asterisk

Commit Graph

Author	SHA1	Message	Date
Matthew Nicholson	95fac13256	Merged revisions 198072 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r198072 \| mnicholson \| 2009-05-29 14:04:24 -0500 (Fri, 29 May 2009) \| 21 lines Merged revisions 198068 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198068 \| mnicholson \| 2009-05-29 13:53:01 -0500 (Fri, 29 May 2009) \| 15 lines Use AST_CDR_NOANSWER instead of AST_CDR_NULL as the default CDR disposition. This change also involves the addition of an AST_CDR_FLAG_ORIGINATED flag that is used on originated channels to distinguish: them from dialed channels. (closes issue #12946) Reported by: meral Patches: null-cdr2.diff uploaded by mnicholson (license 96) Tested by: mnicholson, dbrooks (closes issue #15122) Reported by: sum Tested by: sum ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@198073 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	a66b938920	Merged revisions 197606 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r197606 \| mmichelson \| 2009-05-28 10:32:19 -0500 (Thu, 28 May 2009) \| 22 lines Recorded merge of revisions 197588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197588 \| mmichelson \| 2009-05-28 10:27:49 -0500 (Thu, 28 May 2009) \| 16 lines Allow for media to arrive from an alternate source when responding to a reinvite with 491. When we receive a SIP reinvite, it is possible that we may not be able to process the reinvite immediately since we have also sent a reinvite out ourselves. The problem is that whoever sent us the reinvite may have also sent a reinvite out to another party, and that reinvite may have succeeded. As a result, even though we are not going to accept the reinvite we just received, it is important for us to not have problems if we suddenly start receiving RTP from a new source. The fix for this is to grab the media source information from the SDP of the reinvite that we receive. This information is passed to the RTP layer so that it will know about the alternate source for media. Review: https://reviewboard.asterisk.org/r/252 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@197615 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	c7731d3489	Merged revisions 197543 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r197543 \| mmichelson \| 2009-05-28 09:58:06 -0500 (Thu, 28 May 2009) \| 27 lines Merged revisions 197537 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197537 \| mmichelson \| 2009-05-28 09:49:13 -0500 (Thu, 28 May 2009) \| 21 lines Add flags to chanspy audiohook so that audio stays in sync. There are two flags being added to the chanspy audiohook here. One is the pre-existing AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set, we ensure that the read and write slinfactories on the audiohook do not skew beyond a certain tolerance. In addition, there is a new audiohook flag added here, AST_AUDIOHOOK_SMALL_QUEUE. With this flag set, we do not allow for a slinfactory to build up a substantial amount of audio before flushing it. For this particular issue, this means that the person spying on the call will hear the conversations in real time with very little delay in the audio. (closes issue #13745) Reported by: geoffs Patches: 13745.patch uploaded by mmichelson (license 60) Tested by: snblitz ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@197544 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	76b9a6b5d8	Fix broken attended transfers The bridge was terminating immediately after the attended transfer was completed. The problem was because upon reentering ast_channel_bridge nexteventts was checked to see if it was set and if so could possibly return AST_BRIDGE_COMPLETE. (closes issue #15183) Reported by: andrebarbosa Tested by: andrebarbosa, tootai, loloski git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@197126 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	e5d7298863	Merged revisions 195882 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r195882 \| mnicholson \| 2009-05-21 10:33:55 -0500 (Thu, 21 May 2009) \| 20 lines Merged revisions 195881 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195881 \| mnicholson \| 2009-05-21 10:25:50 -0500 (Thu, 21 May 2009) \| 13 lines This commit prevents cdr records with AST_CDR_FLAG_ANSLOCKED and AST_CDR_FLAG_LOCKED from being updated in certain cases. This is accomplished by adding two functions to update the answer time and disposition of calls that checks for the proper lock flags. These functions are used in the ast_bridge_call() function so that ForkCDR(A) calls are respected. This patch also modifies the way ast_bridge_call() chooses the cdr record to base the bridged_cdr on. Previously the first unlocked cdr record would be chosen, now instead the first cdr record is chosen and forked cdr records are moved to the bridge_cdr. This allows the original cdr record and any forked cdr records to be properly updated with answer and end times. (closes issue #13797) Reported by: sh0t Tested by: sh0t (closes issue #14744) Reported by: deepesh ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@195894 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	4892be94a6	Recorded merge of revisions 195370 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r195370 \| tilghman \| 2009-05-18 15:52:33 -0500 (Mon, 18 May 2009) \| 15 lines Recorded merge of revisions 195366 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195366 \| tilghman \| 2009-05-18 15:24:13 -0500 (Mon, 18 May 2009) \| 8 lines Add a similar dependency on SMDI for voicemail as already exists for ADSI. (closes issue #14846) Reported by: pj Patches: 20090413__bug14846__1.4.diff.txt uploaded by tilghman (license 14) 20090507__issue14846__1.6.0.diff.txt uploaded by tilghman (license 14) 20090507__issue14846__1.6.1.diff.txt uploaded by tilghman (license 14) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@195371 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	3f94c3bd6b	Merged revisions 192357 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192357 \| kpfleming \| 2009-05-05 15:18:21 +0200 (Tue, 05 May 2009) \| 5 lines Correct some flaws in the memory accounting code for stringfields and ao2 objects Under some conditions, the memory allocation for stringfields and ao2 objects would not have supplied valid file/function names for MALLOC_DEBUG tracking, so this commit corrects that. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@192358 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	10a2c0099e	Merged revisions 192318 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192318 \| kpfleming \| 2009-05-05 12:34:19 +0200 (Tue, 05 May 2009) \| 5 lines Properly account for memory allocated for channels and datastores As in previous commits, when channels are allocated (with ast_channel_alloc) or datastores are allocated (with ast_datastore_alloc) properly account for the memory being owned by the caller, instead of the allocator function itself. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@192353 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	60280df797	Merged revisions 192279 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r192279 \| kpfleming \| 2009-05-05 10:51:06 +0200 (Tue, 05 May 2009) \| 5 lines Ensure that string pools allocated to hold stringfields are properly accounted in MALLOC_DEBUG mode This commit modifies the stringfield pool allocator to remember the 'owner' of the stringfield manager the pool is being allocated for, and ensures that pools allocated in the future when fields are populated are owned by that file/function. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@192280 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	5e04e9d398	Merged revisions 191367 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r191367 \| tilghman \| 2009-04-30 12:40:58 -0500 (Thu, 30 Apr 2009) \| 3 lines Detect eaccess (or euidaccess) before using it. Reported by Andrew Lindh via the -dev list. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@191368 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	53843dfb82	Merged revisions 190725 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r190725 \| kpfleming \| 2009-04-27 14:30:54 -0500 (Mon, 27 Apr 2009) \| 13 lines Merged revisions 190721 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r190721 \| kpfleming \| 2009-04-27 14:29:46 -0500 (Mon, 27 Apr 2009) \| 7 lines Fix 'inconsistent line endings' when autoconf 2.63 is used Attempt to make configure script regeneration 'safe' using autoconf 2.63, which embeds a bare CR into the script, thus making Subversion complain about inconsistent line endings This commit changes the MIME type of the configure script to be 'binary' thus making Subversion no longer inspect line endings, and as a bonus 'svn diff' will no longer try to generate diff output for it, which is not generally useful anyway. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@190730 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	08c9634115	Merge of timedwrlock detection broke build. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@190419 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	94b71d104f	Merged revisions 190093 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r190093 \| tilghman \| 2009-04-22 16:38:15 -0500 (Wed, 22 Apr 2009) \| 14 lines Merged revisions 190092 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r190092 \| tilghman \| 2009-04-22 16:35:03 -0500 (Wed, 22 Apr 2009) \| 7 lines Detect availability of pthread_rwlock_timedwrlock() before using it. (closes issue #14930) Reported by: tilghman Patches: 20090420__bug14930.diff.txt uploaded by tilghman (license 14) Tested by: mvanbaak, tilghman ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@190094 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jeff Peeler	62f5546319	Merged revisions 190057 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r190057 \| jpeeler \| 2009-04-22 16:15:55 -0500 (Wed, 22 Apr 2009) \| 9 lines Fix building of chan_h323 with gcc-3.3 There seems to be a bug with old versions of g++ that doesn't allow a structure member to use the name list. Rename list member to group_list in ast_group_info and change the few places it is used. (closes issue #14790) Reported by: stuarth ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@190059 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Doug Bailey	a0c3c3dad9	Merged revisions 189629 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r189629 \| dbailey \| 2009-04-21 09:28:04 -0500 (Tue, 21 Apr 2009) \| 10 lines Merged revisions 189601 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r189601 \| dbailey \| 2009-04-21 09:00:55 -0500 (Tue, 21 Apr 2009) \| 3 lines Add check in configure script to check for GLOB_NOMAGIC and GLOB_BRACE in glob.h This allows config.c to compile when linked against uclibc that does not support these parameters ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@189630 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	3d7bac20f6	typo git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@187490 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	9a69733bc2	Merged revisions 187483 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r187483 \| tilghman \| 2009-04-09 13:40:01 -0500 (Thu, 09 Apr 2009) \| 15 lines Merged revisions 187428 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187428 \| tilghman \| 2009-04-09 13:08:20 -0500 (Thu, 09 Apr 2009) \| 8 lines Race condition between ast_cli_command() and 'module unload' could cause a deadlock. Add lock timeouts to avoid this potential deadlock. (closes issue #14705) Reported by: jamessan Patches: 20090320__bug14705.diff.txt uploaded by tilghman (license 14) Tested by: jamessan ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@187485 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	ec49f90f69	Merged revisions 187302 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r187302 \| tilghman \| 2009-04-08 23:59:05 -0500 (Wed, 08 Apr 2009) \| 14 lines Merged revisions 187300-187301 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187300 \| tilghman \| 2009-04-08 23:31:38 -0500 (Wed, 08 Apr 2009) \| 3 lines Add debugging mode for diagnosing file descriptor leaks. (Related to issue #14625) ........ r187301 \| tilghman \| 2009-04-08 23:32:40 -0500 (Wed, 08 Apr 2009) \| 2 lines Oops, missed this file in the last commit. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@187304 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	01dc8d8ea9	Merged revisions 186321 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186321 \| file \| 2009-04-03 12:52:50 -0300 (Fri, 03 Apr 2009) \| 12 lines Merged revisions 186320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186320 \| file \| 2009-04-03 12:48:56 -0300 (Fri, 03 Apr 2009) \| 5 lines Fix a problem with the crypto variable definitions not actually being defined properly. (closes issue #14804) Reported by: jvandal ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@186322 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	ce37a27180	Merged revisions 184147 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184147 \| russell \| 2009-03-24 20:42:10 -0500 (Tue, 24 Mar 2009) \| 5 lines Fix build issues on Mac OSX. (closes issue #14714) Reported by: ygor ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@184148 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
David Vossel	90b97eddc4	Merged revisions 183436 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183436 \| dvossel \| 2009-03-19 15:30:39 -0500 (Thu, 19 Mar 2009) \| 13 lines Merged revisions 183386 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183386 \| dvossel \| 2009-03-19 14:40:07 -0500 (Thu, 19 Mar 2009) \| 6 lines Cleaning up a few things in detect disconnect patch Initialized ast_call_feature in detect_disconnect to avoid accessing uninitialized memory. Cleaned up /param tags in features.h. No longer send dynamic features in ast_feature_detect. issue #11583 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183437 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	70eaf71dfe	Merged revisions 183242 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183242 \| russell \| 2009-03-19 13:00:15 -0500 (Thu, 19 Mar 2009) \| 10 lines Merged revisions 183241 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183241 \| russell \| 2009-03-19 12:52:52 -0500 (Thu, 19 Mar 2009) \| 2 lines Remove the use of RTLD_NOLOAD, as it is not behaving like expected. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183243 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
David Vossel	9628d62ffa	Merged revisions 183172 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183172 \| dvossel \| 2009-03-19 11:28:33 -0500 (Thu, 19 Mar 2009) \| 20 lines Merged revisions 183126 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183126 \| dvossel \| 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009) \| 17 lines Allow disconnect feature before a call is bridged feature.conf has a disconnect option. By default this option is set to '', but it could be anything. If a user wishes to disconnect a call before the other side answers, only '' will work, regardless if the disconnect option is set to something else. This is because features are unavailable until bridging takes place. The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different. This patch allows features to be detected from outside of the bridge, but not operated on. In this case, the disconnect feature can be detected before briding and handled outside of features.c. (closes issue #11583) Reported by: sobomax Patches: patch-apps__app_dial.c uploaded by sobomax (license 359) 11583.latest-patch uploaded by murf (license 17) detect_disconnect.diff uploaded by dvossel (license 671) Tested by: sobomax, dvossel Review: http://reviewboard.digium.com/r/195/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183199 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	e047ec4d72	Merged revisions 182847 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r182847 \| russell \| 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009) \| 52 lines Merged revisions 182810 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182810 \| russell \| 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) \| 44 lines Fix cases where the internal poll() was not being used when it needed to be. We have seen a number of problems caused by poll() not working properly on Mac OSX. If you search around, you'll find a number of references to using select() instead of poll() to work around these issues. In Asterisk, we've had poll.c which implements poll() using select() internally. However, we were still getting reports of problems. vadim investigated a bit and realized that at least on his system, even though we were compiling in poll.o, the system poll() was still being used. So, the primary purpose of this patch is to ensure that we're using the internal poll() when we want it to be used. The changes are: 1) Remove logic for when internal poll should be used from the Makefile. Instead, put it in the configure script. The logic in the configure script is the same as it was in the Makefile. Ideally, we would have a functionality test for the problem, but that's not actually possible, since we would have to be able to run an application on the _target_ system to test poll() behavior. 2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll() throughout the source tree to ast_poll(). I feel that it is good practice to give the API call a new name when we are changing its behavior and not using the system version directly in all cases. So, normally, ast_poll() is just redefined to poll(). On systems where AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll(). 4) Change poll() in main/poll.c to be ast_internal_poll(). It's worth noting that any code that still uses poll() directly will work fine (if they worked fine before). So, for example, out of tree modules that are using poll() will not stop working or anything. However, for modules to work properly on Mac OSX, ast_poll() needs to be used. (closes issue #13404) Reported by: agalbraith Tested by: russell, vadim http://reviewboard.digium.com/r/198/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182945 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	c0219aa890	Merged revisions 182525 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182525 \| kpfleming \| 2009-03-17 09:38:11 -0500 (Tue, 17 Mar 2009) \| 11 lines Improve behavior of ast_answer() to not lose incoming frames ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations. When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames. This patch rectifies this situation, by storing all incoming frames during the delay period on a list, and then requeuing them onto the channel before returning to the caller. http://reviewboard.digium.com/r/196/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182526 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jeff Peeler	368b57494b	Merged revisions 181135 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r181135 \| jpeeler \| 2009-03-10 23:06:44 -0500 (Tue, 10 Mar 2009) \| 20 lines Fix malloc debug macros to work properly with h323. The main problem here was that cstdlib was undefining free thereby causing the proper debug macros to not be used. ast_h323.cxx has been changed to call ast_free instead to avoid the issue. A few other issues were addressed: - There were a few instances of functions improperly passing ast_free instead of ast_free_ptr. - Some clean up was done to avoid the debug macros intentionally being redefined. (copied below from Kevin's commit, appreciate the help) - disable astmm.h from doing anything when STANDALONE is defined, which is used by the tools in the utils/ directory that use parts of Asterisk header files in hackish ways; also ensure that utils/extconf.c and utils/conf2ael.c are compiled with STANDALONE defined. (closes issue #13593) Reported by: pj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181137 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	2a877c8fcb	Merged revisions 180373 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r180373 \| kpfleming \| 2009-03-05 12:29:38 -0600 (Thu, 05 Mar 2009) \| 15 lines Merged revisions 180372 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180372 \| kpfleming \| 2009-03-05 12:22:16 -0600 (Thu, 05 Mar 2009) \| 9 lines Fix problems when RTP packet frame size is changed During some code analysis, I found that calling ast_rtp_codec_setpref() on an ast_rtp session does not work as expected; it does not adjust the smoother that may on the RTP session, in fact it summarily drops it, even if it has data in it, even if the current format's framing size has not changed. This is not good. This patch changes this behavior, so that if the packetization size for the current format changes, any existing smoother is safely updated to use the new size, and if no smoother was present, one is created. A new API call for smoothers, ast_smoother_reconfigure(), was required to implement these changes. Review: http://reviewboard.digium.com/r/184/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@180377 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
David Vossel	9cad0b7e22	Merged revisions 180032 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r180032 \| dvossel \| 2009-03-03 17:21:18 -0600 (Tue, 03 Mar 2009) \| 14 lines app_read does not break from prompt loop with user terminated empty string In app.c, ast_app_getdata is called to stream the prompts and receive DTMF input. If ast_app_getdata() receives an empty string caused by the user inputing the end of string character, in this case '#', it should break from the prompt loop and return to app_read, but instead it cycles through all the prompts. I've added a return value for this special case in ast_readstring() which uses an enum I've delcared in apps.h. This enum is now used as a return value for ast_app_getdata(). (closes issue #14279) Reported by: Marquis Patches: fix_app_read.patch uploaded by Marquis (license 32) read-ampersanmd.patch2 uploaded by dvossel (license 671) Tested by: Marquis, dvossel Review: http://reviewboard.digium.com/r/177/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@180078 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	aac9d07849	Merged revisions 177732 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r177732 \| tilghman \| 2009-02-20 15:25:37 -0600 (Fri, 20 Feb 2009) \| 10 lines Merged revisions 177701 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177701 \| tilghman \| 2009-02-20 15:15:01 -0600 (Fri, 20 Feb 2009) \| 3 lines This exception does not appear to still be true for Solaris 10, and OpenSolaris definitely needs it to be removed. Fixed for snuff-home on -dev channel. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177756 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
David Vossel	e938398385	Fixes issue with undefined audio codecs in chan_iax2 During iax2 call negotiation, supported codecs are passed in an Information Element containing a 2 byte field where each bit correlates to a specific codec. In 1.6 only audio codec bits 0-12 and 15 are defined, leaving bits 13-14 undefined. By default all bits are enabled unless specified otherwise. Since its a 2 byte field and 13-14 are not defined, these bits are never turned off. In trunk, bits 13-14 are defined, which means 1.6 is advertising support for codecs it does not have when talking to trunk. I fixed this by adding #define for undefined audio codec bits. These bits are then removed from iax2's full bandwidth capabilities. (closes issue #14283) Reported by: jcovert git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177698 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	36f41cee56	Merged revisions 177664 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177664 \| tilghman \| 2009-02-20 11:29:51 -0600 (Fri, 20 Feb 2009) \| 8 lines Allow semicolons to be escaped, when passing arguments to the System command. (closes issue #14231) Reported by: jcovert Patches: 20090113__bug14231__2.diff.txt uploaded by Corydon76 (license 14) corrected_20090113__bug14231__2.diff.txt uploaded by jcovert (license 551) Tested by: jcovert ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177665 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jeff Peeler	fff4e4f7cd	Merged revisions 177387 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r177387 \| jpeeler \| 2009-02-19 10:45:02 -0600 (Thu, 19 Feb 2009) \| 3 lines Fix another merge error from 176708 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177388 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	fece998d9c	Merged revisions 177098 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r177098 \| tilghman \| 2009-02-18 13:05:15 -0600 (Wed, 18 Feb 2009) \| 9 lines Merged revisions 177096 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177096 \| tilghman \| 2009-02-18 12:30:38 -0600 (Wed, 18 Feb 2009) \| 2 lines Document the return value of the update method (as requested on -dev list) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177099 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jeff Peeler	46db811169	Merged revisions 176708 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r176708 \| jpeeler \| 2009-02-17 16:08:00 -0600 (Tue, 17 Feb 2009) \| 23 lines Merged revisions 176701 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176701 \| jpeeler \| 2009-02-17 15:54:34 -0600 (Tue, 17 Feb 2009) \| 17 lines Modify bridging to properly evaluate DTMF after first warning is played The main problem is currently if the Dial flag L is used with a warning sound, DTMF is not evaluated after the first warning sound. To fix this, a flag has been added in ast_generic_bridge for playing the warning which ensures that if a scheduled warning is missed, multiple warrnings are not played back (due to a feature evaluation or waiting for digits). ast_channel_bridge was modified to store the nexteventts in the ast_bridge_config structure as that information was lost every time ast_channel_bridge was reentered, causing a hangup due to incorrect time calculations. (closes issue #14315) Reported by: tim_ringenbach Reviewed on reviewboard: http://reviewboard.digium.com/r/163/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176710 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	ddee1048c7	Merged revisions 176697 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r176697 \| mmichelson \| 2009-02-17 15:40:09 -0600 (Tue, 17 Feb 2009) \| 3 lines Clear up documentation of AST_FRIENDLY_OFFSET in frame.h ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176698 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	f3ae0e00a7	Merged revisions 176255 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r176255 \| kpfleming \| 2009-02-16 15:45:54 -0600 (Mon, 16 Feb 2009) \| 13 lines Merged revisions 176216 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176216 \| kpfleming \| 2009-02-16 15:10:38 -0600 (Mon, 16 Feb 2009) \| 3 lines fix a flaw in the ast_string_field_build() family of API calls; these functions made no attempt to reuse the space already allocated to a field, so every time the field was written it would allocate new space, leading to what appeared to be a memory leak. ........ r176254 \| kpfleming \| 2009-02-16 15:41:46 -0600 (Mon, 16 Feb 2009) \| 3 lines correct a logic error in the last stringfields commit... don't mark additional space as allocated if the string was built using already-allocated space ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176258 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Michiel van Baak	25c01347d8	Merged revisions 175952 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r175952 \| mvanbaak \| 2009-02-16 01:26:59 +0100 (Mon, 16 Feb 2009) \| 10 lines Merged revisions 175921 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175921 \| mvanbaak \| 2009-02-16 00:37:03 +0100 (Mon, 16 Feb 2009) \| 3 lines fix mis-spelling of the word registered. Reported by De_Mon on #asterisk-dev. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176022 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	90ef4eb33e	Merged revisions 175121 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175121 \| mmichelson \| 2009-02-12 10:28:06 -0600 (Thu, 12 Feb 2009) \| 11 lines Make lock information for ao2_trylock be more useful and gnarly Core show locks information involving an ao2_trylock did not show the function that called ao2_trylock, but would instead show ao2_trylock as the source of the lock. This is not useful when trying to debug locking issues. One bizarre note is that this logic is already in 1.4 but somehow did not get merged to trunk or the 1.6.X branches. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@175122 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	a45ec0c30a	Merged revisions 174945 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r174945 \| mmichelson \| 2009-02-11 16:41:01 -0600 (Wed, 11 Feb 2009) \| 29 lines Fix 'd' option for app_dial and add new option to Answer application The 'd' option would not work for channel types which use RTP to transport DTMF digits. The only way to allow for this to work was to answer the channel if we saw that this option was enabled. I realized that this may cause issues with CDRs, specifically with giving false dispositions and answer times. I therefore modified ast_answer to take another parameter which would tell if the CDR should be marked answered. I also extended this to the Answer application so that the channel may be answered but not CDRified if desired. I also modified app_dictate and app_waitforsilence to only answer the channel if it is not already up, to help not allow for faulty CDR answer times. All of these changes are going into Asterisk trunk. For 1.6.0 and 1.6.1, however, all the changes except for the change to the Answer application will go in since we do not introduce new features into stable branches (closes issue #14164) Reported by: DennisD Patches: 14164.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/145 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@174946 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Terry Wilson	af2b34cb56	Merged revisions 172580 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r172580 \| twilson \| 2009-01-30 15:29:12 -0600 (Fri, 30 Jan 2009) \| 44 lines Merged revisions 172517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172517 \| twilson \| 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) \| 37 lines Fix feature inheritance with builtin features When using builtin features like parking and transfers, the AST_FEATURE_* flags would not be set correctly for all instances when either performing a builtin attended transfer, or parking a call and getting the timeout callback. Also, there was no way on a per-call basis to specify what features someone should have on picking up a parked call (since that doesn't involve the Dial() command). There was a global option for setting whether or not all users who pickup a parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT, AUTOMON, or PARKCALL. This patch: 1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the dialplan or with setvar in channels that support it. This variable can be set to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the equivalent dial options), to set what features should be activated on this channel. The patch moves the setting of the features datastores into the bridging code instead of app_dial to help facilitate this. 2) adds global options parkedcallparking, parkedcallhangup, and parkedcallrecording to be similar to the parkedcalltransfers option for globally setting features. 3) has builtin_atxfer call builtin_parkcall if being transfered to the parking extension since tracking everything through multiple masquerades, etc. is difficult and error-prone 4) attempts to fix all cases of return calls from parking and completed builtin transfers not having the correct permissions (closes issue #14274) Reported by: aragon Patches: fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396) Tested by: aragon, otherwiseguy Review http://reviewboard.digium.com/r/138/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@172635 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	07adec4209	Merged revisions 172598 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r172598 \| mmichelson \| 2009-01-30 16:22:04 -0600 (Fri, 30 Jan 2009) \| 3 lines Fix redefinition of flag in channel.h ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@172604 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	491c4a9c68	Merged revisions 172063 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r172063 \| murf \| 2009-01-28 13:31:06 -0700 (Wed, 28 Jan 2009) \| 52 lines Merged revisions 172030 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172030 \| murf \| 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) \| 46 lines This patch fixes h-exten running misbehavior in manager-redirected situations. What it does: 1. A new Flag value is defined in include/asterisk/channel.h, AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the bridge hangup exten code not to run the h-exten there (nor publish the bridge cdr there). It will done at the pbx-loop level instead. 2. In the manager Redirect code, I set this flag on the channel if the channel has a non-null pbx pointer. I did the same for the second (chan2) channel, which gets run if name2 is set... and the first succeeds. 3. I restored the ending of the cdr for the pbx loop h-exten running code. Don't know why it was removed in the first place. 4. The first attempt at the fix for this bug was to place code directly in the async_goto routine, which was called from a large number of places, and could affect a large number of cases, so I tested that fix against a fair number of transfer scenarios, both with and without the patch. In the process, I saw that putting the fix in async_goto seemed not to affect any of the blind or attended scenarios, but still, I was was highly concerned that some other scenarios I had not tested might be negatively impacted, so I refined the patch to its current scope, and jmls tested both. In the process, tho, I saw that blind xfers in one situation, when the one-touch blind-xfer feature is used by the peer, we got strange h-exten behavior. So, I inserted code to swap CDRs and to set the HANGUP_DONT field, to get uniform behavior. 5. I added code to the bridge to obey the HANGUP_DONT flag, skipping both publishing the bridge CDR, and running the h-exten; they will be done at the pbx-loop (higher) level instead. 6. I removed all the debug logs from the patch before committing. 7. I moved the AUTOLOOP set/reset in the h-exten code in res_features so it's only done if the h-exten is going to be run. A very minor performance improvement, but technically correct. (closes issue #14241) Reported by: jmls Patches: 14241_redirect_no_bridgeCDR_or_h_exten_via_transfer uploaded by murf (license 17) Tested by: murf, jmls ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@172065 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	d6ff97e30d	Merged revisions 170943 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r170943 \| russell \| 2009-01-24 20:49:30 -0600 (Sat, 24 Jan 2009) \| 6 lines Change ARRAY_LEN() to be more C++ safe. When the second part of this macro is written as 0[a] instead of a[0], it will force a failure if the macro is used on a C++ object that overloads the [] operator. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@170944 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	4deaa3d67f	Merged revisions 169944 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r169944 \| tilghman \| 2009-01-21 18:44:52 -0600 (Wed, 21 Jan 2009) \| 16 lines Merged revisions 169943 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r169943 \| tilghman \| 2009-01-21 18:43:31 -0600 (Wed, 21 Jan 2009) \| 9 lines AST_RWLOCK_INIT_VALUE is always defined. What we really wanted to ask is whether autoconf detected a static initializer value. This fixes rwlocks on all such platforms (mainly, Mac OS X). (closes issue #13767) Reported by: jcovert Patches: 20090121__bug13767.diff.txt uploaded by Corydon76 (license 14) Tested by: jcovert, Corydon76 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@169945 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	b669fac978	Merged revisions 168832 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168832 \| tilghman \| 2009-01-16 12:49:09 -0600 (Fri, 16 Jan 2009) \| 13 lines Merged revisions 168828 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168828 \| tilghman \| 2009-01-16 12:41:35 -0600 (Fri, 16 Jan 2009) \| 6 lines Fix the conjugation of Russian and Ukrainian languages. (related to issue #12475) Reported by: chappell Patches: vm_multilang.patch uploaded by chappell (license 8) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@168835 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	1b1c2db6bd	Merged revisions 168562 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r168562 \| russell \| 2009-01-13 13:22:13 -0600 (Tue, 13 Jan 2009) \| 10 lines Merged revisions 168561 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168561 \| russell \| 2009-01-13 13:13:05 -0600 (Tue, 13 Jan 2009) \| 2 lines Revert unnecessary indications API change from rev 122314 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@168564 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	5d34e0df03	Merged revisions 166665 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk Due to non-symmetrical updating, I had some fairly interesting conflicts to straighten out in this release. The changes were such that I was compelled to run thru all the same tests as trunk, which turned up some problems, which I fixed. ................ r166665 \| murf \| 2008-12-23 11:13:49 -0700 (Tue, 23 Dec 2008) \| 153 lines Merged revisions 166093 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 In order to merge this 1.4 patch into trunk, I had to resolve some conflicts and wait for Russell to make some changes to res_agi. I re-ran all the tests; 39 calls in all, and made fairly careful notes and comparisons: I don't want this to blow up some aspect of asterisk; I completely removed the KEEPALIVE from the pbx.h decls. The first 3 scenarios involving feature park; feature xfer to 700; hookflash park to Park() app call all behave the same, don't appear to leave hung channels, and no crashes. ........ r166093 \| murf \| 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) \| 131 lines This merges the masqpark branch into 1.4 These changes eliminate the need for (and use of) the KEEPALIVE return code in res_features.c; There are other places that use this result code for similar purposes at a higher level, these appear to be left alone in 1.4, but attacked in trunk. The reason these changes are being made in 1.4, is that parking ends a channel's life, in some situations, and the code in the bridge (and some other places), was not checking the result code properly, and dereferencing the channel pointer, which could lead to memory corruption and crashes. Calling the masq_park function eliminates this danger in higher levels. A series of previous commits have replaced some parking calls with masq_park, but this patch puts them ALL to rest, (except one, purposely left alone because a masquerade is done anyway), and gets rid of the code that tests the KEEPALIVE result, and the NOHANGUP_PEER result codes. While bug 13820 inspired this work, this patch does not solve all the problems mentioned there. I have tested this patch (again) to make sure I have not introduced regressions. Crashes that occurred when a parked party hung up while the parking party was listening to the numbers of the parking stall being assigned, is eliminated. These are the cases where parking code may be activated: 1. Feature one touch (eg. *3) 2. Feature blind xfer to parking lot (eg ##700) 3. Run Park() app from dialplan (eg sip xfer to 700) (eg. dahdi hookflash xfer to 700) 4. Run Park via manager. The interesting testing cases for parking are: I. A calls B, A parks B a. B hangs up while A is getting the numbers announced. b. B hangs up after A gets the announcement, but before the parking time expires c. B waits, time expires, A is redialed, A answers, B and A are connected, after which, B hangs up. d. C picks up B while still in parking lot. II. A calls B, B parks A a. A hangs up while B is getting the numbers announced. b. A hangs up after B gets the announcement, but before the parking time expires c. A waits, time expires, B is redialed, B answers, A and B are connected, after which, A hangs up. d. C picks up A while still in parking lot. Testing this throroughly involves acting all the permutations of I and II, in situations 1,2,3, and 4. Since I added a few more changes (ALL references to KEEPALIVE in the bridge code eliimated (I missed one earlier), I retested most of the above cases, and no crashes. H-extension weirdness. Current h-extension execution is not completely correct for several of the cases. For the case where A calls B, and A parks B, the 'h' exten is run on A's channel as soon as the park is accomplished. This is expected behavior. But when A calls B, and B parks A, this will be current behavior: After B parks A, B is hung up by the system, and the 'h' (hangup) exten gets run, but the channel mentioned will be a derivative of A's... Thus, if A is DAHDI/1, and B is DAHDI/2, the h-extension will be run on channel Parked/DAHDI/1-1<ZOMBIE>, and the start/answer/end info will be those relating to Channel A. And, in the case where A is reconnected to B after the park time expires, when both parties hang up after the joyful reunion, no h-exten will be run at all. In the case where C picks up A from the parking lot, when either A or C hang up, the h-exten will be run for the C channel. CDR's are a separate issue, and not addressed here. As to WHY this strange behavior occurs, the answer lies in the procedure followed to accomplish handing over the channel to the parking manager thread. This procedure is called masquerading. In the process, a duplicate copy of the channel is created, and most of the active data is given to the new copy. The original channel gets its name changed to XXX<ZOMBIE> and keeps the PBX information for the sake of the original thread (preserving its role as a call originator, if it had this role to begin with), while the new channel is without this info and becomes a call target (a "peer"). In this case, the parking lot manager thread is handed the new (masqueraded) channel. It will not run an h-exten on the channel if it hangs up while in the parking lot. The h exten will be run on the original channel instead, in the original thread, after the bridge completes. See bug 13820 for our intentions as to how to clean up the h exten behavior. Review: http://reviewboard.digium.com/r/29/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@166729 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	6964b59097	Merged revisions 166696 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r166696 \| tilghman \| 2008-12-23 14:47:08 -0600 (Tue, 23 Dec 2008) \| 7 lines Allow semicolons and extended characters in user-specified SIP headers. (closes issue #14110) Reported by: gork Patches: 20081222__bug14110__2.diff.txt uploaded by Corydon76 (license 14) Tested by: gork, putnopvut ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@166697 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	70416419ce	Merged revisions 166282 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r166282 \| russell \| 2008-12-22 11:09:36 -0600 (Mon, 22 Dec 2008) \| 12 lines Introduce ast_careful_fwrite() and use in AMI to prevent partial writes. This patch introduces a function to do careful writes on a file stream which will handle timeouts and partial writes. It is currently used in AMI to address the issue that has been reported. However, there are probably a few other places where this could be used. (closes issue #13546) Reported by: srt Tested by: russell http://reviewboard.digium.com/r/104/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@166283 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	2e8b3deab5	When merging the fix for issue #14118 , I found that the issue didn't affect 1.6.0, but in this case that's not an especially good thing, because it means that the fix for issue #13496 was not merged into 1.6.0 in the first place. This commit kills two birds with one stone by putting both fixes in the 1.6.0 branch git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@166278 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	eec3edde9f	Merged revisions 166092,166095 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r166092 \| mmichelson \| 2008-12-19 16:26:16 -0600 (Fri, 19 Dec 2008) \| 28 lines Adding a new dialplan function AUDIOHOOK_INHERIT This function is being added as a method to allow for an audiohook to move to a new channel during a channel masquerade. The most obvious use for such a facility is for MixMonitor when a transfer is performed. Prior to the addition of this functionality, if a channel running MixMonitor was transferred by another party, then the recording would stop once the transfer had completed. By using AUDIOHOOK_INHERIT, you can make MixMonitor continue recording the call even after the transfer has completed. It has also been determined that since this is seen by most as a bug fix and is not an invasive change, this functionality will also be backported to 1.4 and merged into the 1.6.0 branches, even though they are feature-frozen. (closes issue #13538) Reported by: mbit Patches: 13538.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/102/ ........ r166095 \| mmichelson \| 2008-12-19 16:40:57 -0600 (Fri, 19 Dec 2008) \| 5 lines Remove the verbatim tag from the author line I could have sworn I already did that before, though... ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@166097 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	ca1c37e47c	Merged revisions 165723 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r165723 \| russell \| 2008-12-18 13:33:42 -0600 (Thu, 18 Dec 2008) \| 14 lines Remove the need for AST_PBX_KEEPALIVE with the GoSub option from Dial. This is part of an effort to completely remove AST_PBX_KEEPALIVE and other similar return codes from the source. While this usage was perfectly safe, there are others that are problematic. Since we know ahead of time that we do not want to PBX to destroy the channel, the PBX API has been changed so that information can be provided as an argument, instead, thus removing the need for the KEEPALIVE return value. Further changes to get rid of KEEPALIVE and related code is being done by murf. There is a patch up for that on review 29. Review: http://reviewboard.digium.com/r/98/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@165726 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	f83308aa28	Merged revisions 164737 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r164737 \| russell \| 2008-12-16 11:14:01 -0600 (Tue, 16 Dec 2008) \| 22 lines Merged revisions 164736 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164736 \| russell \| 2008-12-16 11:06:29 -0600 (Tue, 16 Dec 2008) \| 14 lines Fix memory leak and invalid reporting issues with DEBUG_THREADLOCALS. One issue was that the ast_mutex_* API was being used within the context of the thread local data destructors. We would go off and allocate more thread local data while the pthread lib was in the middle of destroying it all. This led to a memory leak. Another issue was an invalid argument being provided to the the object_add API call. (closes issue #13678) Reported by: ys Tested by: Russell ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@164738 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	2bfe78f74e	Merged revisions 164423 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r164423 \| mmichelson \| 2008-12-15 13:53:29 -0600 (Mon, 15 Dec 2008) \| 11 lines Merged revisions 164422 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164422 \| mmichelson \| 2008-12-15 13:53:08 -0600 (Mon, 15 Dec 2008) \| 3 lines Add the deadlock note to ast_spawn_extension as well ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@164424 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	f981298c64	Merged revisions 164419 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r164419 \| mmichelson \| 2008-12-15 13:51:24 -0600 (Mon, 15 Dec 2008) \| 12 lines Merged revisions 164416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164416 \| mmichelson \| 2008-12-15 13:45:07 -0600 (Mon, 15 Dec 2008) \| 4 lines Add notes to autoservice and pbx doxygen regarding a potential deadlock scenario so that it is avoided in the future ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@164420 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	87c8ebd562	Merged revisions 164257 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r164257 \| file \| 2008-12-15 11:41:22 -0400 (Mon, 15 Dec 2008) \| 4 lines Make app_fax compatible with newer versions of spandsp. This remains backwards compatible with earlier versions though so do not fret. (closes issue #14073) Reported by: seandarcy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@164265 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	d94b6eeeaf	Merged revisions 163449 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r163449 \| russell \| 2008-12-12 07:55:30 -0600 (Fri, 12 Dec 2008) \| 34 lines Merged revisions 163448 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163448 \| russell \| 2008-12-12 07:44:08 -0600 (Fri, 12 Dec 2008) \| 26 lines Resolve issues that could cause DTMF to be processed out of order. These changes come from team/russell/issue_12658 1) Change autoservice to put digits on the head of the channel's frame readq instead of the tail. If there were frames on the readq that autoservice had not yet read, the previous code would have resulted in out of order processing. This required a new API call to queue a frame to the head of the queue instead of the tail. 2) Change up the processing of DTMF in ast_read(). Some of the problems were the result of having two sources of pending DTMF frames. There was the dtmfq and the more generic readq. Both were used for pending DTMF in various scenarios. Simplifying things to only use the frame readq avoids some of the problems. 3) Fix a bug where a DTMF END frame could get passed through when it shouldn't have. If code set END_DTMF_ONLY in the middle of digit emulation, and a digit arrived before emulation was complete, digits would get processed out of order. (closes issue #12658) Reported by: dimas Tested by: russell, file Review: http://reviewboard.digium.com/r/85/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@163450 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	327262b4bb	Merged revisions 162488 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r162488 \| kpfleming \| 2008-12-09 17:41:02 -0600 (Tue, 09 Dec 2008) \| 1 line it does help if the compiler attribute syntax is correct ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@162489 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	1f2e9439fd	Merged revisions 162414 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162414 \| russell \| 2008-12-09 16:25:06 -0600 (Tue, 09 Dec 2008) \| 16 lines Merged revisions 162413 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162413 \| russell \| 2008-12-09 16:17:39 -0600 (Tue, 09 Dec 2008) \| 8 lines Remove the test_for_thread_safety() function completely. The test is not valid. Besides, if we actually suspected that recursive mutexes were not working, we would get a ton of LOG_ERROR messages when DEBUG_THREADS is turned on. (inspired by a discussion on the asterisk-dev list) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@162415 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	a4189e547f	Merged revisions 162079 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r162079 \| murf \| 2008-12-09 10:18:03 -0700 (Tue, 09 Dec 2008) \| 53 lines Merged revisions 162013 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162013 \| murf \| 2008-12-09 09:31:55 -0700 (Tue, 09 Dec 2008) \| 45 lines (closes issue #14019) Reported by: ckjohnsonme Patches: 14019.diff uploaded by murf (license 17) Tested by: ckjohnsonme, murf This crash was the result of a few small errors that would combine in 64-bit land to result in a crash. 32-bit land might have seen these combine to mysteriously drop the args to an application call, in certain circumstances. Also, in trying to find this bug, I spotted a situation in the flex input, where, in passing back a 'word' to the parser, it would allocate a buffer larger than necessary. I changed the usage in such situations, so that strdup was not used, but rather, an ast_malloc, followed by ast_copy_string. I removed a field from the pval struct, in u2, that was never getting used, and set in one spot in the code. I believe it was an artifact of a previous fix to make switch cases work invisibly with extens. And, for goto's I removed a '!' from before a strcmp, that has been there since the initial merging of AEL2, that might prevent the proper target of a goto from being found. This was pretty harmless on its own, as it would just louse up a consistency check for users. Many thanks to ckjohnsonme for providing a simplified and complete set of information about the bug, that helped considerably in finding and fixing the problem. Now, to get aelparse up and running again in trunk, and out of its "horribly broken" state, so I can run the regression suite! ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@162080 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Sean Bright	d9c4ba0231	Merged revisions 161427 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r161427 \| seanbright \| 2008-12-05 16:08:43 -0500 (Fri, 05 Dec 2008) \| 22 lines Merged revisions 161426 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r161426 \| seanbright \| 2008-12-05 16:02:20 -0500 (Fri, 05 Dec 2008) \| 15 lines Merged revisions 161421 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r161421 \| seanbright \| 2008-12-05 15:50:23 -0500 (Fri, 05 Dec 2008) \| 8 lines Fix build errors on FreeBSD (uint -> unsigned int). (closes issue #14006) Reported by: alphaque Patches: astobj2.h-patch uploaded by alphaque (license 259) (Slightly modified by seanbright) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@161428 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	107dfe787e	Merged revisions 152969,153122,154264,154268,154366,155399,155863,156166,156295,156690,156756,158066,158082,158540,158602,159276 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r152969 \| tilghman \| 2008-10-30 15:35:46 -0500 (Thu, 30 Oct 2008) \| 10 lines Merged revisions 152958 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152958 \| tilghman \| 2008-10-30 15:33:28 -0500 (Thu, 30 Oct 2008) \| 3 lines Cannot join detached threads. See http://www.opengroup.org/onlinepubs/000095399/functions/pthread_join.html (Closes issue #13400) ........ ................ r153122 \| tilghman \| 2008-10-31 11:35:21 -0500 (Fri, 31 Oct 2008) \| 10 lines Merged revisions 153114 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r153114 \| tilghman \| 2008-10-31 11:30:32 -0500 (Fri, 31 Oct 2008) \| 3 lines Turn off qualify on uncached realtime peers. (Closes issue #13383) ........ ................ r154264 \| tilghman \| 2008-11-04 12:59:48 -0600 (Tue, 04 Nov 2008) \| 10 lines Recorded merge of revisions 154263 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154263 \| tilghman \| 2008-11-04 12:58:05 -0600 (Tue, 04 Nov 2008) \| 3 lines Make the monitor thread non-detached, so it can be joined (suggested by Russell on -dev list). ........ ................ r154268 \| rmudgett \| 2008-11-04 13:07:26 -0600 (Tue, 04 Nov 2008) \| 11 lines Merged revisions 154266 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154266 \| rmudgett \| 2008-11-04 13:01:08 -0600 (Tue, 04 Nov 2008) \| 4 lines JIRA ABE-1703 mISDN sets the channel to the wrong state when it receives the indication AST_CONTROL_RINGING. ........ ................ r154366 \| tilghman \| 2008-11-04 14:51:18 -0600 (Tue, 04 Nov 2008) \| 16 lines Merged revisions 154365 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r154365 \| tilghman \| 2008-11-04 14:49:33 -0600 (Tue, 04 Nov 2008) \| 9 lines On busy systems, it's possible for the values checked within a single line of code to change, unless the structure is locked to ensure a consistent state. (closes issue #13717) Reported by: kowalma Patches: 20081102__bug13717.diff.txt uploaded by Corydon76 (license 14) Tested by: kowalma ........ ................ r155399 \| tilghman \| 2008-11-07 16:28:58 -0600 (Fri, 07 Nov 2008) \| 14 lines Merged revisions 155398 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155398 \| tilghman \| 2008-11-07 16:27:32 -0600 (Fri, 07 Nov 2008) \| 7 lines Clarify error message. (closes issue #13809) Reported by: denke Patches: 20081104__bug13809.diff.txt uploaded by Corydon76 (license 14) Tested by: denke ........ ................ r155863 \| mmichelson \| 2008-11-10 15:14:44 -0600 (Mon, 10 Nov 2008) \| 22 lines Merged revisions 155861 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155861 \| mmichelson \| 2008-11-10 15:07:39 -0600 (Mon, 10 Nov 2008) \| 14 lines Channel drivers assume that when their indicate callback is invoked, that the channel on which the callback was called is locked. This patch corrects an instance in chan_agent where a channel's indicate callback is called directly without first locking the channel. This was leading to some observed locking issues in chan_local, but considering that all channel drivers operate under the same expectations, the generic fix in chan_agent is the right way to go. AST-126 ........ ................ r156166 \| russell \| 2008-11-12 11:38:20 -0600 (Wed, 12 Nov 2008) \| 15 lines Merged revisions 156164 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156164 \| russell \| 2008-11-12 11:29:52 -0600 (Wed, 12 Nov 2008) \| 7 lines Move the sanity check that makes sure "always fork" is not set along with the console option to be after the code that reads options from asterisk.conf. This resolves a situation where Asterisk can start taking up 100% when misconfigured. (Thanks to Bryce Porter (x86 on IRC) for letting me log in to his system to figure out what was causing the 100% CPU problem.) ........ ................ r156295 \| tilghman \| 2008-11-12 13:28:22 -0600 (Wed, 12 Nov 2008) \| 13 lines Merged revisions 156294 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156294 \| tilghman \| 2008-11-12 13:26:45 -0600 (Wed, 12 Nov 2008) \| 6 lines If the SLA thread is not started, then reload causes a memory leak. (closes issue #13889) Reported by: eliel Patches: app_meetme.c.patch uploaded by eliel (license 64) ........ ................ r156690 \| tilghman \| 2008-11-13 15:30:41 -0600 (Thu, 13 Nov 2008) \| 14 lines Merged revisions 156688 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156688 \| tilghman \| 2008-11-13 15:24:00 -0600 (Thu, 13 Nov 2008) \| 7 lines Provide more space for all the data which can appear in an originating channel name. (closes issue #13398) Reported by: bamby Patches: manager.c.diff uploaded by bamby (license 430) ........ ................ r156756 \| tilghman \| 2008-11-13 18:43:13 -0600 (Thu, 13 Nov 2008) \| 13 lines Merged revisions 156755 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156755 \| tilghman \| 2008-11-13 18:41:37 -0600 (Thu, 13 Nov 2008) \| 6 lines ast_waitfordigit() requires that the channel be up, for no good logical reason. This prevents While/EndWhile from working within the "h" extension. Reported by: jgalarneau (for ABE C.2) Fixed by: me ........ ................ r158066 \| mmichelson \| 2008-11-20 11:39:06 -0600 (Thu, 20 Nov 2008) \| 20 lines Merged revisions 158053 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158053 \| mmichelson \| 2008-11-20 11:33:06 -0600 (Thu, 20 Nov 2008) \| 12 lines Make sure to set the hangup cause on the calling channel in the case that ast_call() fails. For incoming SIP channels, this was causing us to send a 603 instead of a 486 when the call-limit was reached on the destination channel. (closes issue #13867) Reported by: still_nsk Patches: 13867.diff uploaded by putnopvut (license 60) Tested by: blitzrage ........ ................ r158082 \| mmichelson \| 2008-11-20 11:54:31 -0600 (Thu, 20 Nov 2008) \| 24 lines Merged revisions 158071 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158071 \| mmichelson \| 2008-11-20 11:48:42 -0600 (Thu, 20 Nov 2008) \| 16 lines We don't handle 4XX responses to BYE well. According to section 15 of RFC 3261, we should terminate a dialog if we receive a 481 or 408 in response to our BYE. Since I am aware of at least one phone manufacturer who may sometimes send a 404 as well, I am being liberal and saying that any 4XX response to a BYE should result in a terminated dialog. (closes issue #12994) Reported by: pabelanger Patches: 12994.patch uploaded by putnopvut (license 60) Closes AST-129 ........ ................ r158540 \| russell \| 2008-11-21 16:12:37 -0600 (Fri, 21 Nov 2008) \| 10 lines Merged revisions 158539 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158539 \| russell \| 2008-11-21 16:05:55 -0600 (Fri, 21 Nov 2008) \| 2 lines When compiling with DEBUG_THREADS, report the real file/func/line for ao2_lock/ao2_unlock ........ ................ r158602 \| tilghman \| 2008-11-21 17:14:11 -0600 (Fri, 21 Nov 2008) \| 12 lines Merged revisions 158600 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158600 \| tilghman \| 2008-11-21 17:07:46 -0600 (Fri, 21 Nov 2008) \| 5 lines The passed extension may not be the same in the list as the current entry, because we strip spaces when copying the extension into the structure. Therefore, use the copied item to place the item into the list. (found by lmadsen on -dev, fixed by me) ........ ................ r159276 \| tilghman \| 2008-11-25 15:57:59 -0600 (Tue, 25 Nov 2008) \| 14 lines Merged revisions 159269 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159269 \| tilghman \| 2008-11-25 15:56:48 -0600 (Tue, 25 Nov 2008) \| 7 lines Don't try to send a response on a NULL pvt. (closes issue #13919) Reported by: barthpbx Patches: chan_iax2.c.patch uploaded by eliel (license 64) Tested by: barthpbx ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@160389 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	8411899d44	Merged revisions 147518,147689,148000,148112,148268,148917,148988,149062,149131,149201,149205,149208 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r147518 \| file \| 2008-10-08 09:53:51 -0500 (Wed, 08 Oct 2008) \| 9 lines Merged revisions 147517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r147517 \| file \| 2008-10-08 11:51:42 -0300 (Wed, 08 Oct 2008) \| 2 lines If we receive DTMF make sure that the state of the speech structure goes back to being not ready. (issue #LUMENVOX-8) ........ ................ r147689 \| kpfleming \| 2008-10-08 17:26:55 -0500 (Wed, 08 Oct 2008) \| 9 lines Merged revisions 147681 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r147681 \| kpfleming \| 2008-10-08 17:22:09 -0500 (Wed, 08 Oct 2008) \| 3 lines when parsing a text configuration option, ensure that the buffer on the stack is actually large enough to hold the legal values of that option, and also ensure that sscanf() knows to stop parsing if it would overrun the buffer (without these changes, specifying "buffers=...,immediate" would overflow the buffer on the stack, and could not have worked as expected) ........ ................ r148000 \| tilghman \| 2008-10-09 14:39:34 -0500 (Thu, 09 Oct 2008) \| 11 lines Merged revisions 147997 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r147997 \| tilghman \| 2008-10-09 14:38:33 -0500 (Thu, 09 Oct 2008) \| 4 lines When blank, callerid name and number should display "unknown caller" in voicemail emails. (Closes issue #13643) ........ ................ r148112 \| mmichelson \| 2008-10-09 18:15:33 -0500 (Thu, 09 Oct 2008) \| 26 lines Merged revisions 146026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r146026 \| murf \| 2008-10-03 12:12:54 -0500 (Fri, 03 Oct 2008) \| 18 lines (closes issue #13579) Reported by: dwagner (closes issue #13584) Reported by: dwagner Tested by: murf, putnopvut The thought occurred to me that the res= from the extension spawn was ending up being returned from the bridge. "Thou shalt not poison the return value". Made the change and it appears to allow blind xfers to work as normal. If I'm wrong, reopen the bugs. But it looks good to me! Many thanks to putnopvut for helping me reproduce this! ........ ................ r148268 \| tilghman \| 2008-10-10 11:31:31 -0500 (Fri, 10 Oct 2008) \| 14 lines Merged revisions 148257 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r148257 \| tilghman \| 2008-10-10 11:25:31 -0500 (Fri, 10 Oct 2008) \| 7 lines User not notified of temporary greeting, if ODBC storage is in use. (closes issue #13659) Reported by: moliveras Patches: 20081009__bug13659.diff.txt uploaded by Corydon76 (license 14) Tested by: moliveras ........ ................ r148917 \| tilghman \| 2008-10-14 12:46:48 -0500 (Tue, 14 Oct 2008) \| 11 lines Merged revisions 148916 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r148916 \| tilghman \| 2008-10-14 12:41:08 -0500 (Tue, 14 Oct 2008) \| 4 lines Ensure that mail headers are 7-bit clean, even when UTF-8 characters are used in headers like 'Subject' and 'To'. Closes AST-107. ........ ................ r148988 \| tilghman \| 2008-10-14 14:03:44 -0500 (Tue, 14 Oct 2008) \| 9 lines Merged revisions 148987 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r148987 \| tilghman \| 2008-10-14 14:03:08 -0500 (Tue, 14 Oct 2008) \| 2 lines Some compilers warn, some don't. Fixing. ........ ................ r149062 \| tilghman \| 2008-10-14 15:16:48 -0500 (Tue, 14 Oct 2008) \| 13 lines Merged revisions 149061 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149061 \| tilghman \| 2008-10-14 15:09:06 -0500 (Tue, 14 Oct 2008) \| 6 lines Check correct values in the return of ast_waitfor(); also, get rid of a possible memory leak. (closes issue #13658) Reported by: explidous Patch by: me ........ ................ r149131 \| mmichelson \| 2008-10-14 16:08:48 -0500 (Tue, 14 Oct 2008) \| 15 lines Merged revisions 149130 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149130 \| mmichelson \| 2008-10-14 15:49:02 -0500 (Tue, 14 Oct 2008) \| 7 lines Don't allow reserved characters to be used in register lines in sip.conf. (closes issue #13570) Reported by: putnopvut ........ ................ r149201 \| mmichelson \| 2008-10-14 17:41:13 -0500 (Tue, 14 Oct 2008) \| 20 lines Merged revisions 149200 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149200 \| mmichelson \| 2008-10-14 17:40:42 -0500 (Tue, 14 Oct 2008) \| 12 lines Update the queue with the correct number of calls and whether the call was completed within the service level when a transfer takes place. This way, we do not "break" the leastrecent and fewestcalls strategies by not logging a call until after the transferred call has ended. (closes issue #13395) Reported by: Marquis Patches: app_queue.c.transfer.patch uploaded by Marquis (license 32) ........ ................ r149205 \| mmichelson \| 2008-10-14 18:04:44 -0500 (Tue, 14 Oct 2008) \| 20 lines Merged revisions 149204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149204 \| mmichelson \| 2008-10-14 18:00:01 -0500 (Tue, 14 Oct 2008) \| 12 lines Add a tolerance period for sync-triggered audiohooks so that if packetization of audio is close (but not equal) we don't end up flushing the audiohooks over small inconsistencies in synchronization. Related to issue #13005, and solves the issue for most people who were experiencing the problem. However, a small number of people are still experiencing the problem on long calls, so I am not closing the issue yet ........ ................ r149208 \| mmichelson \| 2008-10-14 18:15:04 -0500 (Tue, 14 Oct 2008) \| 17 lines Merged revisions 149207 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149207 \| mmichelson \| 2008-10-14 18:10:26 -0500 (Tue, 14 Oct 2008) \| 9 lines Call register_peer_exten even in the case that the peer's IP/port does not change. (closes issue #13309) Reported by: dimas Patches: v2-13309.patch uploaded by dimas (license 88) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@160387 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	0a9c41ca89	Merged revisions 160208 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r160208 \| tilghman \| 2008-12-01 18:37:21 -0600 (Mon, 01 Dec 2008) \| 10 lines Merged revisions 160207 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160207 \| tilghman \| 2008-12-01 18:25:16 -0600 (Mon, 01 Dec 2008) \| 3 lines Ensure that Asterisk builds with --enable-dev-mode, even on the latest gcc and glibc. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@160228 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Sean Bright	470870f3fc	Merged revisions 154919 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r154919 \| seanbright \| 2008-11-05 17:01:22 -0500 (Wed, 05 Nov 2008) \| 2 lines Fix a problem found while building res_snmp. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@160096 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	2eb5c30a3a	Merged revisions 159818 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r159818 \| kpfleming \| 2008-11-29 11:57:39 -0600 (Sat, 29 Nov 2008) \| 18 lines incorporates r159808 from branches/1.4: ------------------------------------------------------------------------ r159808 \| kpfleming \| 2008-11-29 10:58:29 -0600 (Sat, 29 Nov 2008) \| 7 lines update dev-mode compiler flags to match the ones used by default on Ubuntu Intrepid, so all developers will see the same warnings and errors since this branch already had some printf format attributes, enable checking for them and tag functions that didn't have them format attributes in a consistent way ------------------------------------------------------------------------ in addition: move some format attributes from main/utils.c to the header files they belong in, and fix up references to the relevant functions based on new compiler warnings ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@159855 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	c74fff5ebc	Merged revisions 158540 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r158540 \| russell \| 2008-11-21 16:12:37 -0600 (Fri, 21 Nov 2008) \| 10 lines Merged revisions 158539 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158539 \| russell \| 2008-11-21 16:05:55 -0600 (Fri, 21 Nov 2008) \| 2 lines When compiling with DEBUG_THREADS, report the real file/func/line for ao2_lock/ao2_unlock ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@158542 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	ac0c9bbc4d	Merged revisions 157706 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r157706 \| kpfleming \| 2008-11-19 06:42:19 -0600 (Wed, 19 Nov 2008) \| 5 lines make some corrections to the ast_agi_register_multiple(), ast_agi_unregister_multiple() and ast_agi_fdprintf() API calls to be consistent with API guidelines also, move UPGRADE.txt to UPGRADE-1.6.txt and make the new UPGRADE.txt contain information about upgrading between Asterisk 1.6 releases ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@157738 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	35a7e1deca	Merged revisions 157306 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r157306 \| mmichelson \| 2008-11-18 12:31:08 -0600 (Tue, 18 Nov 2008) \| 20 lines Merged revisions 157305 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157305 \| mmichelson \| 2008-11-18 12:25:55 -0600 (Tue, 18 Nov 2008) \| 12 lines Fix a crash in the end_bridge_callback of app_dial and app_followme which would occur at the end of an attended transfer. The error occurred because we initially stored a pointer to an ast_channel which then was hung up due to a masquerade. This commit adds a "fixup" callback to the bridge_config structure to allow for end_bridge_callback_data to be changed in the case that a new channel pointer is needed for the end_bridge_callback. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@157307 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	4fbe80cc63	This is the 1.6.0 version of revision 156883 of trunk. This is different in that it preserves the case-sensitiveness of processing queues from configuration. closes issue #13703 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@156889 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Sean Bright	0591cc3029	Merged revisions 155554 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r155554 \| seanbright \| 2008-11-08 20:27:00 -0500 (Sat, 08 Nov 2008) \| 14 lines Merged revisions 155553 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155553 \| seanbright \| 2008-11-08 20:08:07 -0500 (Sat, 08 Nov 2008) \| 6 lines Use static functions here instead of nested ones. This requires a small change to the ast_bridge_config struct as well. To understand the reason for this change, see the following post: http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@155555 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	ede179faec	Merge revision 153709 from trunk ------------------------------------------------------------------------ r153709 \| kpfleming \| 2008-11-02 17:34:39 -0600 (Sun, 02 Nov 2008) \| 3 lines instead of trying to forcibly load res_agi when app_stack is loaded (even if the administrator didn't want it loaded), use GCC weak symbols to determine whether it was loaded already or not; if it was loaded, then use it. ------------------------------------------------------------------------ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@153745 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Terry Wilson	e23be17786	Merged revisions 153181 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r153181 \| twilson \| 2008-10-31 13:55:33 -0500 (Fri, 31 Oct 2008) \| 5 lines Recent CDR fixes moved execution of the 'h' exten into the bridging code, so variables that were set after ast_bridge_call was called would not show up in the 'h' exten. Added a callback function to handle setting variables, etc. from w/in the bridging code. Calls back into a nested function within the function calling ast_bridge_call (closes issue #13793) Reported by: greenfieldtech ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@153265 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Sean Bright	eb87d6dac7	Merged revisions 148200 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r148200 \| seanbright \| 2008-10-09 20:42:13 -0400 (Thu, 09 Oct 2008) \| 12 lines Don't include logger.h in asterisk.h by default as it is causing problems building app_voicemail. Instead, include it where it is needed. This turned out to be a relatively minor issue because other headers include logger.h as well. Need to test -addons before merging this back to 1.6.0. (closes issue #13605) Reported by: tomo1657 Patches: 13605_seanbright.diff uploaded by seanbright (license 71) Tested by: mmichelson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@148204 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Michiel van Baak	de553c362d	Merged revisions 147899 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r147899 \| mvanbaak \| 2008-10-09 19:48:53 +0200 (Thu, 09 Oct 2008) \| 5 lines only include this for OpenBSD. At least FreeBSD is borked when including it (closes issue #13649) Reported by: ys ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@147900 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	062d3041fe	Merged revisions 147807 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r147807 \| murf \| 2008-10-09 08:17:33 -0600 (Thu, 09 Oct 2008) \| 15 lines (closes issue #13557) Reported by: nickpeirson Patches: pbx.c.patch uploaded by nickpeirson (license 579) replace_bzero+bcopy.patch uploaded by nickpeirson (license 579) Tested by: nickpeirson, murf 1. replaced all refs to bzero and bcopy to memset and memmove instead. 2. added a note to the CODING-GUIDELINES 3. add two macros to asterisk.h to prevent bzero, bcopy from creeping back into the source 4. removed bzero from configure, configure.ac, autoconfig.h.in ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@147809 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	cdb76a9f1a	Merged revisions 146928 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r146928 \| tilghman \| 2008-10-06 18:21:02 -0500 (Mon, 06 Oct 2008) \| 3 lines Update documentation; AST_THREADSTORAGE() in trunk only takes a single argument. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@146929 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jeff Peeler	69ffdd5f69	Merged revisions 146920 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r146920 \| jpeeler \| 2008-10-06 17:59:58 -0500 (Mon, 06 Oct 2008) \| 2 lines Mvanbaak said this was needed to compile on OpenBSD, so put it in the OpenBSD section. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@146922 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Michiel van Baak	49fd015123	Merged revisions 146807 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r146807 \| mvanbaak \| 2008-10-06 23:18:13 +0200 (Mon, 06 Oct 2008) \| 2 lines make aescrypt.c compile on OpenBSD again ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@146873 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	ccc3bcfb67	Merged revisions 144949-144951 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r144949 \| kpfleming \| 2008-09-27 10:52:56 -0500 (Sat, 27 Sep 2008) \| 17 lines Merged revisions 144924-144925 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r144924 \| kpfleming \| 2008-09-27 10:00:48 -0500 (Sat, 27 Sep 2008) \| 6 lines improve header inclusion process in a few small ways: - it is no longer necessary to forcibly include asterisk/autoconfig.h; every module already includes asterisk.h as its first header (even before system headers), which serves the same purpose - astmm.h is now included by asterisk.h when needed, instead of being forced by the Makefile; this means external modules will build properly against installed headers with MALLOC_DEBUG enabled - simplify the usage of some of these headers in the AEL-related stuff in the utils directory ........ r144925 \| kpfleming \| 2008-09-27 10:13:30 -0500 (Sat, 27 Sep 2008) \| 2 lines fix some minor issues with rev 144924 ........ ................ r144950 \| kpfleming \| 2008-09-27 11:10:33 -0500 (Sat, 27 Sep 2008) \| 2 lines fix bugs caused by r144949 when MALLOC_DEBUG is defined ................ r144951 \| kpfleming \| 2008-09-27 11:17:43 -0500 (Sat, 27 Sep 2008) \| 1 line remove incorrect comment ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@144991 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	660e02982e	Merged revisions 144523 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r144523 \| murf \| 2008-09-25 15:18:12 -0600 (Thu, 25 Sep 2008) \| 13 lines I added a little verbage to hashtab for the hashtab_destroy func. It was pretty sparsely documented. This update fleshes out the pbx_lua module, to add the switch statements to the extensions in the extensions.lua file, as well as removing them when the module is unloaded. Many thanks to Matt Nicholson for his fine contribution! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@144524 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	3c1c24500b	Merged revisions 142676 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r142676 \| murf \| 2008-09-11 22:50:48 -0600 (Thu, 11 Sep 2008) \| 40 lines Merged revisions 142675 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142675 \| murf \| 2008-09-11 22:29:34 -0600 (Thu, 11 Sep 2008) \| 29 lines Tested by: sergee, murf, chris-mac, andrew, KNK This is a "second attempt" to restore the previous "endbeforeh" behavior in 1.4 and up. In order to capture information concerning all the legs of transfers in all their infinite combinations, I was forced to this particular solution by a chain of logical necessities, the first being that I was not allowed to rewrite the CDR mechanism from the ground up! This change basically leaves the original machinery alone, which allows IVR and local channel type situations to generate CDR's as normal, but a channel flag can be set to suppress the normal running of the h exten. That flag would be set by the code that runs the h exten from the ast_bridge_call routine, to prevent the h exten from being run twice. Also, a flag in the ast_bridge_config struct passed into ast_bridge_call can be used to suppress the running of the h exten in that routine. This would happen, for instance, if you use the 'g' option in the Dial app. Running this routine 'early' allows not only the CDR() func to be used in the h extension for reading CDR variables, but also allows them to be modified before the CDR is posted to the backends. While I dearly hope that this patch overcomes all problems, and introduces no new problems, reality suggests that surely someone will have problems. In this case, please re-open 13251 (or 13289), and we'll see if we can't fix any remaining issues. ** trunk note: some code to suppress the h exten being run from app_queue was added; for the 'continue' option available only in trunk/1.6.x. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@142677 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	c9c22bc30f	Merged revisions 139554 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139554 \| mmichelson \| 2008-08-22 14:45:41 -0500 (Fri, 22 Aug 2008) \| 16 lines Merged revisions 139553 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139553 \| mmichelson \| 2008-08-22 14:45:19 -0500 (Fri, 22 Aug 2008) \| 8 lines Fix compilation when DEBUG_THREAD_LOCALS is selected (closes issue #13298) Reported by: snuffy Patches: bug13298_20080822.diff uploaded by snuffy (license 35) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@139555 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	80d244e8ec	Merged revisions 136947 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r136947 \| tilghman \| 2008-08-09 10:26:27 -0500 (Sat, 09 Aug 2008) \| 18 lines Merged revisions 136946 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r136946 \| tilghman \| 2008-08-09 10:25:36 -0500 (Sat, 09 Aug 2008) \| 10 lines Merged revisions 136945 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r136945 \| tilghman \| 2008-08-09 10:24:36 -0500 (Sat, 09 Aug 2008) \| 2 lines Regression fixes for Solaris ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@136948 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	52d466de03	Merged revisions 136746 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r136746 \| murf \| 2008-08-07 18:48:35 -0600 (Thu, 07 Aug 2008) \| 40 lines Merged revisions 136726 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r136726 \| murf \| 2008-08-07 18:15:34 -0600 (Thu, 07 Aug 2008) \| 32 lines (closes issue #13236) Reported by: korihor Wow, this one was a challenge! I regrouped and ran a new strategy for setting the ~~MACRO~~ value; I set it once per extension, up near the top. It is only set if there is a switch in the extension. So, I had to put in a chunk of code to detect a switch in the pval tree. I moved the code to insert the set of ~~exten~~ up to the beginning of the gen_prios routine, instead of down in the switch code. I learned that I have to push the detection of the switches down into the code, so everywhere I create a new exten in gen_prios, I make sure to pass onto it the values of the mother_exten first, and the exten next. I had to add a couple fields to the exten struct to accomplish this, in the ael_structs.h file. The checked field makes it so we don't repeat the switch search if it's been done. I also updated the regressions. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@136778 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	eac67f84f4	Merged revisions 136542 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r136542 \| kpfleming \| 2008-08-07 12:44:20 -0500 (Thu, 07 Aug 2008) \| 6 lines Merged revisions 136541 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@136543 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	4a854a5e5b	Merged revisions 135900 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135900 \| tilghman \| 2008-08-05 22:04:01 -0500 (Tue, 05 Aug 2008) \| 12 lines Merged revisions 135899 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135899 \| tilghman \| 2008-08-05 22:02:59 -0500 (Tue, 05 Aug 2008) \| 4 lines 1) Bugfix for debugging code 2) Reduce compiler warnings for another section of debugging code (Closes issue #13237) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@135901 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	8417f7bfb1	Merged revisions 135851 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135851 \| mmichelson \| 2008-08-05 19:30:53 -0500 (Tue, 05 Aug 2008) \| 48 lines Merged revisions 135841,135847,135850 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135841 \| mmichelson \| 2008-08-05 19:25:10 -0500 (Tue, 05 Aug 2008) \| 27 lines Merging the issue11259 branch. The purpose of this branch was to take into account "burps" which could cause jitterbuffers to misbehave. One such example is if the L option to Dial() were used to inject audio into a bridged conversation at regular intervals. Since the audio here was not passed through the jitterbuffer, it would cause a gap in the jitterbuffer's timestamps which would cause a frames to be dropped for a brief period. Now ast_generic_bridge will empty and reset the jitterbuffer each time it is called. This causes injected audio to be handled properly. ast_generic_bridge also will empty and reset the jitterbuffer if it receives an AST_CONTROL_SRCUPDATE frame since the change in audio source could negatively affect the jitterbuffer. All of this was made possible by adding a new public API call to the abstract_jb called ast_jb_empty_and_reset. (closes issue #11259) Reported by: plack Tested by: putnopvut ........ r135847 \| mmichelson \| 2008-08-05 19:27:54 -0500 (Tue, 05 Aug 2008) \| 4 lines Revert inadvertent changes to app_skel that occurred when I was testing for a memory leak ........ r135850 \| mmichelson \| 2008-08-05 19:29:54 -0500 (Tue, 05 Aug 2008) \| 3 lines Remove properties that should not be here ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@135852 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	638711f573	Merged revisions 135821 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135821 \| murf \| 2008-08-05 17:45:32 -0600 (Tue, 05 Aug 2008) \| 42 lines Merged revisions 135799 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135799 \| murf \| 2008-08-05 17:13:20 -0600 (Tue, 05 Aug 2008) \| 34 lines (closes issue #12982) Reported by: bcnit Tested by: murf I discovered that also, in the previous bug fixes and changes, the cdr.conf 'unanswered' option is not being obeyed, so I fixed this. And, yes, there are two 'answer' times involved in this scenario, and I would agree with you, that the first answer time is the time that should appear in the CDR. (the second 'answer' time is the time that the bridge was begun). I made the necessary adjustments, recording the first answer time into the peer cdr, and then using that to override the bridge cdr's value. To get the 'unanswered' CDRs to appear, I purposely output them, using the dial cmd to mark them as DIALED (with a new flag), and outputting them if they bear that flag, and you are in the right mode. I also corrected one small mention of the Zap device to equally consider the dahdi device. I heavily tested 10-sec-wait macros in dial, and without the macro call; I tested hangups while the macro was running vs. letting the macro complete and the bridge form. Looks OK. Removed all the instrumentation and debug. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@135822 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	6d5f021393	Merged revisions 134703 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r134703 \| tilghman \| 2008-07-30 17:38:58 -0500 (Wed, 30 Jul 2008) \| 2 lines Oops, wrong define ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@134706 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	7eb4b5bf5b	Merged revisions 134050 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r134050 \| mmichelson \| 2008-07-28 11:00:19 -0500 (Mon, 28 Jul 2008) \| 3 lines merging the zap_and_dahdi_trunk branch up to trunk ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@134052 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	86cef15735	Include the licensing page in 1.6.0 as well. Now, this page exists in 1.4, trunk, and 1.6.0. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@133982 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	7d7e554191	Merged revisions 133171 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r133171 \| mmichelson \| 2008-07-23 14:48:03 -0500 (Wed, 23 Jul 2008) \| 20 lines Merged revisions 133169 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133169 \| mmichelson \| 2008-07-23 14:39:47 -0500 (Wed, 23 Jul 2008) \| 12 lines As suggested by seanbright, the PSEUDO_CHAN_LEN in app_chanspy should be set at load time, not at compile time, since dahdi_chan_name is determined at load time. Also changed the next_unique_id_to_use to have the static qualifier. Also added the dahdi_chan_name_len variable so that strlen(dahdi_chan_name) isn't necessary. Thanks to seanbright for the suggestion. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@133172 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	ccf54d5c34	Merged revisions 132964 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132964 \| kpfleming \| 2008-07-23 11:30:18 -0500 (Wed, 23 Jul 2008) \| 10 lines Merged revisions 132872 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132872 \| kpfleming \| 2008-07-23 06:52:18 -0500 (Wed, 23 Jul 2008) \| 2 lines minor optimization for stringfields: when a field is being set to a larger value than it currently contains and it happens to be the most recent field allocated from the currentl pool, it is possible to 'grow' it without having to waste the space it is currently using (or potentially even allocate a new pool) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@132965 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	de75687750	Merged revisions 132643 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132643 \| kpfleming \| 2008-07-22 14:59:10 -0500 (Tue, 22 Jul 2008) \| 10 lines Merged revisions 132641 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132641 \| kpfleming \| 2008-07-22 14:49:11 -0500 (Tue, 22 Jul 2008) \| 2 lines use renamed libpri API call for controlling this feature (was improperly named before) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@132644 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	bd2b20c845	Merged revisions 132511 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132511 \| tilghman \| 2008-07-21 16:00:47 -0500 (Mon, 21 Jul 2008) \| 2 lines (Step 2 of 2) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@132513 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	1ab6423d16	Merged revisions 132510 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132510 \| tilghman \| 2008-07-21 15:59:03 -0500 (Mon, 21 Jul 2008) \| 5 lines Optionally build integer-based routines for FSK tone decoding (but default to the more accurate float-based routines). (Closes issue #11679) (Step 1 of 2) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@132512 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	5b266380e2	Merged revisions 132390 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132390 \| russell \| 2008-07-21 09:47:41 -0500 (Mon, 21 Jul 2008) \| 16 lines Remove libresample from the Asterisk source tree. It is now available in its own repository, and must be installed like any other library for Asterisk to use. The two modules that require it are codec_resample and app_jack. To install libresample: $ svn co http://svn.digium.com/svn/libresample/trunk libresample $ cd libresample $ ./configure $ make $ sudo make install This code is currently in our own repository because the build system did not include the appropriate targets for building a dynamic library or for installing the library. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@132391 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	c01f4f499e	Merged revisions 131986 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131986 \| tilghman \| 2008-07-18 11:48:18 -0500 (Fri, 18 Jul 2008) \| 10 lines Merged revisions 131985 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131985 \| tilghman \| 2008-07-18 11:46:23 -0500 (Fri, 18 Jul 2008) \| 2 lines Preserve ABI compatibility with last change ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@131987 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	5abbe30e52	Merged revisions 131982 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131982 \| tilghman \| 2008-07-18 11:33:56 -0500 (Fri, 18 Jul 2008) \| 10 lines Merged revisions 131970 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131970 \| tilghman \| 2008-07-18 11:30:31 -0500 (Fri, 18 Jul 2008) \| 2 lines Make the ast_assert call within ast_sched_del report something useful. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@131984 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago

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