2007-08-07 Russell Bryant * Asterisk 1.4.10 released. 2007-08-07 20:57 +0000 [r78488] Russell Bryant * res/res_config_odbc.c: Fix the build of this module on 64-bit platforms 2007-08-07 19:43 +0000 [r78450] Mark Michelson * 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 * 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 * channels/chan_sip.c: Allow chan_sip to build in devmode 2007-08-07 19:09 +0000 [r78415] Tilghman Lesher * 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 * 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 * 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 * main/channel.c: Add additional DTMF log messages to help when debugging issues. 2007-08-06 20:44 +0000 [r78184-78242] Russell Bryant * 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 * 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 * 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 * 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 * 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 * cdr/cdr_pgsql.c: Portability fix for devmode compiling (closes bug #10382) 2007-08-05 04:15 +0000 [r78143] Russell Bryant * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * apps/app_voicemail.c: Voicemail with ODBC_STORAGE defined does not compile cleanly (missing def) 2007-08-01 21:08 +0000 [r77883] Jason Parker * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * res/res_odbc.c: Missing newline 2007-07-27 17:04 +0000 [r77536-77540] Joshua Colp * 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 * channels/chan_sip.c: "re-invite" was misspelled 2007-07-26 23:19 +0000 [r77460] Joshua Colp * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /: so are my fingers... * /: autotagexternals script is still obviously misbehaving... * /: use autotagged externals 2007-07-25 17:14 +0000 [r77071] Joshua Colp * 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 * main/rtp.c: set the sequence number in a frame for all frame types 2007-07-25 00:18 +0000 [r76983] Steve Murphy * 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 * /, 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 * Asterisk 1.4.9 released. 2007-07-24 16:42 +0000 [r76803-76805] Jason Parker * /: 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * /: 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 * /, 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 * 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 * 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 * 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 * 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 * /, 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 * channels/chan_skinny.c: Remove some duplicate code. 2007-07-19 18:59 +0000 [r75969-75978] Mark Michelson * 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 * /, 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 * 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 * /, 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 * 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 * channels/chan_skinny.c: Umm, why are we transmitting dialtone on cfwdall? 2007-07-18 20:00 +0000 [r75712] Joshua Colp * 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 * 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 * /, 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 * 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 * apps/app_meetme.c: Using a freed frame causes crashes (closes issue #9317) 2007-07-17 Russell Bryant * Asterisk 1.4.8 released. 2007-07-17 20:57 +0000 [r75441-75450] Russell Bryant * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * /: block change that is not relevant here 2007-07-13 20:36 +0000 [r75108] Russell Bryant * /, 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * /, 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 * /, 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 * /, 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 * 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 * 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 * /: 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 * 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 * /, 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 * Asterisk 1.4.7.1 released. 2007-07-10 16:00 +0000 [r74323] Russell Bryant * res/res_musiconhold.c: fix an uninitialized variable 2007-07-10 15:38 +0000 [r74317] Jason Parker * 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 * 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 * /, 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 * Asterisk 1.4.7 released. 2007-07-09 21:31 +0000 [r74162-74211] Russell Bryant * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * /, 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 * 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 * /, 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * /: What the heck. This should not have happened. * /: use autotagged externals 2007-07-03 12:38 +0000 [r73053] Tilghman Lesher * 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 * 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 * 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 * 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 * /: 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 * main/channel.c: Added additional DTMF debug messages for when emulation occurs. 2007-07-02 08:41 +0000 [r72850-72852] Christian Richter * 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 * 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 * 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 * 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 * Asterisk 1.4.6 released. 2007-06-29 16:31 +0000 [r72630] Russell Bryant * /: 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 * 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 * 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 * configure, include/asterisk/autoconfig.h.in: regenerate the configure script for rizzo 2007-06-28 19:29 +0000 [r72453-72489] Luigi Rizzo * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * /: 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 * 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 * 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 * 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 * 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 * pbx/pbx_dundi.c: Make unloading of pbx_dundi actually work. 2007-06-26 23:02 +0000 [r71953] Mark Michelson * 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 * main/rtp.c: Don't dereference a pointer that may be NULL here. Issue 10017. 2007-06-26 19:00 +0000 [r71877] Mark Michelson * 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 * /: 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 * 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 * 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 * 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 * doc/asterisk-mib.txt: Fix a typo in the Asterisk mib. (issue #10048, Matti) 2007-06-25 01:10 +0000 [r71412-71430] Joshua Colp * /, 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 * 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 * 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 * 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 * 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 * /: 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * apps/app_meetme.c: Remove a couple of duplicate unlocks 2007-06-21 13:58 +0000 [r70677] Joshua Colp * apps/app_voicemail.c: Fix building with ODBC storage enabled. (issue #10025 reported by denisgalvao) 2007-06-21 13:00 +0000 [r70656] Steve Murphy * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /: 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * Makefile: Tweak paths for BSD systems (issue #10001, stuarth) 2007-06-18 13:55 +0000 [r69625] Joshua Colp * 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 * Asterisk 1.4.5 released. 2007-06-15 20:18 +0000 [r69579] Russell Bryant * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * apps/app_directory.c: Issue 9947 - fn2 was unused / incorrectly used 2007-06-11 16:57 +0000 [r68733] Christian Richter * 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 * 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 * 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 * pbx/pbx_config.c: "dialplan save" produced garbage in the config file 2007-06-08 22:23 +0000 [r68527] Russell Bryant * /, 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 * 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 * /, 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * channels/chan_features.c: Disable chan_features by default in menuselect 2007-06-07 19:30 +0000 [r68192] Russell Bryant * main/strcompat.c: Include stdarg.h for build issues on Solaris (issue #9381) 2007-06-07 18:39 +0000 [r68071-68157] Joshua Colp * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * main/rtp.c: Add a missing \n. (pointed out by jcmoore on IRC) 2007-06-04 19:31 +0000 [r67064-67068] Joshua Colp * 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 * 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 * 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 * res/res_jabber.c: Issue 9739 - Malformed jid causes a crash 2007-06-04 15:47 +0000 [r67018-67020] Russell Bryant * 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 * 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 * 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 * 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 * doc/asterisk.8: Issue 9850 - update preferred command line syntax 2007-05-31 18:41 +0000 [r66775] Russell Bryant * 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 * /, 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 * /, 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * doc/datastores.txt: Update datastores documentation. (issue #9801 reported by mnicholson) 2007-05-29 09:41 +0000 [r66363] Olle Johansson * /, 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 * channels/chan_zap.c: Make the usedistinctiveringdetection option work again. (issue #9823 reported by premeau) 2007-05-27 04:12 +0000 [r66244] Jason Parker * 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 * configure, configure.ac: have to check for OSP toolkit _after_ checking for OpenSSL 2007-05-25 14:41 +0000 [r66159] Tilghman Lesher * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * .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 * funcs/func_math.c: merged qwell's func_math patch for issue 9507 2007-05-24 15:08 +0000 [r65863] Joshua Colp * main/rtp.c: I like it when the RTP stack compiles myself... 2007-05-24 15:05 +0000 [r65857] Olle Johansson * 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 * 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 * channels/chan_gtalk.c: Issue #8536 - Caller ID not set in CDR for jingle 2007-05-24 14:42 +0000 [r65839] Joshua Colp * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * apps/app_meetme.c: Remove a double const. 2007-05-22 14:02 +0000 [r65408] BJ Weschke * apps/app_followme.c: Fix a problem with flag recognition. 2007-05-22 13:09 +0000 [r65394] Russell Bryant * /, 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 * 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 * 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 * 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 * /, 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 * /, 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 * channels/chan_sip.c: Issue 9487 - stop media flows at hangup of call 2007-05-18 08:58 +0000 [r64904] Christian Richter * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * /: 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 * 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 * 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 * channels/chan_sip.c: Change -2 to XMIT_ERROR to clarify a bit more 2007-05-14 19:13 +0000 [r64306] Russell Bryant * 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 * 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 * 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 * 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 * main/channel.c: Add hangupcause when we lack codecs for transcoding 2007-05-12 22:27 +0000 [r64044-64114] Joshua Colp * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * main/callerid.c: Add test for negative offsets in cid data to prevent infinite loops. 2007-05-10 20:46 +0000 [r63749] Olle Johansson * /, 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 * 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 * 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 * 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 * /, 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 * 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 * /, 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * configs/manager.conf.sample: explanation for httptimeout in manager.conf 2007-05-03 16:44 +0000 [r62989] Joshua Colp * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * /, 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 * 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 * /, 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 * 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 * configs/iax.conf.sample: Remove unused (and potentially confusing) jitterbuffer options from sample config. 2007-04-30 14:36 +0000 [r62369] Joshua Colp * 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 * 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 * Asterisk 1.4.4 released. 2007-04-27 21:10 +0000 [r62218] Russell Bryant * 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 * /, 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 * 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 * /, 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 * /, 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 * 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 * /, 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 * 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 * 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 * Asterisk 1.4.3 released. 2007-04-24 21:34 +0000 [r61781-61787] Russell Bryant * 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 * 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 * 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 * /, 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 * 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 * 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 * main/rtp.c: Remove a stray debug message introduced by a recent commit. 2007-04-20 19:51 +0000 [r61694] Jason Parker * /, 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 * 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 * /, 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 * 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 * sounds/Makefile: allow external build systems to extract the required sound file versions 2007-04-18 20:46 +0000 [r61674-61676] Olle Johansson * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * apps/app_queue.c: Use the appropriate interface name with COMPLETECALLER. Issue 9395. 2007-04-09 18:32 +0000 [r60989] Steve Murphy * 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 * res/res_jabber.c: Add final new line after JabberEvent 2007-04-09 17:22 +0000 [r60936] Jason Parker * /, 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * formats/format_wav.c: modified default GAIN for issue 5823, thanks jrwalliker 2007-04-05 22:35 +0000 [r60323] Steve Murphy * 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 * 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 * main/http.c: Add the MIME type for gif by request from Pari 2007-04-05 12:55 +0000 [r60214] Joshua Colp * /, 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * /, 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 , reported via -dev list) ........ 2007-03-28 03:38 +0000 [r59281-59289] Tilghman Lesher * 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 * /, 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 * 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 * /, 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 * 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 * channels/chan_sip.c: Oops, this should be case insensitive 2007-03-26 21:41 +0000 [r59228] Steve Murphy * 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 * channels/chan_sip.c: Change this to a single dp function to make oej happy. 2007-03-26 20:06 +0000 [r59225] Steve Murphy * 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 * 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 * 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 * apps/app_speech_utils.c: Make SpeechBackground obey the digit timeout value. 2007-03-26 17:53 +0000 [r59207-59209] Russell Bryant * 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 * 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 * 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 * 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 * /, 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 * /: 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 * 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 * 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 * 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 * main/logger.c: Fix defines for inline stack backtraces (only used by developers anyway) 2007-03-20 20:42 +0000 [r59076] Joshua Colp * 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 * 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 <> (bug 9316) is here... 2007-03-20 13:16 +0000 [r59064] Christian Richter * 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 * Asterisk 1.4.2 released. 2007-03-19 22:29 +0000 [r59049] Tilghman Lesher * funcs/func_strings.c: Oops, this should have been a %d all along 2007-03-19 15:52 +0000 [r59042] Joshua Colp * 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 * configs/sip_notify.conf.sample: Fix unescaped semicolon (reported via -dev list) 2007-03-18 20:37 +0000 [r59037] Olle Johansson * 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 * 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 * 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 * 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 * 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 * doc/asterisk.tex: add a link to the rubber homepage 2007-03-15 23:11 +0000 [r58939] Tilghman Lesher * 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 * 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 * funcs/func_realtime.c: Function works fine, but the documentation is backwards. 2007-03-15 22:25 +0000 [r58931] Russell Bryant * 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 * 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 * 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 * 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 * 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 * /, 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 * /: 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 * 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 * 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 * 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 * 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 * 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 * res/res_odbc.c: Fix segfault (Issue 9236) 2007-03-08 20:54 +0000 [r58474] Russell Bryant * 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 * 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 * 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 * 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 * 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 * /: 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 * /, 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * main/say.c: submitted patch for Georgian language, issue 9010, submitted by Alexander Shaduri 2007-03-03 00:02 +0000 [r57591] Russell Bryant * configs/sla.conf.sample: add missing configuration template. Thanks to Lacy Moore on asterisk-users for pointing this out\! 2007-03-02 Russell Bryant * Asterisk 1.4.1 released. 2007-03-02 23:03 +0000 [r57556] Russell Bryant * 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 * /, 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 * 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 * 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 * 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 * doc/osp.txt: 1. Corrected a typo for www.etsi.org. Thank Patrick. 2007-02-28 23:01 +0000 [r57144-57207] Russell Bryant * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * build_tools/make_version_h: build special version strings for AADK/S800i builds 2007-02-23 17:58 +0000 [r56341] Russell Bryant * 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 * /: 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 * 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 * 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 * doc/osp.txt: Update OSP documentation for v1.4. 2007-02-22 10:33 +0000 [r56125] Olle Johansson * channels/chan_sip.c: Move message from verbose to debug 2007-02-22 02:39 +0000 [r56094] Steve Murphy * sounds/Makefile: updated the sound tarball versions in Makefile 2007-02-22 01:24 +0000 [r56011-56055] Russell Bryant * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * /, 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * /: Block patch only needed in 1.2 2007-02-14 16:56 +0000 [r54375] Matt Frederickson * 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 * 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 * 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 * /: Blocking patch for 1.2 only 2007-02-12 19:17 +0000 [r54066-54103] Russell Bryant * 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 * configure, configure.ac: Make the --without-oss argument work. (issue #9026 reported by puzzled) 2007-02-12 15:38 +0000 [r54002] Russell Bryant * configs/users.conf.sample: Fix a typo where "vmpassword" should be "vmsecret" 2007-02-10 09:09 +0000 [r53878-53881] Paul Cadach * 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 ' command back 2007-02-10 06:06 +0000 [r53850] Kevin P. Fleming * 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 * 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 * 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 * 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 * 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 * main/acl.c: Don't output debug unless we asked for it 2007-02-08 17:54 +0000 [r53601] Joshua Colp * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * channels/chan_sip.c: Formatting 2007-02-05 17:06 +0000 [r53150-53152] Joshua Colp * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /: 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 * /, 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 * 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 * /: Going backwards, blame file. 2007-02-01 21:11 +0000 [r53086-53088] Joshua Colp * /, 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 * 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 * 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 * 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 * 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 * 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 * configs/sip.conf.sample: Add explanation of port= in combination with defaultip= (thanks jsmith) 2007-02-01 13:17 +0000 [r53060] Christian Richter * channels/chan_misdn.c: we update the name on any first reply of our setup 2007-02-01 11:07 +0000 [r53057] Paul Cadach * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * pbx/pbx_config.c: Argument number correction 2007-01-29 21:36 +0000 [r52611-52647] Russell Bryant * 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 * apps/app_voicemail.c: Use ast_calloc instead of malloc. 2007-01-29 17:57 +0000 [r52535] Steve Murphy * 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 * 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 * 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 * 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 * 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 * /, 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 * /, 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 * 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 * 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 * 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 * configure: rebuild configure script to reflect last chan_h323 related changes. 2007-01-24 12:57 +0000 [r51979-51989] Christian Richter * 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 * 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 * acinclude.m4: Allow x64 builds of H.323 (please, rebuild configure) 2007-01-24 00:59 +0000 [r51829-51848] Russell Bryant * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /: 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 * pbx/pbx_ael.c: This fixes 8836, according to dnatural 2007-01-22 19:13 +0000 [r51360-51407] Joshua Colp * 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 * 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 * 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 * 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 * /: block this patch since it is already here 2007-01-18 22:50 +0000 [r51265] Jason Parker * 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 * 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 * /, 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 * 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 * 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 * 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 * main/channel.c: Fix an issue with deprecated commands 2007-01-18 17:49 +0000 [r51236] Tilghman Lesher * 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 * 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 * 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 * 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 * /, 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 * apps/app_voicemail.c: re-add "password" for realtime voicemail 2007-01-17 06:36 +0000 [r51182] Joshua Colp * 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 * 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 * channels/chan_iax2.c: Move rescheduling of lagrq/pings into the scheduler callback. 2007-01-17 00:20 +0000 [r51165-51170] Jason Parker * 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 * /, 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 * apps/app_voicemail.c: minor things i missed before i get jumped on 2007-01-16 17:39 +0000 [r51148] Joshua Colp * /, 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 * 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 * 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 * 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 * 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 * Makefile.rules: Filter out a few CFLAGS that are not valid CXXFLAGS. 2007-01-15 23:10 +0000 [r50988] Tilghman Lesher * /: 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 * 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 * main/asterisk.c: re-add deprecated "show version" CLI command. 2007-01-15 16:36 +0000 [r50895] Joshua Colp * 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 * 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 * main/astmm.c: Add missing newlines for two memory CLI commands. 2007-01-14 05:13 +0000 [r50782] Tilghman Lesher * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * /, 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 * 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 * configure, acinclude.m4: one more time... * configure, acinclude.m4: proper fix for r49712 * configure, acinclude.m4: if --with-foo= 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * channels/chan_zap.c: remove comment that is unrelated to this function 2007-01-02 12:08 +0000 [r49145] Olle Johansson * 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 * 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 * 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 * 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 * 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 * /: 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 * 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 * sounds/Makefile: new versions of sounds 2006-12-28 19:52 +0000 [r49024] Jason Parker * 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 * 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 * 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 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * Asterisk 1.4.0 released. 2006-12-22 22:33 +0000 [r48870-48906] Jason Parker * 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 * /, 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 * 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 * channels/Makefile: suppress compiler warnings in this module until it can be improved 2006-12-19 21:12 +0000 [r48585] Joshua Colp * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * Asterisk 1.4.0-beta4 released. 2006-12-12 04:13 +0000 [r48401] Joshua Colp * 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 * 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 * 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 * funcs/func_strings.c: STRFTIME() does not actually require an argument (issue 8540) 2006-12-11 05:36 +0000 [r48377-48381] Joshua Colp * 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 * 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 * 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 * 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 * /, 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 * 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 * /, channels/chan_sip.c: Issue #8258 - fix handling of 487 being retransmitted to Asterisk 2006-12-06 16:15 +0000 [r48323] Russell Bryant * 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 * /, channels/chan_sip.c: Don't send Contact on MESSAGE 2006-12-05 20:42 +0000 [r48279] Jason Parker * 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 * 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 * 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 * 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 * /: 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * main/cli.c: Double-unlock error (reported by blitzrage on IRC) 2006-12-01 17:41 +0000 [r48177] Olle Johansson * 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 * 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 * /, channels/chan_sip.c: Issue 8319 - change noncecount before using it. 2006-11-30 20:28 +0000 [r48143-48162] Joshua Colp * /: 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 * 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 * 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 * 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 * 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 * configs/sip.conf.sample: Clarify RTP timers. Sorry, grandma. 2006-11-29 04:26 +0000 [r48101] Joshua Colp * apps/app_voicemail.c: Don't crash if the mailstream was not created. 2006-11-28 18:26 +0000 [r48095] Jason Parker * Makefile: Export several more variables in top level Makefile. Inspired by issue 8438. 2006-11-28 16:57 +0000 [r48054-48088] Joshua Colp * 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 * /, 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 * main/manager.c: Remove a couple of unused variables (issue #8380, casper) 2006-11-27 15:32 +0000 [r48038] Joshua Colp * 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 * channels/chan_sip.c: Change logging message 2006-11-26 00:26 +0000 [r48015-48017] Steve Murphy * 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 * /, 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 * main/translate.c: bug 8189 posted this fix for main/translate.c for PLC 2006-11-24 15:46 +0000 [r47989] Christian Richter * 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 * /, channels/chan_sip.c: Don't allocate unused variable. 2006-11-22 21:47 +0000 [r47944] Joshua Colp * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * apps/app_voicemail.c: Fix ODBC_STORAGE for when context is NULL 2006-11-15 21:33 +0000 [r47707] Joshua Colp * 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 * main/file.c: don't try to call fclose() if fopen() failed 2006-11-15 20:31 +0000 [r47698] Olle Johansson * channels/chan_sip.c: - Improve SIP history - Never send reply to ACK (again...) 2006-11-15 20:31 +0000 [r47684-47697] Kevin P. Fleming * 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 * 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 * /, 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 * 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 * main/term.c: more formatting cleanup, and avoid running off the end of the string 2006-11-15 00:14 +0000 [r47639] Joshua Colp * 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 * 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 * 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 * 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 * 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 * 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 * /, 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 * /, 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 * 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 * main/cli.c, channels/chan_sip.c: Re-enable old deprecated commands 2006-11-13 19:10 +0000 [r47572] Olle Johansson * /, 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 * 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 * 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 * /, 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 * channels/chan_sip.c: Only produce error message about sip history once 2006-11-13 05:48 +0000 [r47527] Russell Bryant * 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 * 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 * 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 * /, 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 * main/term.c, /, main/logger.c, include/asterisk/term.h: woohoo safe output! 2006-11-10 22:23 +0000 [r47480] Matt Frederickson * 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 * 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 * 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 * /, channels/chan_sip.c: Issue 8010 - Fix support for multipart SDP (alphaque) 2006-11-10 17:13 +0000 [r47444] Luigi Rizzo * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 ". 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 * 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 * 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 * 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 * /, channels/chan_sip.c: Destroy dialog properly at unload (rizzo) 2006-11-07 23:46 +0000 [r47303] Steve Murphy * 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 * 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 * 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 * 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 * channels/chan_sip.c: Issue 8303 (lrizzo) - break instead of continue. 2006-11-07 13:13 +0000 [r47250] Olle Johansson * /, 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 * /, 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 * 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 * 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 * channels/chan_sip.c: This fix introduced via bug 8233 2006-11-03 17:53 +0000 [r47107-47108] Luigi Rizzo * 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 * 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 * 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 * /, 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 * channels/misdn/isdn_lib.c: find_free_chan_in_stack usage fix 2006-11-02 16:45 +0000 [r46937] Kevin P. Fleming * 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 * /, 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 * /, channels/chan_sip.c: Dont overwrite pkt->flags (from 1.2) 2006-11-02 14:02 +0000 [r46845-46883] Russell Bryant * /, 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 * channels/chan_gtalk.c: bind address support from bug 8164 2006-11-01 19:49 +0000 [r46802] Steve Murphy * res/res_config_odbc.c: a fix for bug 8251; the var_val needs to accept longer strings or mass confusion and a lot of lost time is the result 2006-11-01 18:39 +0000 [r46780] Joshua Colp * 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 * 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 * 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 * main/config.c: Cleanups suggested by Russell. 2006-11-01 16:39 +0000 [r46744] Russell Bryant * 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 * main/translate.c: Fix "core show translation" output. Issue #8243, patch by Damin. 2006-10-31 21:47 +0000 [r46711-46714] Kevin P. Fleming * 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 * /: 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 * 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 * 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 * 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 * 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 * 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 * utils/Makefile: Don't build muted on OpenBSD, it is not supported. 2006-10-27 19:03 +0000 [r46370] Russell Bryant * 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 * contrib/asterisk-ng-doxygen: Put some doxygen pressure on Christian :-) 2006-10-27 17:39 +0000 [r46358-46363] Russell Bryant * 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 * 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 * /, 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 * CHANGES: Corrections to changes (Multiparking is not included) 2006-10-26 16:31 +0000 [r46329] Russell Bryant * 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 * CHANGES: update to reflect G.722 addition 2006-10-26 04:18 +0000 [r46298] Russell Bryant * doc/backtrace.txt: update backtrace documentation to reflect changes in 1.4 (issue #8230, kshumard) 2006-10-26 01:37 +0000 [r46287] Mark Spencer * main/config.c, main/manager.c: Fix config comment code preservation code (thanks murf!) 2006-10-25 20:14 +0000 [r46276] Olle Johansson * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * res/res_agi.c: Fix FastAGI when there is no pid (bug #7628, #8147) 2006-10-24 19:29 +0000 [r46130] Joshua Colp * channels/chan_iax2.c: We need to initialize our scheduler pthread condition... yes. 2006-10-24 08:34 +0000 [r46114-46117] Luigi Rizzo * 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 * 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 * 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 * 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 * 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 * /, 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 * 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 * 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 * 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 * 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 * 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 * channels/chan_sip.c: Don't leak the actual thread-specific sip_pvt struct 2006-10-18 23:49 +0000 [r45621] Kevin P. Fleming * 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 * 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 * channels/chan_sip.c: remove duplicate declarations 2006-10-18 04:09 +0000 [r45464] Luigi Rizzo * 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 * 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 * 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 * Asterisk 1.4.0-beta3 released. 2006-10-17 22:31 +0000 [r45408-45410] Kevin P. Fleming * 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 * 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 * configs/sip.conf.sample: Adding information about Marks direct-RTP hack to the docs... 2006-10-17 17:22 +0000 [r45327] Kevin P. Fleming * LICENSE: provide licensing language for IAXy firmware file 2006-10-16 20:06 +0000 [r45246-45280] Joshua Colp * 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 * 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 * 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 * 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 * 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 * 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 * channels/chan_misdn.c: avoiding warning, fixing potential bug 2006-10-13 18:42 +0000 [r45051-45079] Joshua Colp * 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 * 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 * 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 * 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 * 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 * channels/chan_sip.c: merge formatting and minor code simplifications from trunk 2006-10-12 20:34 +0000 [r44982] Matt O'Gorman * channels/chan_gtalk.c: fix for bug 7764. 2006-10-12 19:14 +0000 [r44956-44971] Kevin P. Fleming * 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 * 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 * 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 * main/manager.c: append_event must be called while holding the session lock 2006-10-12 10:24 +0000 [r44911] Russell Bryant * 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 * 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 * channels/misdn/isdn_lib.c: removed warning because of missing prototype declaration 2006-10-10 19:23 +0000 [r44830] Olle Johansson * 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 * 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 * 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 * 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 * 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 * channels/chan_sip.c: more merge from trunk (comments and change a static function name) 2006-10-10 15:23 +0000 [r44788] Joshua Colp * channels/chan_sip.c: Only set DTMF information if an RTP structure exists 2006-10-10 13:50 +0000 [r44786] Christian Richter * 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 * 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 * 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 * 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 * channels/chan_sip.c: do not dereference p if we know it is NULL 2006-10-07 14:39 +0000 [r44684] Paul Cadach * 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 * 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 * 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 * 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 * CHANGES: various cleanups 2006-10-06 18:46 +0000 [r44581-44605] Joshua Colp * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * channels/chan_sip.c: fix Polycom presence notification again 2006-10-04 22:52 +0000 [r44407-44409] Luigi Rizzo * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * channels/h323/ast_h323.cxx: Fix preparation of type and presentation of calling number 2006-10-03 00:01 +0000 [r44240] Matt O'Gorman * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * main/rtp.c: Allow one-way RTP streams (device->Asterisk) 2006-09-30 16:28 +0000 [r44080] Luigi Rizzo * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * channels/h323/ast_h323.cxx: Extend call establishment timeout 2006-09-28 17:31 +0000 [r43913-43915] Joshua Colp * 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 * 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 * 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 * channels/h323/ast_h323.cxx: Fix compiler warning 2006-09-28 15:29 +0000 [r43864-43873] BJ Weschke * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * channels/chan_sip.c: Make rfc2833compensate a global option. 2006-09-27 04:35 +0000 [r43756] Russell Bryant * 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 * 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 * 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 * .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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * 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 * Asterisk 1.4.0-beta2 released. 2006-09-21 16:08 +0000 [r43404-43405] Kevin P. Fleming * main/Makefile: remove this change... it requires binutils 2.17 2006-09-20 23:19 +0000 [r43396] Jason Parker * build_tools/make_version: fix minor typo in the way version is handled 2006-09-20 Kevin P. Fleming * Asterisk 1.4.0-beta1 released.