aboutsummaryrefslogtreecommitdiffstats
path: root/main/audiohook.c
AgeCommit message (Collapse)AuthorFilesLines
2008-12-19Merged revisions 166162 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r166162 | mmichelson | 2008-12-19 17:45:00 -0600 (Fri, 19 Dec 2008) | 6 lines Get rid of an extra space. I don't know how this crept back in when I had already fixed it earlier ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@166163 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-19Merged revisions 166092,166095 via svnmerge from mmichelson1-0/+60
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@166097 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-02Merged revisions ↵tilghman1-2/+8
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@160387 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-14Merged revisions 130635 via svnmerge from russell1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r130635 | russell | 2008-07-14 05:39:23 -0500 (Mon, 14 Jul 2008) | 10 lines Merged revisions 130634 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130634 | russell | 2008-07-14 05:38:14 -0500 (Mon, 14 Jul 2008) | 2 lines Bump up the debug level for a message. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@130636 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-11Merged revisions 130237 via svnmerge from mmichelson1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r130237 | mmichelson | 2008-07-11 15:03:55 -0500 (Fri, 11 Jul 2008) | 11 lines Merged revisions 130236 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130236 | mmichelson | 2008-07-11 15:03:23 -0500 (Fri, 11 Jul 2008) | 3 lines Remove redundant logic ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@130238 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-11Merged revisions 130174 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r130174 | mmichelson | 2008-07-11 14:14:15 -0500 (Fri, 11 Jul 2008) | 15 lines Merged revisions 130173 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130173 | mmichelson | 2008-07-11 14:13:29 -0500 (Fri, 11 Jul 2008) | 7 lines Fix a typo in audiohook_read_frame_both. While this change has not been proven to fix any specific issue, it is incorrect and could cause unforeseen problems. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@130175 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-08Merged revisions 113297 via svnmerge from file1-4/+13
https://origsvn.digium.com/svn/asterisk/trunk ................ r113297 | file | 2008-04-08 12:05:35 -0300 (Tue, 08 Apr 2008) | 12 lines Merged revisions 113296 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113296 | file | 2008-04-08 12:03:43 -0300 (Tue, 08 Apr 2008) | 4 lines If audio suddenly gets fed into one side of a channel after a lapse of frames flush the other factory so that old audio does not remain in the factory causing the sync code to not execute. (closes issue #12296) Reported by: jvandal ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@113298 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-12Merged revisions 108084 via svnmerge from file1-14/+35
https://origsvn.digium.com/svn/asterisk/trunk ................ r108084 | file | 2008-03-12 15:29:33 -0300 (Wed, 12 Mar 2008) | 12 lines Merged revisions 108083 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108083 | file | 2008-03-12 15:26:37 -0300 (Wed, 12 Mar 2008) | 4 lines Add a trigger mode that triggers on both read and write. The actual function that returns the combined audio frame though will wait until both sides have fed in audio, or until one side stops (such as the case when you call Wait). (closes issue #11945) Reported by: xheliox ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@108085 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-20*mumble*file1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103842 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-20file not found.file1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103840 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-20Minor test...file1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103838 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-13Remove a duplicate lock of the audiohook lock when destroying manipulaterussell1-1/+0
audiohooks. This causes an error when we attempt to destroy the lock later when freeing the audiohook. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98581 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11I am no longer Rockin'file1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98432 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11Testing something...file1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98424 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-30Adding support for the "automixmonitor" dial and queue options.mmichelson1-0/+80
This works in much the same way as the automonitor, except that instead of using the monitor app, it uses the mixmonitor app. By providing an 'x' or 'X' as a dial or queue option, a DTMF sequence may be entered (as defined in features.conf) to start the one-touch mixmonitor. This patch also introduces some new API calls to the audiohooks code for searching for an audiohook by type and for searching for a running audiohook by type. Big thanks to joetester for writing the initial patch, testing it and patiently waiting for it to be committed. (closes issue #10185, reported and patched by xmarksthespot) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90388 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-21remove a bunch of useless #include "options.h"rizzo1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89511 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-19another bunch of include removals (errno.h and asterisk/logger.h)rizzo1-2/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89425 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-16Start untangling header inclusion in a way that does not affectrizzo1-4/+0
build times - tested, there is no measureable difference before and after this commit. In this change: use asterisk/compat.h to include a small set of system headers: inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h, stdlib.h, alloca.h, stdio.h Where available, the inclusion is conditional on HAVE_FOO_H as determined by autoconf. Normally, source files should not include any of the above system headers, and instead use either "asterisk.h" or "asterisk/compat.h" which does it better. For the time being I have left alone second-level directories (main/db1-ast, etc.). git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89333 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-08use %d and cast to int instead of %zd for size_t object,rizzo1-2/+2
this helps portability. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89109 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-08improve linked-list macros in two ways:kpfleming1-14/+8
- the *_CURRENT macros no longer need the list head pointer argument - add AST_LIST_MOVE_CURRENT to encapsulate the remove/add operation when moving entries between lists git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89106 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-06Fix memory issue that crept up with Russell's testing. It is *not* proper to ↵file1-2/+0
free the frame we get in ast_write. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@81858 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-05Doxygen cleanups/fixes.qwell1-0/+2
Closes issue #10654, patch by snuffy git-svn-id: http://svn.digium.com/svn/asterisk/trunk@81560 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-21Minor tweak. Don't manipulate volume of the audio in the buffer if no audio ↵file1-20/+22
is actually there. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@80157 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-08Merge audiohooks branch into trunk. This is a new API for developers to ↵file1-0/+625
listen and manipulate the audio going through a channel. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@78649 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-24 19:17:53 +0000