10 17:13 +0000 [r47444] Luigi Rizzo <rizzo@icir.org> - - * build_tools/prep_moduledeps: grep -m is not available on BSD, so - use head -1 instead - -2006-11-10 16:53 +0000 [r47437] Joshua Colp <jcolp@digium.com> - - * apps/app_chanspy.c: Only split up extension and context if a - value exists. (issue #8332 reported by loloski) - -2006-11-10 16:51 +0000 [r47436] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * channels/chan_mgcp.c, main/cli.c, channels/chan_sip.c, - channels/chan_skinny.c, channels/chan_h323.c, - channels/chan_iax2.c: Discussion of these CLI changes resulted in - more consistency (Bug 8236) - -2006-11-10 16:36 +0000 [r47432-47433] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_queue.c: if adding a queue member is LOG_NOTICE, then - removing them should be LOG_NOTICE, not LOG_DEBUG - - * apps/app_queue.c: reflect addition/removal of dynamic queue - members in queue_log, so that people using dialplan replacement - for AgentCallbackLogin can still track login/logout (issue #7736, - reported/patched by whoiswes but this commit was written by me - and covers all three paths for AQM/RQM) - -2006-11-10 13:04 +0000 [r47414-47418] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Rip out half implementation of 491 response - support, since it wasn't implemented properly and caused memory - leaks in the case of us getting 491's, which Asterisk actually - sends... Since it is a bit too complicated to fix this, I'll rip - it out of 1.4 and put it on the to-do-list for future releases. - Now, we handle this as congestion, which it really is. Issue - #8331 - - * channels/chan_sip.c: Fix bit definition for SIP_PAG2_CALL_ONHOLD. - Thanks fenlander! - -2006-11-10 03:44 +0000 [r47398-47405] Joshua Colp <jcolp@digium.com> - - * channels/chan_h323.c: Fix building of chan_h323 by completeing - some structure definitions. (issue #8327 reported by Mithraen) - - * apps/app_voicemail.c: Do conversion in a more easier to read and - working way for \r, \n, and \t. (issue #8324 reported by - johnlange) - -2006-11-09 21:26 +0000 [r47391] Russell Bryant <russell@digium.com> - - * apps/app_voicemail.c, channels/chan_zap.c, - build_tools/prep_moduledeps: Work around an issue that caused - menuselect to display a bogus description for app_voicemail and - chan_zap. These modules use some preprocessor directives to - determine what it will report to Asterisk as its description. - However, the way we extract this information from the source - files for menuselect is not smart enough to figure this out. - (issue #8326, #8328) - -2006-11-09 16:53 +0000 [r47380] Joshua Colp <jcolp@digium.com> - - * channels/chan_phone.c, /: Merged revisions 47379 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r47379 | file | 2006-11-09 11:48:05 -0500 (Thu, 09 Nov - 2006) | 2 lines Don't include compiler.h on kernels 2.6.18 and - higher as, well, it's apparently going to be removed. This should - make all you FC6 fans happy as your Asterisk will now build - without any mods. ........ - -2006-11-09 16:28 +0000 [r47352-47377] Russell Bryant <russell@digium.com> - - * main/cli.c: fix tab completion for "core debug channel" and "core - no debug channel" - - * main/cli.c: Fix "core show channel". Also, fix tab completion for - both "core show channel" and "core show channels". - - * main/cli.c: Fix "core debug channel <whatever>". I guess someone - needs to go through and audit every CLI command that changed - number of arguments ... - - * main/asterisk.c: revert the previous change, which actually - modified the deprecated command, "show profile". Now, actually - apply the change to "core show profile". - - * main/asterisk.c: Fix argument parsing for the "core show profile" - CLI command (fixed by rizzo in his branch, team/rizzo/astobj2) - - * main/cli.c: Fix another CLI command, "core show uptime" ... - (issue #8323, reported by johnlange, fixed by myself) - - * main/asterisk.c: fix "core show version" to reflect the new - number of arguments for this CLI command (issue #8316, kshumard) - -2006-11-08 23:14 +0000 [r47344-47348] Steve Murphy <murf@digium.com> - - * main/channel.c: This update fixes 7531 - - * channels/chan_skinny.c: Committed in behalf of 8190. - -2006-11-08 21:46 +0000 [r47333-47338] Kevin P. Fleming <kpfleming@digium.com> - - * main/frame.c: the battle over CLI command formats has broken - stuff... - - * channels/chan_sip.c: add simple fix for SDP to report proper - sample rate for G.722 media sessions - -2006-11-08 17:03 +0000 [r47323-47331] Russell Bryant <russell@digium.com> - - * utils/streamplayer.c: I occasionally get email from users that - are trying to figure out what this does, or due to some - misunderstanding as to what it is supposed to do, can't get it to - work. So, I have added some text here to hopefully explain what - this application does and does not do. - - * channels/chan_gtalk.c: Make this module build again - - * configure, configure.ac, acinclude.m4: Copy the macros from - libtool.m4 to our own acinclude.m4 such that libtool is no longer - required to be installed to be able to generated the configure - script. - -2006-11-08 07:43 +0000 [r47309-47310] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Destroy dialog properly at unload (rizzo) - -2006-11-07 23:46 +0000 [r47303] Steve Murphy <murf@digium.com> - - * channels/chan_oss.c, main/channel.c, channels/chan_phone.c, - channels/chan_misdn.c, channels/chan_skinny.c, - channels/chan_features.c, channels/chan_h323.c, - channels/chan_alsa.c, channels/chan_nbs.c, channels/chan_mgcp.c, - include/asterisk/stringfields.h, apps/app_voicemail.c, - main/pbx.c, channels/chan_vpb.cc, channels/chan_local.c, - channels/chan_zap.c, channels/chan_sip.c, res/res_features.c, - channels/chan_agent.c, main/utils.c, include/asterisk/channel.h, - channels/chan_gtalk.c, channels/chan_iax2.c: These mods are to - solve the problem in bug 7506. It's a lot of rework to solve a - fairly small problem... such is life. - -2006-11-07 20:14 +0000 [r47284-47287] Joshua Colp <jcolp@digium.com> - - * channels/chan_local.c: Make MOH work as it did before in - chan_local, without this then it can go funky when transfers and - MOH are involved. (issue #7671 reported by jmls) - -2006-11-07 18:56 +0000 [r47279] Kevin P. Fleming <kpfleming@digium.com> - - * configs/musiconhold.conf.sample: clean up sample config, and make - native file playback the more obvious default choice - -2006-11-07 18:38 +0000 [r47275] Matt O'Gorman <mogorman@digium.com> - - * apps/app_voicemail.c: large overhaul to voicemail imap support. - Allows support for more imap servers, also a better - implementation of several parts of the original work. patch - provided by 8033 with major upgrades. - -2006-11-07 17:30 +0000 [r47268] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue 8303 (lrizzo) - break instead of - continue. - -2006-11-07 13:13 +0000 [r47250] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Fixing the attack shield so it doesn't - produce attacks... Issue 8265 - never reply to an ACK - -2006-11-07 01:25 +0000 [r47239] Russell Bryant <russell@digium.com> - - * /, res/res_musiconhold.c: Merged revisions 47238 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r47238 | russell | 2006-11-06 20:22:58 -0500 (Mon, 06 - Nov 2006) | 5 lines If random order is enabled for files mode - music on hold, set a random initial position, instead of always - starting at the first file, and doing the random operation only - when switching to the next file. (bug reported by John Lange on - the asterisk-dev mailing list) ........ - -2006-11-04 18:32 +0000 [r47199] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #8284: Fixes to Invite/replaces and - transfer from "john" Thank you! - -2006-11-04 18:10 +0000 [r47192-47196] Russell Bryant <russell@digium.com> - - * main/cli.c: Fix another bug in "core set debug" ... - - * main/asterisk.c, main/cli.c: Really fix the "core set debug" and - "core set verbose" CLI commands. - - * main/cli.c: fix the "atleast" option to the "core set verbose" - and "core set debug" CLI commands - -2006-11-03 23:17 +0000 [r47176] Steve Murphy <murf@digium.com> - - * channels/chan_sip.c: This fix introduced via bug 8233 - -2006-11-03 17:53 +0000 [r47107-47108] Luigi Rizzo <rizzo@icir.org> - - * bootstrap.sh: align bootstrap.sh with the version in trunk (needs - to be blocked as it is already in trunk) - - * configure.ac: add proper environment vars to detect modules on - freebsd. (already applied to trunk so it needs to be blocked - there) - -2006-11-02 23:49 +0000 [r47051-47053] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * main/rtp.c, main/udptl.c, channels/chan_skinny.c, res/res_agi.c, - channels/chan_h323.c, apps/app_queue.c, res/res_jabber.c: More - changes making the CLI more consistent with "category verb - arguments" (continuation of issue 8236) - - * main/config.c, main/cli.c, main/channel.c, main/manager.c, - channels/chan_skinny.c, channels/chan_features.c, res/res_agi.c, - main/http.c, main/file.c, main/logger.c, main/image.c, - res/res_indications.c, main/asterisk.c, res/res_odbc.c, - channels/chan_mgcp.c, apps/app_voicemail.c, main/pbx.c, - channels/chan_local.c, main/frame.c, channels/chan_sip.c, - res/res_features.c, channels/chan_agent.c, res/res_crypto.c, - res/res_musiconhold.c, channels/chan_iax2.c, apps/app_queue.c: - Reverse change of "show" to "list" and make several other - commands more consistent with "category verb arguments" - -2006-11-02 19:56 +0000 [r46992-47015] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Move check for codec translation to - sip_call() instead of in add_sdp. No one bothers with the result - of add_sdp anyway... Yet... - - * channels/chan_sip.c: Disable code for T38 over TCP and RTP since - there's no trace of actual functionality for it :-) - -2006-11-02 17:49 +0000 [r46965] Russell Bryant <russell@digium.com> - - * /, res/res_musiconhold.c: Merged revisions 46964 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r46964 | russell | 2006-11-02 12:47:56 -0500 (Thu, 02 - Nov 2006) | 3 lines ignore files in a music on hold directory - that begin with '.' (issue #8249, cboie) ........ - -2006-11-02 17:17 +0000 [r46963] Nadi Sarrar <ns@beronet.com> - - * channels/misdn/isdn_lib.c: find_free_chan_in_stack usage fix - -2006-11-02 16:45 +0000 [r46937] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: don't send INVITE when we have determined - that we can't offer any audio formats due to lack of transcoding - support (or incorrect configuration) - -2006-11-02 16:06 +0000 [r46930] Joshua Colp <jcolp@digium.com> - - * /, channels/chan_sip.c: Merged revisions 46920 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r46920 | file | 2006-11-02 11:02:27 -0500 (Thu, 02 Nov 2006) | 2 - lines Repeat after me oej: I will at least make sure my code - compiles before I commit it. ........ - -2006-11-02 15:24 +0000 [r46901] Olle Johansson <oej@edvina.net> - - * /, channels/chan_sip.c: Dont overwrite pkt->flags (from 1.2) - -2006-11-02 14:02 +0000 [r46845-46883] Russell Bryant <russell@digium.com> - - * /, main/callerid.c: Add the missing call to free described in - issue #8268. Also, add a bunch of missing calls to free in - callerid_feed_jp(). - - * main/say.c: fix saying one hundred and two hundred in hebrew - (issue #7810, eldadran) - - * Makefile, configure, codecs/gsm/Makefile, configure.ac, - build_tools/strip_nonapi, makeopts.in: Fixes for - cross-compilation on mips (issue #8058, ywalther, with some - modifications) - - * aclocal.m4, build_tools/menuselect-deps.in, configure, - build_tools/embed_modules.xml, configure.ac: Add a check in the - configure script to determine whether ld is GNU ld or not. This - is needed because module embedding only works for gnu ld. GNU ld - is now listed as a dependency for all of the module embedding - options in menuselect. (issue #8143) - -2006-11-01 20:35 +0000 [r46822] Matt O'Gorman <mogorman@digium.com> - - * channels/chan_gtalk.c: bind address support from bug 8164 - -2006-11-01 19:49 +0000 [r46802] Steve Murphy <murf@digium.com> - - * res/res_config_odbc.c: a fix for bug 8251; the var_val needs to - accept longer strings or mass confusion and a lot of lost time is - the result - -2006-11-01 18:39 +0000 [r46780] Joshua Colp <jcolp@digium.com> - - * main/Makefile: Force poll() emulation for Darwin to always be on. - It's too broken to consider being used. This resolves the console - issue OSX users have been seeing. I would have liked to autoconf - this but I haven't been able to come up with a test case that - works. Que sera. - -2006-11-01 18:26 +0000 [r46778] Russell Bryant <russell@digium.com> - - * res/res_monitor.c, /: Merged revisions 46776 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r46776 | russell | 2006-11-01 13:24:17 -0500 (Wed, 01 Nov 2006) | - 9 lines soxmix and Asterisk expect different file extensions for - certain formats. This was already handled for the wav49 format. - However, it was not handled for ulaw and alaw. I fixed this in - such a way that using the alternate extensions for ulaw and alaw - will only happen if we know we're calling soxmix, and not a - custom script defined using the MONITOR_EXEC variable. The wav49 - processing was left alone so that external scripts will see no - behavior change. (issue #7550, reported by mnicholson, proposed - patch by junky, committed fix is a bit different) ........ - -2006-11-01 18:21 +0000 [r46775] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: It's another round of chan_iax2 fixes! - Should hopefully fix the deadlock issues people have been - reporting. IAXtel now has qualify turned on for 800 peers and it - is handling it fine. - -2006-11-01 17:48 +0000 [r46760] Steve Murphy <murf@digium.com> - - * main/config.c: Cleanups suggested by Russell. - -2006-11-01 16:39 +0000 [r46744] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c: Prevent an infinite loop when config - processing gets to a jitterbuffer option - -2006-10-31 22:02 +0000 [r46716] Jason Parker <jparker@digium.com> - - * main/translate.c: Fix "core show translation" output. Issue - #8243, patch by Damin. - -2006-10-31 21:47 +0000 [r46711-46714] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/translate.h, main/translate.c: add an API so - that translators can activate/deactivate themselves when needed - - * include/asterisk/translate.h, main/translate.c: revert changes - that were the wrong way to address this... proper fix coming - - * main/translate.c: let's set the seen flag early enough to - actually make a difference... - - * include/asterisk/translate.h, main/translate.c: don't re-do setup - operations for translators that can dynamically register - themselves - -2006-10-31 15:49 +0000 [r46663] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * /: Blocked revisions 46662 via svnmerge ........ r46662 | - tilghman | 2006-10-31 09:46:04 -0600 (Tue, 31 Oct 2006) | 3 lines - Move thread-unsafe initializer to the module loading code; add - the corresponding function to the module unload to fix a memory - leak. ........ - -2006-10-31 10:56 +0000 [r46583-46631] Olle Johansson <oej@edvina.net> - - * main/enum.c, funcs/func_enum.c, include/asterisk/enum.h: Issue - #8089 - Fix the ENUM support (picking one record by number). - Thanks otmar! - - * /, channels/chan_sip.c, configs/sip.conf.sample: Support ;rport - when we're supposed to support ;rport. Issue #7473. - - * /, channels/chan_sip.c: If peer fails ACL check, fail peer at - REGISTER - - * channels/chan_sip.c: Fix T38 too. Thanks, tgrman ! - -2006-10-31 06:30 +0000 [r46554-46563] Russell Bryant <russell@digium.com> - - * contrib/init.d/rc.redhat.asterisk: Start Asterisk later in the - boot process to ensure it starts after stuff like MySQL (issue - #8253, Alric) - - * /, main/utils.c: Merged revisions 46560 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r46560 | russell | 2006-10-31 01:18:36 -0500 (Tue, 31 Oct 2006) | - 3 lines When handling the case where the hostname is just an IPV4 - numeric address, be sure to set the address type. (issue #8247, - alexr) ........ - - * /, res/res_agi.c: Merged revisions 46557 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r46557 | russell | 2006-10-31 01:13:09 -0500 (Tue, 31 Oct 2006) | - 3 lines fix some copy/paste bugs in the checking of arguments for - the "control stream file" AGI command (issue #8255, mnicholson) - ........ - - * main/translate.c: Add a small tweak to the code that checks to - see whether destination formats are translatable based on the - source format. If we have already determined that there is no - translation path in one direction, don't bother checking the - other direction. - -2006-10-30 22:19 +0000 [r46511-46526] Kevin P. Fleming <kpfleming@digium.com> - - * main/translate.c: when unregistering a translator, don't rebuild - the translation matrix unless needed when filtering formats out - of an offer, ensure we check for translation ability in both - directions - - * include/asterisk/linkedlists.h: ensure that items removed from a - list are always unlinked from the list (next pointer set to NULL) - -2006-10-30 21:09 +0000 [r46474-46506] Joshua Colp <jcolp@digium.com> - - * configure, configure.ac: Don't explicitly link in crypt as it is - not used on some platforms. - - * channels/chan_iax2.c: We need to lock the pvt structure during - retransmission as another worker thread may be doing something as - well. - -2006-10-30 16:27 +0000 [r46382-46433] Olle Johansson <oej@edvina.net> - - * main/asterisk.c, apps/app_voicemail.c, include/asterisk/file.h, - include/asterisk/doxyref.h, channels/chan_sip.c, - main/ast_expr2f.c, include/asterisk/module.h, - formats/format_ogg_vorbis.c, main/app.c, - include/asterisk/channel.h, include/asterisk/lock.h, - include/asterisk/frame.h: Issue #8246 - Doxygen fixes from - kshumard. An extra big thankyou is given to everyone that - contributes to doxygen! THANK YOU! - - * main/rtp.c, /: Bind RTCP to the same IP as RTP - - * /, channels/chan_sip.c: Issue #7869 - Stop retransmission of 302 - redirects (imported from 1.2) - - * /, channels/chan_sip.c: Issue #7608 - Notifications sent with - wrong content-type (imported from 1.2, modified) - - * channels/chan_sip.c, CHANGES: Backport of patch for #7828 that - was reported for trunk, but obviously exists in 1.4 too. - - * channels/chan_sip.c: Restoring the old logic, since working - around it and fixing it seemed too complicated. - The - SIP_OUTGOING flag indicates the direction of the last transaction - in the dialog. - The initreq stores the last request in the - dialog, the request that opened the latest transaction. Please - now retry all the 1.4 bug reports with mixed to/from headers, - tags etc in ACK, BYE, CANCEL. Thanks! - - * channels/chan_sip.c: Accepting a message twice may be - misinterpreted... - - * channels/chan_sip.c: - 183 is not reliable message... - Error - should not have SDP - -2006-10-28 16:37 +0000 [r46377] Joshua Colp <jcolp@digium.com> - - * utils/Makefile: Don't build muted on OpenBSD, it is not - supported. - -2006-10-27 19:03 +0000 [r46370] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c: move the copy of the default settings to the - global settings back out of process_zap, so that they aren't - overwritten when process_zap is called multiple times - -2006-10-27 18:29 +0000 [r46367] Olle Johansson <oej@edvina.net> - - * contrib/asterisk-ng-doxygen: Put some doxygen pressure on - Christian :-) - -2006-10-27 17:39 +0000 [r46358-46363] Russell Bryant <russell@digium.com> - - * main/asterisk.c, res/res_agi.c, apps/app_externalivr.c, - res/res_musiconhold.c: We should always be using _exit() after a - fork() or vfork() instead of exit(). This is because exit() does - some extra cleanup which in some implementations of vfork(), for - example, can actually modify the state of the parent process, - causing very weird bugs or crashes. (issue #7971, Nick Gavrikov) - - * /: Blocked revisions 46361 via svnmerge ........ r46361 | russell - | 2006-10-27 12:36:07 -0500 (Fri, 27 Oct 2006) | 5 lines We - should always be using _exit() after a fork() or vfork() instead - of exit(). This is because exit() does some extra cleanup which - in some implementations of vfork(), for example, can actually - modify the state of the parent process, causing very weird bugs - or crashes. (issue #7971, Nick Gavrikov) ........ - - * channels/chan_zap.c: Instead of iterating all of the options once - to look for jitterbuffer options, and then again for everything - else, move the processing of jitterbuffer options into the main - loop so that there are no erroneous messages about ignoring - unknown options. (issue #8226) - -2006-10-27 10:03 +0000 [r46351-46353] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, - channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c: - Merged revisions 46350 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r46350 | crichter | 2006-10-27 11:24:01 +0200 (Fr, 27 Okt 2006) | - 1 line fixed a bug which caused chan_misdn to try to allocate 2 - times the same channel on high load, which then caused - instability of mISDN. removed a useless function from isdn_lib.c - ........ - - * channels/misdn_config.c: fixed not compile issue, which was just - introduced - - * channels/misdn_config.c, channels/chan_misdn.c, /, - channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample: - Merged revisions 46176 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r46176 | crichter | 2006-10-25 10:41:59 +0200 (Mi, 25 Okt 2006) | - 1 line added nttimeout option to configure wether we disconnect - calls on NT timeouts or not during an overlapdial session - ........ - -2006-10-26 17:57 +0000 [r46335-46340] Jason Parker <jparker@digium.com> - - * /, contrib/scripts/astgenkey.8: Merged revisions 46337 via - svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r46337 | qwell | 2006-10-26 12:47:52 -0500 (Thu, 26 Oct 2006) | 2 - lines oops - somebody forgot to change this - long ago, probably. - ........ - - * CHANGES: grammar check - -2006-10-26 16:38 +0000 [r46331] Olle Johansson <oej@edvina.net> - - * CHANGES: Corrections to changes (Multiparking is not included) - -2006-10-26 16:31 +0000 [r46329] Russell Bryant <russell@digium.com> - - * main/translate.c: - If the source has no audio or no video - portion, do not call powerof() to get the format index. - Don't - run through the audio and video loops if there is no audio or - video portion of the source If 0 is passed to powerof, it will - return -1. This value of -1 was then being used as an array index - in these loops, which caused a crash on some systems. Other than - this issue, this code works as we expected it to. If a format is - not in the source, and we have to translation path to it, it is - not offered in the list of acceptable destination formats. (fixes - issue #8231) - -2006-10-26 12:15 +0000 [r46317] Kevin P. Fleming <kpfleming@digium.com> - - * CHANGES: update to reflect G.722 addition - -2006-10-26 04:18 +0000 [r46298] Russell Bryant <russell@digium.com> - - * doc/backtrace.txt: update backtrace documentation to reflect - changes in 1.4 (issue #8230, kshumard) - -2006-10-26 01:37 +0000 [r46287] Mark Spencer <markster@digium.com> - - * main/config.c, main/manager.c: Fix config comment code - preservation code (thanks murf!) - -2006-10-25 20:14 +0000 [r46276] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Old todo note - Don't add Contact header on - BYE and Cancel - -2006-10-25 19:24 +0000 [r46253-46255] Russell Bryant <russell@digium.com> - - * configure.ac: fix error output when checking for openh323 to - refer to openh323 instead of pwlib (issue #8222, misaksen) - -2006-10-25 19:16 +0000 [r46252] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Somewhat ugly code to try to fix issue - #7608. Since the problem was not very well defined, the fix is a - bit fuzzy too... Thanks to Luigi for accidentally spotting the - possible problem! - -2006-10-25 19:08 +0000 [r46249] Russell Bryant <russell@digium.com> - - * apps/app_queue.c: update warning message to include "agi" option - (issue #8225, jmls) - -2006-10-25 18:13 +0000 [r46237-46248] Kevin P. Fleming <kpfleming@digium.com> - - * sounds/Makefile: use 1.4.3 extra sounds with corrected silence - files - - * sounds/sounds.xml, sounds/Makefile: add support for prebuilt - G.722 prompts and music on hold files - -2006-10-25 15:56 +0000 [r46214-46216] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: show settings doesn't produce a list of - similar objects, it should stay a "show" - -2006-10-25 14:32 +0000 [r46200] Kevin P. Fleming <kpfleming@digium.com> - - * main/cli.c, main/cdr.c, channels/chan_phone.c, pbx/pbx_spool.c, - channels/chan_features.c, pbx/pbx_ael.c, channels/chan_h323.c, - pbx/pbx_realtime.c, channels/chan_alsa.c, apps/app_sms.c, - main/image.c, channels/chan_nbs.c, apps/app_rpt.c, main/db.c, - cdr/cdr_custom.c, channels/chan_mgcp.c, - apps/app_parkandannounce.c, apps/app_voicemail.c, - channels/chan_sip.c, apps/app_softhangup.c, apps/app_record.c, - res/res_adsi.c, main/utils.c, apps/app_ices.c, - pbx/dundi-parser.c, channels/chan_iax2.c, apps/app_queue.c, - apps/app_getcpeid.c: apparently developers are still not aware - that they should be use ast_copy_string instead of strncpy... fix - up many more users, and fix some bugs in the process - -2006-10-25 04:58 +0000 [r46165] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * main/pbx.c: WaitExten truncates decimals of times to wait, - instead of accepting them (Bug 8208) - -2006-10-25 00:26 +0000 [r46152-46154] Kevin P. Fleming <kpfleming@digium.com> - - * main/rtp.c, main/frame.c, main/translate.c, formats/format_pcm.c, - channels/chan_h323.c, channels/chan_iax2.c, - include/asterisk/frame.h: add passthrough and file format support - for G.722 16KHz audio (issue #5084, original patch by andrew, - updated by mithraen) - - * channels/chan_sip.c, main/translate.c: code zone experiment: - don't offer formats in the outbound INVITE that aren't either - passthrough or translatable - - * main/translate.c: if multiple translators are registered for the - same source/dest combination, ensure that the lowest-cost one is - always inserted earlier in the list - -2006-10-24 20:30 +0000 [r46142] Mark Spencer <markster@digium.com> - - * res/res_agi.c: Fix FastAGI when there is no pid (bug #7628, - #8147) - -2006-10-24 19:29 +0000 [r46130] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: We need to initialize our scheduler pthread - condition... yes. - -2006-10-24 08:34 +0000 [r46114-46117] Luigi Rizzo <rizzo@icir.org> - - * main/http.c: merge 45152 don't leak descriptors in http.c - - * channels/chan_sip.c: merge 45966 refer_to_domain potentially - containing options - - * channels/chan_sip.c: merge 46026 improper checks on get_header() - return values - - * channels/chan_sip.c: merge 46045 prevent NULL args to - ast_strdupa() in chan_sip.c - -2006-10-24 05:23 +0000 [r46093] Russell Bryant <russell@digium.com> - - * Makefile: Restore the ability to remove the firmware directory - without causing the installation to fail (issue #8111) - -2006-10-24 03:53 +0000 [r46080-46083] Kevin P. Fleming <kpfleming@digium.com> - - * main/translate.c: ensure that the translation matrix is properly - lock-protected every place it is used - - * include/asterisk/translate.h, main/translate.c: add an API call - to allow channel drivers to determine which media formats are - compatible (passthrough or transcode) with the format an existing - channel is already using - - * doc/imapstorage.txt: simplify and correct voicemail IMAP storage - build instructions - -2006-10-24 03:01 +0000 [r46078] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * main/channel.c: Pass through a frame if we don't know what it is, - rather than trying to pass a NULL, which will segfault a channel - driver (Bug 8149) - -2006-10-24 01:27 +0000 [r45999-46067] Russell Bryant <russell@digium.com> - - * utils/muted.c, utils/ael_main.c: In muted.c, check the return - value of strdup. In ael_main.c, check the return value of calloc. - (issue #8157) In passing fix a few minor bugs in ael_main.c. The - last argument to strncpy() was a hard-coded 100, where it should - have been 99. I changed this to use sizeof() - 1. - - * apps/app_meetme.c: Fix the descriptions of some of the - MeetMeAdmin options (issue #8098, mflorell) - - * res/res_jabber.c: don't crash when an incoming message has no - "from" (issue #8205, jmls) - -2006-10-23 00:27 +0000 [r45928] Joshua Colp <jcolp@digium.com> - - * /, cdr/cdr_odbc.c: Merged revisions 45927 via svnmerge from - https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ - r45927 | file | 2006-10-22 20:25:28 -0400 (Sun, 22 Oct 2006) | 2 - lines Don't leak memory mmmk? ........ - -2006-10-22 21:44 +0000 [r45916] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, /: Merged revisions 45808 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r45808 | crichter | 2006-10-21 14:35:13 +0200 (Sat, 21 - Oct 2006) | 1 line fixed issue, that if chan_misdn is loaded and - couldn't be initialized it would cause a segfault after 'reload'. - Reported by Drew/Matt thx. ........ - -2006-10-21 18:49 +0000 [r45818] Russell Bryant <russell@digium.com> - - * res/res_monitor.c: Add a couple missing unregistrations of - manager actions and remove duplicate unregistrations of - applications. (issue #8194, jmls) - -2006-10-21 18:48 +0000 [r45775-45817] Joshua Colp <jcolp@digium.com> - - * main/loader.c: Don't use promotion on Darwin because it doesn't - seem to work quite right in all cases, this should solve the - unresolved symbol issue people have been seeing. - - * Makefile: Pass DESTDIR and ASTSBINDIR so that the utilities get - installed in the proper location (reported on asterisk-dev - mailing list) - -2006-10-20 07:44 +0000 [r45741] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Let's understand SIP: - REFER can create - dialog, Asterisk does not support it yet - NOTIFY can create - dialog in Asterisk's implementation (voicemail) even though we - don't support the server side of it. In this case, the standard - is a side issue ;-) - Added extened functionality for unsupported - methods (PING, PUBLISH) so we don't create PVT's for those - either. Russellb needs to judge what to do with this in 1.2, but - I think the current implementation n 1.2 is a bug since we're - sending bad replies to NOTIFY and REFER outside of dialogs - -2006-10-19 17:24 +0000 [r45678-45694] Joshua Colp <jcolp@digium.com> - - * res/res_jabber.c: Let's remember to unregister JabberStatus too - (issue #8184 reported by jmls) - - * /, apps/app_externalivr.c: Merged revisions 45691 via svnmerge - from https://origsvn.digium.com/svn/asterisk/branches/1.2 - ........ r45691 | file | 2006-10-19 13:16:37 -0400 (Thu, 19 Oct - 2006) | 2 lines Respect language selection when seeing if the - file exists (issue #8178 reported by mnicholson) ........ - - * channels/chan_sip.c: If the jitterbuffer is forced on then we - can't partially bridge (reported by wangster on #asterisk-dev) - -2006-10-19 00:59 +0000 [r45622] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Don't leak the actual thread-specific - sip_pvt struct - -2006-10-18 23:49 +0000 [r45621] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: don't leak memory when a chan_sip thread is - destroyed that has a thread-local temp_pvt allocated - -2006-10-18 21:03 +0000 [r45595] Joshua Colp <jcolp@digium.com> - - * main/asterisk.c: Don't modify things if we are using vfork as - this is very bad and may cause unexpected behavior (issue #7970 - reported by Nick Gavrikov) - -2006-10-18 11:54 +0000 [r45517] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: remove duplicate declarations - -2006-10-18 04:09 +0000 [r45464] Luigi Rizzo <rizzo@icir.org> - - * main/http.c: merge from trunk: move ast_variables_destroy() to a - better place in handle_uri() to avoid leaking memory on non - existing files. - -2006-10-18 03:02 +0000 [r45452] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Don't segfault if you're using a channel driver that - doesn't turn RTCP on - -2006-10-18 02:41 +0000 [r45439-45441] Russell Bryant <russell@digium.com> - - * main/channel.c: Don't attempt to access private data members of - the pthread_mutex_t object, because this does not work on all - linux systems. Instead, just access the reentrancy field in the - ast_mutex_info struct when DEBUG_THREADS is enabled. If - DEBUG_CHANNEL_LOCKS is enabled, the developer probably has - DEBUG_THREADS on as well. (issue #8139, me) - - * configs/sip_notify.conf.sample: update entry to reboot a snom - phone (issue #7850, pnlarsson) - -2006-10-17 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.4.0-beta3 released. - -2006-10-17 22:31 +0000 [r45408-45410] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/stringfields.h, main/ast_expr2.c, - main/channel.c, channels/chan_sip.c, channels/chan_iax2.c: - optimize the 'quick response' code a bit more... no more malloc() - or memset() for each response expand stringfields API a bit to - allow reusing the stringfield pool on a structure when needed, - and remove some unnecessary code when the structure was being - freed - -2006-10-17 20:38 +0000 [r45378-45381] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Don't create a "real" pvt structure for - requests that shouldn't be able to create one. Instead use a - temporary pvt and fill it with enough information so we can send - a reply. - -2006-10-17 17:39 +0000 [r45329] Olle Johansson <oej@edvina.net> - - * configs/sip.conf.sample: Adding information about Marks - direct-RTP hack to the docs... - -2006-10-17 17:22 +0000 [r45327] Kevin P. Fleming <kpfleming@digium.com> - - * LICENSE: provide licensing language for IAXy firmware file - -2006-10-16 20:06 +0000 [r45246-45280] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c, apps/app_directed_pickup.c: Backport of new - directed pickup (BE-85). - -2006-10-16 13:59 +0000 [r45196-45213] Olle Johansson <oej@edvina.net> - - * CREDITS: Adding Inotel to credits for SIP transfers. Thanks for - your support! - - * channels/chan_sip.c: Don't destroy dialog for unexpected REFER - response... - -2006-10-14 04:38 +0000 [r45143] Steve Murphy <murf@digium.com> - - * funcs/func_rand.c: update the doc string for both AEL and - extensions.conf users. - -2006-10-13 23:02 +0000 [r45125] Kevin P. Fleming <kpfleming@digium.com> - - * main/acl.c don't drop the entire permit/deny list when an attempt - is made to add an invalid entry (BE-92) - -2006-10-13 21:06 +0000 [r45104-45106] Joshua Colp <jcolp@digium.com> - - * res/res_speech.c: Clear the quiet flag too since we are - restarting a recognition again (reported on -dev by Stephan - Edelman) - - * res/res_speech.c: Check return value from engine in case of - failure (ie: out of licenses) (reported on -dev mailing list) - -2006-10-13 20:52 +0000 [r45103] Steve Murphy <murf@digium.com> - - * pbx/ael/ael-test/ref.ael-vtest17 (added), - pbx/ael/ael-test/ael-vtest17/extensions.ael (added), - pbx/ael/ael-test/ael-vtest17 (added), - pbx/ael/ael-test/ref.ael-test3, pbx/pbx_ael.c: Bug 8128 fixed in - this release via these changes - -2006-10-13 19:19 +0000 [r45088] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: avoiding warning, fixing potential bug - -2006-10-13 18:42 +0000 [r45051-45079] Joshua Colp <jcolp@digium.com> - - * codecs/lpc10/placev.c, codecs/lpc10/irc2pc.c, - codecs/lpc10/decode.c, codecs/lpc10/dcbias.c, - codecs/lpc10/pitsyn.c, codecs/lpc10/voicin.c, - codecs/lpc10/difmag.c, codecs/lpc10/hp100.c, - codecs/lpc10/synths.c, codecs/lpc10/preemp.c, - codecs/lpc10/rcchk.c, codecs/lpc10/lpfilt.c, - codecs/lpc10/mload.c, codecs/lpc10/lpcenc.c, - codecs/lpc10/vparms.c, codecs/lpc10/dyptrk.c, - codecs/lpc10/lpcini.c, codecs/lpc10/random.c, - codecs/lpc10/ham84.c, codecs/lpc10/chanwr.c, - codecs/lpc10/placea.c, codecs/lpc10/tbdm.c, - codecs/lpc10/analys.c, codecs/lpc10/onset.c, - codecs/lpc10/energy.c, codecs/lpc10/deemp.c, - codecs/lpc10/lpcdec.c, codecs/lpc10/ivfilt.c, - codecs/lpc10/median.c, codecs/lpc10/encode.c, - codecs/lpc10/bsynz.c, codecs/lpc10/prepro.c, - codecs/lpc10/invert.c: And file said... let the compiler warnings - STOP! - - * apps/app_chanspy.c: Turn on volume adjustment if it needs to be on (issue #8136 - reported by mnicholson) - - * apps/app_playback.c: Move say.conf existence check to do_say - function since it is called from multiple places (issue #8144 - reported by kshumard) - -2006-10-13 16:19 +0000 [r45049] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_iax2.c: when sending a call to a peer, use the proper socket if - we have multiple bindings (reported on asterisk-dev) - -2006-10-13 16:01 +0000 [r45031-45040] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Complete merging in RPID screen changes - (issue #8101 reported by hristo, patch by oej in revision 44757) - - * main/dnsmgr.c: Pass the right value to usleep for sleeping, and always add - the background refresh item back into the scheduler if enabled - since it is deleted during reload. (issue #8142 reported by - p_lindheimer) - -2006-10-13 15:41 +0000 [r45027] Kevin P. Fleming <kpfleming@digium.com> - - * configure, include/asterisk/autoconfig.h.in, configure.ac, - main/utils.c: use a configure script test for PMTU discovery - control instead of just assuming it's available on Linux - -2006-10-13 14:45 +0000 [r44994-45026] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed some - echocandisable issues when bridged. this caused a kernel panic - sometimes.. also some minor formatting fixes - - * channels/misdn/isdn_msg_parser.c: fixed issue that the hangupcause - got a wrong isdn cause at RELEASE_COMPLETE - -2006-10-12 22:07 +0000 [r44992] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: merge formatting and minor code - simplifications from trunk - -2006-10-12 20:34 +0000 [r44982] Matt O'Gorman <mogorman@digium.com> - - * channels/chan_gtalk.c: fix for bug 7764. - -2006-10-12 19:14 +0000 [r44956-44971] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: we can only send one 'a=ptime' attribute per - media session, not one for each format - - * main/netsock.c, include/asterisk/utils.h, channels/chan_sip.c, - main/utils.c: ensure that IAX2 and SIP sockets allow UDP - fragmentation when running on Linux (thanks to Brian Candler on - the asterisk-dev list for the tip) - -2006-10-12 16:56 +0000 [r44945] Russell Bryant <russell@digium.com> - - * main/manager.c: fix a silly typo in a comment that I saw while - reading the commit list - -2006-10-12 16:08 +0000 [r44942] Joshua Colp <jcolp@digium.com> - - * Makefile: Pass off AUDIO_LIBS so muted can link on OSX (issue - #8135 reported by ssokol) - -2006-10-12 12:55 +0000 [r44921] Nadi Sarrar <ns@beronet.com> - - * main/manager.c: append_event must be called while holding the - session lock - -2006-10-12 10:24 +0000 [r44911] Russell Bryant <russell@digium.com> - - * res/res_jabber.c: change some debug output to use LOG_DEBUG - instead of verbose output - -2006-10-11 16:57 +0000 [r44888] Jason Parker <jparker@digium.com> - - * main/db1-ast/Makefile: These are already set by the parent - Makefile.. There is no need to have this here (it doesn't - actually work anyways). - -2006-10-11 09:18 +0000 [r44854] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c: removed warning because of missing - prototype declaration - -2006-10-10 19:23 +0000 [r44830] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Do not set default/global values in the - variable declaration, set it in reload_config() - -2006-10-10 17:21 +0000 [r44819] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Move some stuff around so that a NOTIFY - dialog won't hang around until the end of the world under certain - circumstances - -2006-10-10 16:44 +0000 [r44809] Paul Cadach <paul@odt.east.telecom.kz> - - * main/channel.c, funcs/func_channel.c, include/asterisk/channel.h: - CHANNEL() function sometime mix parameter and value - -2006-10-10 16:42 +0000 [r44808] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * funcs/func_logic.c: Lost of a bit of logic when this was - simplified between 1.2 and 1.4 (Bug 8117) - -2006-10-10 16:30 +0000 [r44806] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Bail out if we have no refer structure and - we get a refer response - -2006-10-10 16:21 +0000 [r44805] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: more merge from trunk (comments and change a - static function name) - -2006-10-10 15:23 +0000 [r44788] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Only set DTMF information if an RTP - structure exists - -2006-10-10 13:50 +0000 [r44786] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/chan_misdn.c: (re)added - support of dynamically enabling hdlc on bchannels - -2006-10-10 08:25 +0000 [r44776-44777] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: whitespace changes related to previous - commit - - * channels/chan_sip.c: merge a few code simplifications that have - gone into trunk during last week, to reduce differences between - the two branches and make porting fixes easier. - -2006-10-09 16:12 +0000 [r44764] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Fix a problem where phones that go - "missing" never got unregistered. Issue #8067, reported by pj, - patch by Anthony LaMantia (with minor whitespace modifications) - -2006-10-09 15:46 +0000 [r44759-44760] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: iaxs[callno] may go away if we try to avoid - the deadlock - - * channels/chan_iax2.c: Properly avoid a collision with iax2_hangup - (issue #8115 reported by vazir) - -2006-10-08 14:14 +0000 [r44746] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: do not dereference p if we - know it is NULL - -2006-10-07 14:39 +0000 [r44684] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/h323/ast_h323.cxx, channels/chan_h323.c, - channels/h323/ast_h323.h, channels/h323/chan_h323.h: Propagate - caller's transfer capability too - -2006-10-07 11:37 +0000 [r44650-44665] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: put common code in a - function to avoid repetitions. - - * channels/chan_sip.c: remove hardwired usage of 5060, use - DEFAULT_SIP_PORT instead - - * channels/chan_sip.c: option_debug checking - before printing to debug channel. - - * channels/chan_sip.c: backport simplifications on sip_register, - usage of ast_set2_flag(), and fixes to the handling of failed - module loading. - - * channels/chan_sip.c: improve and document function - get_in_brackets(), introducing a helper function - find_closing_quote() of more general use. - -2006-10-06 21:28 +0000 [r44629-44631] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/linkedlists.h: ensure that mutex locks inside - list heads are initialized properly on platforms that require - constructor initialization (issue #8029, patch from timrobbins) - - * CHANGES: remove Jingle as per mog - -2006-10-06 21:08 +0000 [r44628] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Remove the seqno check for RFC2833, the handler is - smart enough to not need it. - -2006-10-06 21:07 +0000 [r44627] Kevin P. Fleming <kpfleming@digium.com> - - * CHANGES: various cleanups - -2006-10-06 18:46 +0000 [r44581-44605] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: When the sequence number rolls over then reset the - recorded sequence number for DTMF (issue #8106 reported by - bungalow) - - * main/file.c: Even more frames to treat as though the remote side - disappeared (issue #8097 reported by eldadran) - -2006-10-06 15:59 +0000 [r44567] Luigi Rizzo <rizzo@icir.org> - - * main/manager.c, main/http.c: make sure sockets are blocking when - they should be blocking. - -2006-10-06 12:53 +0000 [r44559-44563] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: fixed segfault which happens during - hold/transfer action - - * channels/chan_misdn.c: if INFORMATION Message come with keypad - instead of called party number, we just use the keypad as called - party number. - - * channels/misdn/isdn_lib.c, channels/misdn_config.c, - channels/misdn/isdn_lib.h, channels/chan_misdn.c, - channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample: - added the option 'reject_cause' to make it possible to set - the RELEASE_COMPLETE - cause on the 3. incoming PMP channel, - which is automatically rejected because chan_misdn does not - support that kind of callwaiting. Therefore chan_misdn supports - now 3 incoming channels on a PMP BRI Port. misdn_lib_get_free_bc - now gets the info if the requested channel is incoming or - outgoing to make the 3. channel possible - - * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, - channels/chan_misdn.c: fixed the hold/retrieve/transfer issues, - removed a useless bc field, added setting of frame.delivery fields, - some minor code cleanups - -2006-10-05 19:57 +0000 [r44502] Joshua Colp <jcolp@digium.com> - - * main/file.c: Treat busy control frames as hangup in the file streaming - core (issue #8097 reported by eldadran) - -2006-10-05 18:21 +0000 [r44488] Steve Murphy <murf@digium.com> - - * pbx/pbx_ael.c: This mod fixes a problem pointed out by dgarstang. - Many thanks to Doug! - -2006-10-05 18:01 +0000 [r44486] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: One more T.38 fix! Don't leave a reinvite - hanging by a thread if the other side is already setup with T.38 - -2006-10-05 16:10 +0000 [r44476] Kevin P. Fleming <kpfleming@digium.com> - - * main/app.c: don't segfault when an argument without a close - parenthesis is found stop parsing as soon as that situation - occurs - -2006-10-05 15:22 +0000 [r44465-44466] Steve Murphy <murf@digium.com> - - * CHANGES: I put the accumulated changes from the commit logs and - inspection, into CHANGES. Hope everyone approves! - - * configs/muted.conf.sample, utils/muted.c: Hang on a minute, the - install process sticks muted.conf in /etc/asterisk, so that's - where muted should look for it, right? - -2006-10-05 02:40 +0000 [r44450] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Don't totally bail out if T.38 was - negotiated - -2006-10-05 01:42 +0000 [r44433-44436] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: fix Polycom presence notification again - -2006-10-04 22:52 +0000 [r44407-44409] Luigi Rizzo <rizzo@icir.org> - - * utils/Makefile: as far as i can tell astman only uses newt... - - * Makefile: put linker flags in ASTLDFLAGS where they belong - -2006-10-04 21:17 +0000 [r44390-44393] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: remove workaround for old Polycom firmware SUBSCRIBE - requests add workaround for new Polycom firmware SUBSCRIBE - requests (bug is known to exist in 2.0.1 firmware) - - * include/asterisk.h, main/utils.c: make LOW_MEMORY builds actually - work - -2006-10-04 19:57 +0000 [r44380] Steve Murphy <murf@digium.com> - - * pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael.tab.c, - pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-ntest12, - pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3, - pbx/pbx_ael.c, pbx/ael/ael-test/ref.ael-test4, - pbx/ael/ael-test/ref.ael-test5, pbx/ael/ael-test/ref.ael-test6, - pbx/ael/ael-test/ref.ael-test7, pbx/ael/ael-test/ref.ael-test8, - pbx/ael/ael-test/ael-test16/extensions.ael (added), - pbx/ael/ael-test/ael-test16 (added), pbx/ael/ael.y, - pbx/ael/ael-test/ref.ael-test11, pbx/ael/ael-test/ref.ael-test14, - pbx/ael/ael-test/ref.ael-test15, pbx/ael/ael-test/ref.ael-ntest9, - pbx/ael/ael-test/ref.ael-test16 (added): These changes fix the - problems reported in bug 8090 - -2006-10-04 19:47 +0000 [r44378] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_oss.c, main/cdr.c, channels/chan_phone.c, - main/manager.c, pbx/pbx_spool.c, res/res_smdi.c, - channels/chan_skinny.c, channels/chan_h323.c, main/http.c, - channels/chan_alsa.c, pbx/pbx_dundi.c, apps/app_mixmonitor.c, - main/asterisk.c, channels/chan_mgcp.c, main/autoservice.c, - include/asterisk/utils.h, main/dnsmgr.c, channels/chan_zap.c, - channels/chan_sip.c, apps/app_meetme.c, res/res_snmp.c, - main/devicestate.c, main/utils.c, res/res_musiconhold.c, - channels/chan_iax2.c, apps/app_queue.c, res/res_jabber.c: update - thread creation code a bit reduce standard thread stack size - slightly to allow the pthreads library to allocate the stack+data - and not overflow a power-of-2 allocation in the kernel and waste - memory/address space add a new stack size for 'background' - threads (those that don't handle PBX calls) when LOW_MEMORY is - defined - -2006-10-04 17:04 +0000 [r44337-44365] Steve Murphy <murf@digium.com> - - * configs/muted.conf.sample: I've been meaning to add some - explanation about muted... here it is - - * configs/manager.conf.sample: CLI reverbification update to this - config file - - * apps/app_macro.c: In response to bug 7776, a Warning has been - added to the doc string for Macro(). - -2006-10-04 00:25 +0000 [r44322] Kevin P. Fleming <kpfleming@digium.com> - - * main/asterisk.c, main/loader.c, main/term.c, Makefile, - include/asterisk.h: ensure that local include files are always - used avoid a duplicate function name (term_init()) - -2006-10-03 22:35 +0000 [r44312] Matt O'Gorman <mogorman@digium.com> - - * channels/chan_gtalk.c, res/res_jabber.c: fix issue with dialing - client without resource. - -2006-10-03 20:18 +0000 [r44298] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_queue.c: fix a logic error in my previous fix to the queue - reload code - -2006-10-03 18:42 +0000 [r44286] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/h323/ast_h323.cxx: Change default presentation indicator - to "user provided not screened" if octet 3a missed in - CallingPartyNumber IE - -2006-10-03 18:35 +0000 [r44284] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Use VideoSupport instead so it is considered - a valid XML attribute name. (issue #8075 reported by renemendoza) - -2006-10-03 18:30 +0000 [r44283] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/h323/ast_h323.cxx: Fix preparation of type and - presentation of calling number - -2006-10-03 00:01 +0000 [r44240] Matt O'Gorman <mogorman@digium.com> - - * doc/jingle.txt, channels/chan_jingle.c (removed), - include/asterisk/jabber.h, configs/jingle.conf.sample (removed), - res/res_jabber.c: updated res_jabber for even better component - support, soon will be jep-0100 compliant. also removed - chan_jingle and infromed info from jingle.txt, chan_gtalk still - works and should be used in this version. - -2006-10-02 20:11 +0000 [r44199-44215] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Change the fd on the I/O context in case it - changed during the reload, which is indeed possible. (issue #7943 - reported by eclubb) - - * contrib/init.d/rc.redhat.asterisk: We should be using $AST_SBIN - instead of hardcoding the path for the error message (issue #7942 - reported by eclubb) - -2006-10-02 18:52 +0000 [r44186] Paul Cadach <paul@odt.east.telecom.kz> - - * configs/users.conf.sample, pbx/pbx_config.c: Missed part of - userconf functionality for chan_h323 - -2006-10-02 17:25 +0000 [r44169] Joshua Colp <jcolp@digium.com> - - * main/io.c: Shrink when current_ioc is unused. It is set to -1 when - unused, not 0. (issue #7941 reported by eclubb) - -2006-10-02 17:16 +0000 [r44166-44167] Paul Cadach <paul@odt.east.telecom.kz> - - * doc/realtime.txt: Typo fix - - * channels/chan_h323.c: Optimization of oh323_indicate(): less - locks - less problems, plus single exit point - -2006-10-02 02:38 +0000 [r44146] Mark Spencer <markster@digium.com> - - * channels/chan_sip.c, channels/chan_iax2.c: Don't use Channel when - you're not talking about a channel :) - -2006-10-01 19:32 +0000 [r44135] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/chan_h323.c: Do not simulate any audio tones if we got - PROGRESS message - -2006-10-01 18:30 +0000 [r44111-44125] Russell Bryant <russell@digium.com> - - * Makefile: Fix a problem that cuased AST_DATA_DIR in defaults.h to - be empty. The cause is that since ASTDATADIR is explicitly - exported using "export ASTDATADIR" at the top of the Makefile, - make no longer considers the variable "undefined", so the - Makefile can't use ?= to set ASTDATADIR if not yet set. (issue - #8063, reported by akohlsmith, fixed by me) - - * configs/queues.conf.sample: Fix the name of the "eventmemberstatus" - option in the sample queues.conf (issue #8065, adamg) - -2006-10-01 15:01 +0000 [r44109] Luigi Rizzo <rizzo@icir.org> - - * channels/chan_sip.c: sync with trunk - move variable declarations - to the beginning of a block. - -2006-09-30 19:20 +0000 [r44090] Paul Cadach <paul@odt.east.telecom.kz> - - * main/rtp.c: Allow one-way RTP streams (device->Asterisk) - -2006-09-30 16:28 +0000 [r44080] Luigi Rizzo <rizzo@icir.org> - - * codecs/lpc10/Makefile, Makefile, main/Makefile: fix two recent - build problems: - with AST_DEVMODE, building codecs/lpc10 fails - because of lots of warnings, and the configure step in editline - fails as well. Fix this by removing the -Werror in these steps. - - on FreeBSD (but probably on other platforms as well), the final - link of asterisk fails because AST_LIBS was not exported to the - subdirs Makefiles. Add a proper fix in the top-level Makefile (a - possible alternative way is to add "export AST_LIBS" near the - beginning of the file). With this fix, i believe that some of the - platform-specific conditionals in main/Makefile are redundant - (because they should be already dealt with in the top level - Makefile) but i don't have a platform to check. Merging to head - will happen in a moment. - -2006-09-30 16:12 +0000 [r44068-44078] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/chan_sip.c: Fix issue #7928 correctly. Next is a comment - of previous fix: Issue #7928 - Don't send both 404 and 503. Fix - by phsultan with a small fix by me, myself or I. Thanks, - Philippe! (This was caused by my changes to the transaction - handling) - - * channels/chan_sip.c: Found some buggy SIP clients (phones Planet - VIP-153T firmware 1.0, Linksys PAP2 firmware 3.1.9(LSc)) which - sends ACK not on OK message only (when remote party answers) but - on RINGING message too, so when we send 200 OK message, we get - unidentified ACK message (because INVITE acknowledged on RINGING - message already), so 200 OK retransmits within its retransmission - interval then call gets dropped. If someone else knows how to - provide workaround for such cases, please, fix it in correct way. - Thanks to ssh from #asteriskru for provide access to his box to - study and fix this case. - -2006-09-29 22:51 +0000 [r44055-44057] Kevin P. Fleming <kpfleming@digium.com> - - * agi, utils: ignore temporary files made by the Makefiles during a - build - - * codecs/lpc10/Makefile, main/db1-ast/Makefile, agi/Makefile, - codecs/Makefile, utils/Makefile, configure, - build_tools/embed_modules.xml, codecs/gsm/Makefile, configure.ac, - Makefile.moddir_rules, Makefile.rules, codecs/ilbc/Makefile, - pbx/Makefile, res/Makefile, channels/Makefile: fix a few build - system bugs, and convert Makefiles to be compatible with GNU make - 3.80 - -2006-09-29 22:35 +0000 [r44053] Jason Parker <jparker@digium.com> - - * main/asterisk.c, main/cli.c: Fix a bug with the removal of - 'atleast' argument to 'core verbose' and 'core debug'. Add that - argument back in. - -2006-09-29 21:09 +0000 [r44022-44043] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/h323/ast_h323.cxx: Set TON/PRESENTATION information more - carefully when no CallingNumber IE available - - * channels/h323/ast_h323.cxx: Fake display name by called number on - incoming calls (until passing connected number/connected name is - not implemented) - - * channels/h323/ast_h323.cxx: Ported code refers to H.450 - add - includes - - * channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Properly - pass TON/PRESENTATION information - original - H323Connection::SendSignalSetup() destroys Q.931 fields. - -2006-09-29 18:49 +0000 [r44011-44012] Kevin P. Fleming <kpfleming@digium.com> - - * main/Makefile: yet another place where we were not using the - correct CFLAGS by default - - * main/Makefile: missed one conversion to ASTCFLAGS - -2006-09-29 18:30 +0000 [r44009] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/h323/ast_h323.cxx, channels/chan_h323.c, - channels/h323/ast_h323.h, channels/h323/chan_h323.h: Pass - TON/PRESENTATION information too - -2006-09-29 18:25 +0000 [r43952-44008] Kevin P. Fleming <kpfleming@digium.com> - - * main/db1-ast/Makefile, Makefile, codecs/Makefile, utils/Makefile, - main/Makefile, codecs/gsm/Makefile, Makefile.moddir_rules, - Makefile.rules, pbx/Makefile, channels/Makefile: don't abuse - CFLAGS and LDFLAGS for build of Asterisk components, because they - are also then used for non-Asterisk components (like menuselect); - use our own variables instead - - * configure, configure.ac: support --without-curl in configure - script - - * Makefile.rules: another cross-compile fix - - * Makefile: a couple more environment settings that can't leak into - the menuselect build - - * main/cli.c: proper fix for ast_group_t change - - * include/asterisk/lock.h: eliminate compiler warning when - DEBUG_CHANNEL_LOCKS is enabled and users of this header file - don't also include channel.h - -2006-09-28 20:11 +0000 [r43944] Jason Parker <jparker@digium.com> - - * apps/app_queue.c: Fix incorrect argument order for member names, - on persisted members. Issue 8047, patch by jmls. - -2006-09-28 18:05 +0000 [r43932-43933] Joshua Colp <jcolp@digium.com> - - * apps/app_playback.c, res/res_monitor.c, - include/asterisk/logger.h, channels/chan_misdn.c, res/res_smdi.c, - channels/chan_skinny.c, apps/app_rpt.c, channels/chan_mgcp.c, - main/udptl.c, main/frame.c, funcs/func_timeout.c, - channels/chan_sip.c, apps/app_festival.c, - channels/iax2-provision.c, apps/app_alarmreceiver.c, - res/res_musiconhold.c, apps/app_followme.c, channels/chan_iax2.c: - Put in missing \ns on the end of ast_logs (issue #7936 reported - by wojtekka) - -2006-09-28 17:35 +0000 [r43919] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_queue.c: fix buggy (and overly complex) loop used during reload - of app_queue for static member list updating - -2006-09-28 17:34 +0000 [r43918] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/h323/ast_h323.cxx: Extend call establishment timeout - -2006-09-28 17:31 +0000 [r43913-43915] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Make sure the pvt exists before accessing - it again as it may have gone away (issue #7562 reported by Seb7 - and issue #7939 reported by sorg) - - * main/cli.c: Warning be gone! - -2006-09-28 16:41 +0000 [r43899] BJ Weschke <bweschke@btwtech.com> - - * apps/app_queue.c: app_queue is comparing the device names incorrectly - while checking their statuses. It's internal list of interfaces - includes the dial string, while the argument passed to this - function does not have the dial string (/n for a local channel). - This causes it to ignore the device state changes because it - thinks it belongs to none of its members. (#8040 reported and - patch by tim_ringenbach) - -2006-09-28 16:17 +0000 [r43893] Joshua Colp <jcolp@digium.com> - - * apps/app_meetme.c: Stop the stream after waitstream returns so that our - formats get restored. (issue #7370 reported by kryptolus) - -2006-09-28 15:56 +0000 [r43877] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/h323/ast_h323.cxx: Fix compiler warning - -2006-09-28 15:29 +0000 [r43864-43873] BJ Weschke <bweschke@btwtech.com> - - * apps/app_queue.c: Fix race conditioon crash with get_member_status (#7864 - - tim_ringenbach reported and patched) - - * apps/app_queue.c: Autopause not working for queue members. (#8042 - - jmls reported and patch) - -2006-09-28 12:58 +0000 [r43861-43862] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Force - remote side to start media on outgoing PROGRESS message - - * include/asterisk/compiler.h: Put attribute tag at correct place - -2006-09-28 11:03 +0000 [r43852] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, - channels/chan_misdn.c: fixed a bug which led to chan_list zombies, - when the call could not be properly established in misdn_call. - also removed the ACK_HDLC stuff which is not really needed. - -2006-09-28 10:51 +0000 [r43843-43846] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/h323/ast_h323.cxx: Do not open transmit channel until - TCS is received - - * main/file.c: Don't warn on HOLD/UNHOLD control frames - - * main/file.c: Don't treat unknown control frames as voice - -2006-09-27 20:21 +0000 [r43816] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Avoid inability to lock directory log message by - creating the directory ahead of time. (Issue 7631) - -2006-09-27 19:44 +0000 [r43801-43803] Jason Parker <jparker@digium.com> - - * apps/app_playback.c, main/pbx.c: Fix an issue with PLAYBACKSTATUS - not being set under certain circumstances. Fix a minor issue, to - make it use the filenames that were parsed, instead of the entire - argument string. Fix Background() to return -1 like Playback(), - if no args are specified. - -2006-09-27 19:10 +0000 [r43783-43798] Joshua Colp <jcolp@digium.com> - - * main/rtp.c: Compensate for out of order packets better if RFC2833 - compensation is turned on. - - * channels/chan_iax2.c: Get rid of two functions from a time now - past (we THINK these are from pre-recursive lock time) that may - be contributing to two open issues on the bug tracker (7562/7939) - and that has the potential to just make bad things happen if the - timing is right. - -2006-09-27 16:55 +0000 [r43779] Russell Bryant <russell@digium.com> - - * main/channel.c,res/res_features.c: Fix a problem that occurred if - a user entered a digit - that matched a bridge feature that was configured using multiple - digits, and the digit that was pressed timed out in the feature - digit timeout period. For example, if blind transfer is - configured as '##', and a user presses just '#'. In this - situation, the call would lock up and no longer pass any frames. - (issue #7977 reported by festr, and issue #7982 reported by - michaels and valuable input provided by mneuhauser and kuj. Fixed - by me, with testing help and peer review from Joshua Colp). There - are a couple of issues involved in this fix: 1) When - ast_generic_bridge determines that there has been a timeout, it - returned AST_BRIDGE_RETRY. Then, when ast_channel_bridge gets - this result, it calls ast_generic_bridge over again with the same - timestamp for the next event. This results in an endless loop of - nothing until the call is terminated. This is resolved by simply - changing ast_generic_bridge to return AST_BRIDGE_COMPLETE when it - sees a timeout. 2) I also changed ast_channel_bridge such that if - in the process of calculating the time until the next event, it - knows a timeout has already occured, to immediately return - AST_BRIDGE_COMPLETE instead of attempting to bridge the channels - anyway. 3) In the process of testing the previous two changes, I - ran into a problem in res_features where ast_channel_bridge would - return because it determined that there was a timeout. However, - ast_bridge_call in res_features would then determine by its own - calculation that there was still 1 ms before the timeout really - occurs. It would then proceed, and since the bridge broke out and - did *not* return a frame, it interpreted this as the call was - over and hung up the channels. The reason for this was because - ast_bridge_call in res_features and ast_channel_bridge in - channel.c were using different times for their calculations. - channel.c uses the start_time on the bridge config, which is the - time that the feature digit was recieved. However, res_features - had another time, 'start', which was set right before calling - ast_channel_bridge. 'start' will always be slightly after - start_time in the bridge config, and sometimes enough to round up - to one ms. This is fixed by making ast_bridge_call use the same - time as ast_channel_bridge for the timeout calculation. ........ - -2006-09-27 16:24 +0000 [r43775] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, channels/Makefile: removed the chan_misdn - versioning, since Asterisk has it's own - -2006-09-27 16:23 +0000 [r43774] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Make rfc2833compensate a global option. - -2006-09-27 04:35 +0000 [r43756] Russell Bryant <russell@digium.com> - - * apps/app_voicemail.c: Backport revision 43754 from the trunk, - which removes an unused buffer from mm_login to close bug 8038, - as well as addresses some formatting and coding guidelines issues - in passing. Originally, I did not commit this to 1.4 since it is - not necessarily fixing a bug. However, since the IMAP storage - code is brand new, I decided it would be better to make the - change here as well, in case someone has to work on this code to - address issues in the very near future. I don't want to make - unnecessary merge problems going to the trunk. - -2006-09-27 02:32 +0000 [r43739] Steve Murphy <murf@digium.com> - - * configs/extensions.ael.sample: This change to extensions.ael was - to fix bug 8031; the install scripts are causing it to be copied - to /etc/asterisk/extensions.ael, and because it is a fairly - direct conversion of the original extensions.conf, the macro and - context names clash with the existing extensions.conf. So, I put - an ael- in front of all macros and contexts, and checked every - goto and macro call. Also, this file compiles under aelparse. - -2006-09-26 20:56 +0000 [r43710] Russell Bryant <russell@digium.com> - - * main/asterisk.c: Back in revision 4798, this message was changed from - using ast_cli() to directly calling write(). During this change, - checking if this was a remote console was removed. This caused - this message about using "exit" or "quit" to exit an Asterisk - console to come up in times where it did not make sense. This - change restores the check to see if this is a remote console - before printing the message. (fixes BE-65) - -2006-09-26 20:47 +0000 [r43707] Joshua Colp <jcolp@digium.com> - - * .cleancount, main/cli.c, channels/chan_sip.c, - include/asterisk/channel.h: Use proper type to represent the group variable - (issue #8025 reported by makoto) - -2006-09-26 20:30 +0000 [r43700-43703] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Add missing newline character in the warning - message about deprecated TOS values in configuration. - - * apps/app_voicemail.c: When parsing the sections of voicemail.conf that contain - mailbox definitions, don't introduce a length limit on the - definition by using a 256 byte temporary storage buffer. Instead, - make the temporary buffer just as big as it needs to be to hold - the entire mailbox definition. (fixes BE-68) - -2006-09-26 20:19 +0000 [r43695-43697] Joshua Colp <jcolp@digium.com> - - * channels/chan_local.c: Strip options off the argument passed for - devicestate in chan_local. (issue #8034 reported by pcardozo) - - * apps/app_chanspy.c, main/channel.c, main/slinfactory.c: Slight - overhaul of the whisper support. 1. We need to duplicate the - frame from ast_translate 2. We need to ensure we always have - signed linear coming in for signed linear combining. 3. We need - to ensure we are always feeding signed linear out. 4. Properly - store and restore write format when beeping on the channel we are - whispering on. 5. Properly discontinue the stream on the channel - for the beep. (issue #8019 reported by timkelly1980) - -2006-09-26 18:34 +0000 [r43676] Kevin P. Fleming <kpfleming@digium.com> - - * sounds/Makefile: update to use 1.4.3 core sounds, with corrected - beep/beeperr/tt-monkeys files - -2006-09-26 18:08 +0000 [r43650-43674] Jason Parker <jparker@digium.com> - - * doc/rtp-packetization.txt, main/frame.c: Issue #8015, patch by - Dan Austin. Maximum values were incorrect, which is why this is - being put in 1.4 - - * channels/chan_skinny.c: Add proper codec support to chan_skinny. - Works with at least ulaw, alaw, and g729a. This is technically a - "new feature", but there are justifications for it. I found a bug - with the recent rtp packetization changes, which caused the media - setup to fail under certain circumstances, particularly when - using allow=all, or having no allow= statements (globally or on - the device). I could have either removed the rtp packetization - features, or I could add proper codec support (which, without, I - think most people would consider to be a bug anyways). - -2006-09-25 22:07 +0000 [r43640-43642] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Should have moved these lines up in the - merge, instead of removing them - - * apps/app_voicemail.c: Two bugs when forwarding voicemail (Issue 7824): 1) - delete=yes was ignored 2) maxmessages was ignored - -2006-09-25 21:26 +0000 [r43626-43635] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/h323/cisco-h225.cxx, channels/h323/cisco-h225.h, - channels/h323/cisco-h225.asn: Fix ASN1 description of - non-standard Cisco extensions - - * channels/h323/ast_h323.cxx, channels/chan_h323.c: Backport - changes of trunk: 1) r43540: Avoid possible deadlock on channel - destruction 2) r43590: Disable fastStart if requested by remote - side - -2006-09-25 15:23 +0000 [r43616] Jason Parker <jparker@digium.com> - - * sounds/Makefile: One more fix for sounds installation - this time - for portability. Reported to asterisk-dev mailing list. - -2006-09-25 14:52 +0000 [r43605] Steve Murphy <murf@digium.com> - - * formats/format_ogg_vorbis.c: This tiny fix prevents asterisk from - crashing if trying to play an OGG moh file. - -2006-09-25 06:15 +0000 [r43582] Paul Cadach <paul@odt.east.telecom.kz> - - * channels/h323/caps_h323.cxx, channels/h323/compat_h323.h, - channels/chan_h323.c: Merged revisions 43472,43495 from trunk - -2006-09-24 14:58 +0000 [r43553-43564] Russell Bryant <russell@digium.com> - - * channels/iax2-provision.c: Fix a CLI command registration issue - where an erroneous message claiming that "iax2 show provisioning" - was already registered. This was because this command was - registering itself as both the command, as well as the command it - is deprecating. (issue #8022, reported by bjweeks, fixed by - myself) - - * channels/chan_iax2.c:Check to see if the channel that is activating the - IAXPEER function is actually an IAX2 channel before proceeding to - process it to avoid crashing. (issue #8017, reported by admott, - fixed by myself) - -2006-09-22 23:44 +0000 [r43524] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile: don't output the 'build complete' message when the - target being run is already going to do an installation - -2006-09-22 22:12 +0000 [r43518] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Allow chan_skinny.so to be unloaded - properly. Remove reload support, since it doesn't - actually...work. - -2006-09-22 21:36 +0000 [r43505-43508] Steve Murphy <murf@digium.com> - - * pbx/pbx_ael.c: This commits a change to return - MODULE_LOAD_FAILURE on error, and SUCCESS (instead of 0) when all - goes well for bug 8004 - - * pbx/pbx_ael.c: If the extensions.ael file not found, or - unreadable, we return AST_MODULE_LOAD_DECLINE, as per bug # 8004. - -2006-09-22 17:25 +0000 [r43492] Jason Parker <jparker@digium.com> - - * main/cli.c: Make sure we explicitly set the CLI command to not be - deprecated, if it isn't. - -2006-09-22 16:42 +0000 [r43486-43489] Kevin P. Fleming <kpfleming@digium.com> - - * sounds/Makefile: use rebuilt extra sounds - - * main/channel.c: all the Linux systems I have don't use - '__m_count' for this field, so I don't know where this came - from... - -2006-09-22 15:47 +0000 [r43477-43484] Russell Bryant <russell@digium.com> - - * include/asterisk/threadstorage.h: backport the compatability fix - to use attribute_malloc instaed of __attribute__ ((malloc)) - - * channels/chan_misdn.c: return AST_MODULE_LOAD_DECLIDE if mISDN - could not be configured (issue #8006, Mithraen) - - * main/frame.c: Suppress a compiler warning about the use of a - potentially uninitialized variable. It couldn't actually happen, - though. - -2006-09-22 03:01 +0000 [r43469] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: First shot at unload_module in - chan_skinny.. More to come. - -2006-09-21 23:50 +0000 [r43466] Matt O'Gorman <mogorman@digium.com> - - * include/asterisk/jabber.h, channels/chan_gtalk.c, - res/res_jabber.c: updates for better compontent support - -2006-09-21 23:24 +0000 [r43464] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * res/res_odbc.c, configs/res_odbc.conf.sample: Twould help if we - actually documented how the new features in res_odbc actually - work. (Oops) - -2006-09-21 22:21 +0000 [r43454-43456] Joshua Colp <jcolp@digium.com> - - * channels/chan_oss.c: Some more clean up in the load function for - chan_oss (issue #8002 reported by Mithraen with minor mods by - moi) - - * channels/chan_mgcp.c: Clean up chan_mgcp's module load function - (issue #8001 reported by Mithraen with mods by moi) - -2006-09-21 21:21 +0000 [r43450] Kevin P. Fleming <kpfleming@digium.com> - - * main/Makefile, build_tools/strip_nonapi (added): add another - attempt to strip non-API symbols from the final binary... script - will need to be extended to work on non-Linux systems - -2006-09-21 20:22 +0000 [r43410-43445] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_url.c: Fix documentation to reflect how Url() really - works - - * cdr/cdr_tds.c, configure, configure.ac: TDS 0.64 updates - -2006-09-21 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.4.0-beta2 released. - -2006-09-21 16:08 +0000 [r43404-43405] Kevin P. Fleming <kpfleming@digium.com> - - * main/Makefile: remove this change... it requires binutils 2.17 - -2006-09-20 23:19 +0000 [r43396] Jason Parker <jparker@digium.com> - - * build_tools/make_version: fix minor typo in the way version is - handled - -2006-09-20 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.4.0-beta1 released. diff --git a/res/res_config_odbc.c b/res/res_config_odbc.c index 95e96990d..ac3cffd16 100644 --- a/res/res_config_odbc.c +++ b/res/res_config_odbc.c @@ -57,7 +57,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$") struct custom_prepare_struct { const char *sql; const char *extra; - va_list *ap; + va_list ap; }; static SQLHSTMT custom_prepare(struct odbc_obj *obj, void *data) @@ -67,7 +67,8 @@ static SQLHSTMT custom_prepare(struct odbc_obj *obj, void *data) const char *newparam, *newval; SQLHSTMT stmt; va_list ap; - va_copy(ap, *(cps->ap)); + + va_copy(ap, cps->ap); res = SQLAllocHandle(SQL_HANDLE_STMT, obj->con, &stmt); if ((res != SQL_SUCCESS) && (res != SQL_SUCCESS_WITH_INFO)) { @@ -115,8 +116,9 @@ static struct ast_variable *realtime_odbc(const char *database, const char *tabl SQLSMALLINT nullable; SQLLEN indicator; va_list aq; - struct custom_prepare_struct cps = { .sql = sql, .ap = &ap }; - + struct custom_prepare_struct cps = { .sql = sql }; + + va_copy(cps.ap, ap); va_copy(aq, ap); if (!table) @@ -240,9 +242,10 @@ static struct ast_config *realtime_multi_odbc(const char *database, const char * SQLSMALLINT decimaldigits; SQLSMALLINT nullable; SQLLEN indicator; - struct custom_prepare_struct cps = { .sql = sql, .ap = &ap }; - + struct custom_prepare_struct cps = { .sql = sql }; va_list aq; + + va_copy(cps.ap, ap); va_copy(aq, ap); if (!table) @@ -357,8 +360,9 @@ static int update_odbc(const char *database, const char *table, const char *keyf const char *newparam, *newval; int res; va_list aq; - struct custom_prepare_struct cps = { .sql = sql, .ap = &ap, .extra = lookup }; - + struct custom_prepare_struct cps = { .sql = sql, .extra = lookup }; + + va_copy(cps.ap, ap); va_copy(aq, ap); if (!table) |