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2007-08-08Merge audiohooks branch into trunk. This is a new API for developers to ↵Joshua Colp
listen and manipulate the audio going through a channel. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78649 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-06Extend the ast_senddigit and ast_dtmf_stream API calls to allow the duration ↵Joshua Colp
of the DTMF digit(s) to be specified and make the SendDTMF application have the capability to use it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78278 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-06Merged revisions 78275 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r78275 | file | 2007-08-06 18:41:13 -0300 (Mon, 06 Aug 2007) | 2 lines Add additional DTMF log messages to help when debugging issues. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78276 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-30print formats as 0x%x instead of %d in a warning message.Luigi Rizzo
Being bitmasks, it is a lot easier to read this way. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77793 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-30Merged revisions 77785 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77785 | russell | 2007-07-30 13:55:15 -0500 (Mon, 30 Jul 2007) | 3 lines file and I both committed changes for issue #10301. Remove a duplicated assignment to restore the original value of the previous channel. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77786 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-30Merged revisions 77780 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77780 | russell | 2007-07-30 12:29:43 -0500 (Mon, 30 Jul 2007) | 16 lines (closes issue #10301) Reported by: fnordian Patches: asterisk-1.4.9-channel.c.patch uploaded by fnordian (license 110) Additional changes by me Fix some problems in channel_find_locked() which can cause an infinite loop. The reference to the previous channel is set to NULL in some cases. These changes ensure that the reference to the previous channel gets restored before needing it again. I'm not convinced that the code that is setting it to NULL is really the right thing to do. However, I am making these changes to fix the obvious problem and just leaving an XXX comment that it needs a better explanation that what is there now. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77781 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-30Merged revisions 77771 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77771 | file | 2007-07-30 12:47:52 -0300 (Mon, 30 Jul 2007) | 6 lines (closes issue #10301) Reported by: fnordian Patches: asterisk-1.4.9-channel.c.patch uploaded by fnordian (license 110) Restore previous behavior where if we failed to lock the channel we wanted we would return to exactly the same point as if we had just reentered the function. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-26Merged revisions 77460 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77460 | file | 2007-07-26 20:19:04 -0300 (Thu, 26 Jul 2007) | 4 lines (closes issue #10302) Reported by: litnialex If a DTMF end frame comes from a channel without a begin and it is going to a technology that only accepts end frames (aka INFO) then use the minimum DTMF duration if one is not in the frame already. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77461 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-26Do a massive conversion for using the ast_verb() macroRussell Bryant
(closes issue #10277, patches by mvanbaak) Basically, this changes ... if (option_verbose > 2) ast_verbose(VERBOSE_PREFIX_3, "Something\n"); to ... ast_verb(3, "Something\n"); git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77299 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-25Merged revisions 77154 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77154 | mmichelson | 2007-07-25 16:52:47 -0500 (Wed, 25 Jul 2007) | 3 lines chan->emulate_dtmf_duration is an unsigned int, not a signed int, so use %u instead of %d in the format string ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77155 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-20Merged revisions 76132 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r76132 | russell | 2007-07-20 13:22:24 -0500 (Fri, 20 Jul 2007) | 6 lines Use the define that specifies the default length of an artificially created DTMF digit in the ast_senddigit() function. The define is set to 100ms by default, which is the same thing that this function was using. But, using the define lets changes take effect in this case, as well as the others where it was already used. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19After some study, thought, comparing, etc. I've backed out the previous ↵Steve Murphy
universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-18This corrects the problem with flags and %lld formats on 64-bit machines, ↵Steve Murphy
where uint64_t is NOT acceptable for %lld, and also works on 32-bit machines. At least, with gcc. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75585 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-17via 10206, I have added an option (e) to Dial to allow the h exten to get ↵Steve Murphy
run on peer. Had to upgrade ast_flag stuff to 64 bits to do this. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75400 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-12Merged revisions 74922 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r74922 | file | 2007-07-12 16:17:59 -0300 (Thu, 12 Jul 2007) | 2 lines Whoops... didn't want this to be returned to 0 each iteration. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@74923 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-12Merged revisions 74888 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r74888 | file | 2007-07-12 14:16:28 -0300 (Thu, 12 Jul 2007) | 2 lines When waiting for a digit ensure that a begin frame was received with it, not just an end frame. (issue #10084 reported by rushowr) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@74891 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-05Merged revisions 73355 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r73355 | file | 2007-07-05 11:21:44 -0300 (Thu, 05 Jul 2007) | 10 lines Merged revisions 73349 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r73349 | file | 2007-07-05 11:19:14 -0300 (Thu, 05 Jul 2007) | 2 lines Tweak spy locking. (issue #9951 reported by welles) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@73359 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-02Merged revisions 72888 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r72888 | file | 2007-07-02 11:32:59 -0300 (Mon, 02 Jul 2007) | 2 lines Added additional DTMF debug messages for when emulation occurs. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@72889 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-27Merged revisions 72257 via svnmerge from Joshua Colp
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
2007-06-27Merged revisions 72148 via svnmerge from Joshua Colp
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
2007-06-19Merged revisions 70062 via svnmerge from Steve Murphy
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
2007-06-19Merged revisions 69987 via svnmerge from Joshua Colp
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
2007-06-14Convert uses of strdup() to ast_strdup()Russell Bryant
(issue #9983, eliel) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-14Add a massive set of changes for converting to use the ast_debug() macro.Russell Bryant
(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
2007-06-12Merged revisions 69010 via svnmerge from Russell Bryant
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
2007-06-12Minor code cleanup.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@68901 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-11Change channel list to read/write list... I'm crazy.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@68685 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-11Merged revisions 68683 via svnmerge from Joshua Colp
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
2007-06-07Merged revisions 68157 via svnmerge from Joshua Colp
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
2007-06-06Issue 9869 - replace malloc and memset with ast_calloc, and other coding ↵Tilghman Lesher
guidelines changes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67864 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-06Merged revisions 67716 via svnmerge from Russell Bryant
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
2007-05-24Merged revisions 66076 via svnmerge from Russell Bryant
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
2007-05-24Merged revisions 66070 via svnmerge from Russell Bryant
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
2007-05-22 Add a new feature for Music on Hold. If you set the "digit" option for aRussell Bryant
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
2007-05-14Merged revisions 64240 via svnmerge from Joshua Colp
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
2007-05-14Merged revisions 64157 via svnmerge from Olle Johansson
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
2007-05-09Merged revisions 63698 via svnmerge from Joshua Colp
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
2007-05-09Merged revisions 63612 via svnmerge from Russell Bryant
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
2007-05-09Merged revisions 63608 via svnmerge from Russell Bryant
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
2007-05-07Merged revisions 63286 via svnmerge from Joshua Colp
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
2007-05-03Merged revisions 62942 via svnmerge from Russell Bryant
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
2007-05-02Merged revisions 62789 via svnmerge from Russell Bryant
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
2007-05-02Merged revisions 62689 via svnmerge from Steve Murphy
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
2007-04-26Merged revisions 62005 via svnmerge from Russell Bryant
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
2007-04-25Merged revisions 61805 via svnmerge from Joshua Colp
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
2007-04-24Merged revisions 61781 via svnmerge from Russell Bryant
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
2007-04-23Merged revisions 61763 via svnmerge from Russell Bryant
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
2007-04-10Issue 6082 - New DTMF event for managerTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61324 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-10Merged revisions 60989 via svnmerge from Steve Murphy
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
2007-03-30Merged revisions 59522 via svnmerge from Steve Murphy
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
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