95 Commits (600d737dceb8a98a2135aa809e623cc26c58905c)

Author	SHA1	Message	Date
Joshua Colp	809c1398d7	Merged revisions 72257 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r72257 | file | 2007-06-27 16:25:24 -0400 (Wed, 27 Jun 2007) | 10 lines Merged revisions 72256 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r72256 | file | 2007-06-27 16:23:24 -0400 (Wed, 27 Jun 2007) | 2 lines I may possibly get shot for doing this... but... defer CDR processing until after the channel has been dealt with. This should eliminate all of the issues with channels going funky (SIP/PRI) when you are posting CDRs to a database that is either slow or unavailable and do not want to enable batching. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@72258 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	7feaaaaf04	Merged revisions 72148 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r72148 | file | 2007-06-27 13:31:50 -0400 (Wed, 27 Jun 2007) | 2 lines Make the ast_read_noaudio API call behave better under circumstances where DTMF emulation was happening and a generator was setup. (issue #10065 reported by stevefeinstein) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@72149 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Steve Murphy	cd97d6a687	Merged revisions 70062 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r70062 | murf | 2007-06-19 12:23:23 -0600 (Tue, 19 Jun 2007) | 9 lines Merged revisions 70053 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r70053 | murf | 2007-06-19 12:07:59 -0600 (Tue, 19 Jun 2007) | 1 line This fixes 9246, where channel variables are not available in the 'h' exten, on a 'ZOMBIE' channel. The fix is to consolidate the channel variables during a masquerade, and then copy the merged variables back onto the clone, so the zombie has the same vars that the 'original' has. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@70063 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	afe63aec33	Merged revisions 69987 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r69987 | file | 2007-06-19 12:24:31 -0400 (Tue, 19 Jun 2007) | 10 lines Merged revisions 69986 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r69986 | file | 2007-06-19 12:21:29 -0400 (Tue, 19 Jun 2007) | 2 lines Update BRIDGEPEER variable if set to the new channel name when a masquerade happens. (issue #9699 reported by dimas) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69988 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	b179e2155f	Convert uses of strdup() to ast_strdup() ... (issue #9983, eliel) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69436 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	055d82cbce	Add a massive set of changes for converting to use the ast_debug() macro. ... (issue #9957, patches from mvanbaak, caio1982, critch, and dimas) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69327 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	cf06bdb312	Merged revisions 69010 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r69010 | russell | 2007-06-12 14:13:41 -0500 (Tue, 12 Jun 2007) | 12 lines In ast_channel_make_compatible(), just return if the channels' read and write formats already match up. There are code paths that call this function on a pair of channels multiple times. This made calls fail that were using g729 in some cases. The reason is that codec_g729a will unregister itself from the list of available translators will all licenses are in use. So, the first time the function got called, the right translation path was allocated. However, the second time it got called, the code would not find a translation path to/from g729 and make the call fail, even if the channel actually already had a g729 translation path allocated. (SPD-32) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69011 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	b3475e429a	Minor code cleanup. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@68901 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	a9043635ed	Change channel list to read/write list... I'm crazy. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@68685 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	eb56cb6b92	Merged revisions 68683 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r68683 | file | 2007-06-11 10:33:12 -0400 (Mon, 11 Jun 2007) | 10 lines Merged revisions 68682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r68682 | file | 2007-06-11 10:29:58 -0400 (Mon, 11 Jun 2007) | 2 lines Improve deadlock handling of the channel list. (issue #8376 reported by one47) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@68684 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	ef7b38e065	Merged revisions 68157 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r68157 | file | 2007-06-07 14:39:52 -0400 (Thu, 07 Jun 2007) | 2 lines Fix logic when doing a name based channel search for a structure when you want to start from a specific point in the channel list. (issue #9324 reported by slavon) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@68158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Tilghman Lesher	9d05ff8ed5	Issue 9869 - replace malloc and memset with ast_calloc, and other coding guidelines changes ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67864 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	18101ff0e3	Merged revisions 67716 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r67716 | russell | 2007-06-06 11:55:59 -0500 (Wed, 06 Jun 2007) | 13 lines Merged revisions 67715 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r67715 | russell | 2007-06-06 11:40:51 -0500 (Wed, 06 Jun 2007) | 5 lines We have some bug reports showing crashes due to a double free of a channel. Add a sanity check to ast_channel_free() to make sure we don't go on trying to free a channel that wasn't found in the channel list. (issue #8850, and others...) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67717 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	2b25070414	Merged revisions 66076 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r66076 | russell | 2007-05-24 17:23:59 -0500 (Thu, 24 May 2007) | 1 line if the string field init fails, clean up the stuff that was allocated already ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66077 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	9065eb1054	Merged revisions 66070 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r66070 | russell | 2007-05-24 17:07:39 -0500 (Thu, 24 May 2007) | 2 lines Check the result of ast_string_field_init() in ast_channel_alloc() ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66072 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	90d6885701	Add a new feature for Music on Hold. If you set the "digit" option for a ... class in musiconhold.conf, a caller on hold may press this digit to switch to listening to that music class. This involved adding a new callback for generators, which allow generators to get notified of DTMF from the channel they are running on. Then, a callback was implemented for the music on hold generators. (patch from bbryant) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65505 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	f02ea9e076	Merged revisions 64240 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64240 | file | 2007-05-14 13:23:51 -0400 (Mon, 14 May 2007) | 2 lines Fix scenario where if a phone that simply called Echo() put itself on hold it could never get off hold. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64242 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	78d3244554	Merged revisions 64157 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64157 | oej | 2007-05-14 12:39:12 +0200 (Mon, 14 May 2007) | 2 lines Add hangupcause when we lack codecs for transcoding ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	9427cf5dd6	Merged revisions 63698 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r63698 | file | 2007-05-09 15:22:39 -0400 (Wed, 09 May 2007) | 2 lines Use the DTMF frame on the channel when returning a DTMF frame from AST_FRAME_NULL or AST_FRAME_VOICE. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63699 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	56254ee491	Merged revisions 63612 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r63612 | russell | 2007-05-09 11:55:27 -0500 (Wed, 09 May 2007) | 5 lines Modify ast_senddigit_begin() to use the same assumptions used elsewhere in the code in that if a channel does not have a send_digit_begin() callback, it only cares about DTMF END events. (pointed out by Michael Neuhauser on the asterisk-dev list) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63697 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	39dd02d542	Merged revisions 63608 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r63608 | russell | 2007-05-09 11:43:50 -0500 (Wed, 09 May 2007) | 5 lines Only call ast_senddigit_begin() in ast_senddigit() if the channel has a send_digit_begin() callback. Checking the END_DTMF_ONLY flag was the wrong thing to do, because that flag indicates that a *bridged* channel only wants DTMF END events coming from this channel. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63609 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	5394364048	Merged revisions 63286 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r63286 | file | 2007-05-07 17:45:01 -0400 (Mon, 07 May 2007) | 10 lines Merged revisions 63285 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r63285 | file | 2007-05-07 17:39:52 -0400 (Mon, 07 May 2007) | 2 lines Properly handle what happens during a masquerade in relation to group counting. (issue #9657 reported by ramonpeek) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63287 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	1e3b1a576c	Merged revisions 62942 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r62942 | russell | 2007-05-03 10:23:13 -0500 (Thu, 03 May 2007) | 17 lines Fix YADB (Yet Another DTMF Bug) ((C) Russell Bryant, 2007, TM, Patent Pending). This set of changes came from a debugging session I had with Dwayne Hubbard. When he called into his home FXO, ran the Echo application, and pressed a digit, the digit would be echoed back and would never end. This is fixed, along with a couple other little improvements. * When chan_zap is in the middle of playing a digit to a channel, it feeds back null frames, not voice frames. So, I have modified ast_read to check the timing on emulated DTMF when it receives null frames, in addition to where it was doing this on voice frames. * Make a tweak to setting the duration on emulated DTMF digits. If there was no duration specified, it set it to be the minimum, instead of the default. * Instead of timing the emulated digits off of the number of samples in audio frames that pass through, just use time values. Now there is no code in this section that assumes 8kHz audio. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62943 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	255f151582	Merged revisions 62789 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r62789 | russell | 2007-05-02 17:59:09 -0500 (Wed, 02 May 2007) | 20 lines Merge changes from team/russell/inband_dtmf ... Fix some issues related to generating inband DTMF. There are two changes here: 1) The list of DTMF tones in the senddigit_begin() function explicitly specified 100ms of the tone followed by 100ms of silence. This really broke things with the way that Asterisk now wants complete control over when the digit begins and ends. So, regardless of what Asterisk really wanted to do, this was going to play out the tone at the length it wanted to. This caused various problems like DTMF translation to inband to be extremely unreliable. The list of tones has been changed so that the correct DTMF tone is played indefinitely until Asterisk tells it to stop. 2) ast_write() had to be modified to let a DTMF_END frame get processed even when a generator is present. This is how the tone will finally get stopped. (issues #8944, #9250, #9348, maybe others. Thanks to mdu113 from #8944 for the testing and feedback!) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62791 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Steve Murphy	fe7068a51b	Merged revisions 62689 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r62689 | murf | 2007-05-02 11:10:50 -0600 (Wed, 02 May 2007) | 1 line a)In chan_zap, set the clid, src fields in channel_alloc call. b)in the channel_alloc func, set the cid_num and name fields from the arglist[blush]. c) don't update the channel app & app data fields if you are in the 'h' extension. d)the load_module func in cdr_radius needs to return DECLINE, SUCCESS. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62690 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	007fa5e0be	Merged revisions 62005 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r62005 | file | 2007-04-25 22:19:51 -0500 (Wed, 25 Apr 2007) | 2 lines Missed an ast_app_group_discard during merge. Thanks blitzrage! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62006 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	8b2b3e172b	Merged revisions 61805 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61805 | file | 2007-04-25 15:21:54 -0400 (Wed, 25 Apr 2007) | 10 lines Merged revisions 61804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61804 | file | 2007-04-25 14:52:50 -0400 (Wed, 25 Apr 2007) | 2 lines Merge rewritten group counting support. No more storing data on the variable list of the channels. That was bad, mmmk? (issue #7497 reported by sabbathbh) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61806 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	94459660a3	Merged revisions 61781 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61781 | russell | 2007-04-24 14:00:06 -0500 (Tue, 24 Apr 2007) | 6 lines Improve DTMF handling in ast_read() even more in response to a discussion on the asterisk-dev mailing list. I changed the enforced minimum length of a digit from 100ms to 80ms. Furthermore, I made it now enforce a gap of 45ms in between digits. These values are not configurable in a configuration file right now, but they can be easily changed near the top of main/channel.c. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61782 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	fa0e814a69	Merged revisions 61763 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61763 | russell | 2007-04-23 12:57:32 -0500 (Mon, 23 Apr 2007) | 4 lines Ensure that digits passing through Asterisk have a reasonable minimum length. It is currently 100 ms. If someone thinks this should be different, feel free to speak up. (related to issues #8944, #9250, and #9348) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61764 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Tilghman Lesher	47dd5a15af	Issue 6082 - New DTMF event for manager ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61324 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Steve Murphy	ecaf781933	Merged revisions 60989 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60989 | murf | 2007-04-09 12:32:07 -0600 (Mon, 09 Apr 2007) | 1 line This is a big improvement over the current CDR fixes. It may still need refinement, but this won't have as many folks bothered. This also adds the mods from 1.4/r.61136; ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61152 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Steve Murphy	09c0d56c5c	Merged revisions 59522 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59522 | murf | 2007-03-30 11:51:17 -0600 (Fri, 30 Mar 2007) | 1 line several changes via kpflemings review ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59523 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Steve Murphy	0f11d3c8c3	Merged revisions 59486 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59486 | murf | 2007-03-30 08:11:59 -0600 (Fri, 30 Mar 2007) | 1 line These mods fix CDR issues from 8221, 8593, 8680, 8743, and perhaps others. Mainly with CDRs generated from transfer situations. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59500 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Tilghman Lesher	590cb3a6fa	Expand datastores to add the notion of inheritance. This will be needed for ... the conversion of IAX2 variables from the current custom method to ast_storage. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57691 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	b4a29c3782	Constify the list of codec preferences. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57293 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Tilghman Lesher	b0f60e7496	Merged revisions 56685 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r56685 | tilghman | 2007-02-25 08:46:41 -0600 (Sun, 25 Feb 2007) | 11 lines Merged revisions 56684 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r56684 | tilghman | 2007-02-25 08:38:03 -0600 (Sun, 25 Feb 2007) | 3 lines Issue 9130 - If prev is the last item on the channel list, then evaluating additional conditions (e.g. name prefix) will cause a NULL dereference. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56686 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	75d387acbc	Doxygen additions, corrections ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56665 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	afc99294fa	Merged revisions 56231 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r56231 | file | 2007-02-22 13:49:39 -0500 (Thu, 22 Feb 2007) | 10 lines Merged revisions 56230 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r56230 | file | 2007-02-22 13:44:24 -0500 (Thu, 22 Feb 2007) | 2 lines Only change the original or clone channel if it's the channel behind the proxy channel, not if it's just a regular bridged channel. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56232 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	ba32ee49d0	Adding Realtime Text support (T.140) to Asterisk ... T.140/RFC 2793 is a live communication channel, originally created for IP based text phones for hearing impaired. Feels very much like the old Unix talk application. This code is developed and disclaimed by John Martin of Aupix, UK. Tested for interoperability by myself and Omnitor in Sweden, the company that wrote most of the specifications. A big thank you to everyone involved in this. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54838 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	8470ee5cd0	Merged revisions 54290 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r54290 | file | 2007-02-13 20:09:40 -0500 (Tue, 13 Feb 2007) | 2 lines Add G722 to ast_best_codec. If anyone disagrees with it's placement, feel free to change it. (issue #9045 reported by gork) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54291 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	10c4a5fef7	Simplify a small bit of logic. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54003 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Paul Cadach	85ad583544	Merged revisions 53879 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53879 | pcadach | 2007-02-10 01:07:11 -0800 (Сбт, 10 Фев 2007) | 1 line Provide correct DTMF duration ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53883 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	dab41a355d	Merged revisions 51848 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r51848 | russell | 2007-01-23 18:59:58 -0600 (Tue, 23 Jan 2007) | 14 lines Merged revisions 51843 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51843 | russell | 2007-01-23 18:57:28 -0600 (Tue, 23 Jan 2007) | 6 lines Fix an issue related to synchronization of recordings when using Monitor(). The bug is a miscalculation of the amount to seek the stream for writing to disk when the number of samples coming in and out of a channel do not match up. (issue #8298, #8887, report and patch by guillecabeza, patch files created and testing done by whoiswes) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51850 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	21b53af31d	Cosmetic changes. Make main source files better conform to coding guidelines and standards. (issue #8679 reported by johann8384) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51486 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	dcca8f345f	Merged revisions 51311 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines Merge the changes from the /team/group/vldtmf_fixup branch. The main bug being addressed here is a problem introduced when two SIP channels using SIP INFO dtmf have their media directly bridged. So, when a DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk would try to emulate a digit of some length by first sending a DTMF BEGIN frame and sending a DTMF END later timed off of incoming audio. However, since there was no audio coming in, the DTMF_END was never generated. This caused DTMF based features to no longer work. To fix this, the core now knows when a channel doesn't care about DTMF BEGIN frames (such as a SIP channel sending INFO dtmf). If this is the case, then Asterisk will not emulate a digit of some length, and will instead just pass through the single DTMF END event. Channel drivers also now get passed the length of the digit to their digit_end callback. This improves SIP INFO support even further by enabling us to put the real digit duration in the INFO message instead of a hard coded 250ms. Also, for an incoming INFO message, the duration is read from the frame and passed into the core instead of just getting ignored. (issue #8597, maybe others...) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51314 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Luigi Rizzo	5b9114fa73	include "asterisk/zapata.h" to get the zaptel headers. ... this should be the last one left around... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51299 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Jason Parker	11dd11e5a1	Merged revisions 51241 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51241 | qwell | 2007-01-18 12:28:29 -0600 (Thu, 18 Jan 2007) | 2 lines Fix an issue with deprecated commands ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51242 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	c71d6c12f7	Don't hold channel lock while sleeping/waiting for audio stream to get setup. (issue #8834 reported by phsultan) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51193 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	d986f00e73	Merged revisions 50727 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r50727 | file | 2007-01-13 01:00:24 -0500 (Sat, 13 Jan 2007) | 2 lines Only write a frame out to the channel if one exists. There are cases where one may not and would therefore cause the channel driver to segfault. (issue #8434 reported by slimey) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@50728 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Kevin P. Fleming	17ea9c930e	make the automatic post-answer delay happen only when the answer is 'automatic' (not done by the Answer() dialplan application) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@50571 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago