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2010-04-16Merged revisions 257713 via svnmerge from dhubbard1-5/+6
https://origsvn.digium.com/svn/asterisk/trunk ................ r257713 | dhubbard | 2010-04-16 16:22:30 -0500 (Fri, 16 Apr 2010) | 28 lines Merged revisions 257686 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257686 | dhubbard | 2010-04-16 16:15:43 -0500 (Fri, 16 Apr 2010) | 21 lines Make the mixmonitor thread process audio frames faster Mantis issue 17078 reports MixMonitor recordings have shorter durations than the call duration. This was because the mixmonitor thread was not processing frames from the audiohook fast enough. The mixmonitor thread would slowly fall behind the most recent audio frame and when the channel hangs up, the mixmonitor thread would exit without processing the same number of frames as the channel; leaving the mixmonitor recording shorter than actual call duration. This revision fixes this issue by moving the ast_audiohook_trigger_wait() and the subsequent audiohook.status check into the block where the ast_audiohook_read_frame() function returns NULL. (closes issue #17078) Reported by: geoff2010 Patches: dw-M17078.patch uploaded by dhubbard (license 733) Tested by: dhubbard, geoff2010 Review: https://reviewboard.asterisk.org/r/611/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@257740 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-05Merged revisions 237656 via svnmerge from mvanbaak1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r237656 | mvanbaak | 2010-01-05 17:08:12 +0100 (Tue, 05 Jan 2010) | 6 lines Make CLI command 'mixmonitor start|stop <channel> work again. (closes issue #16534) Reported by: jlaguilar Fix as suggested by jlaguilar in the bugreport ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@237657 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-19Merged revisions 230509 via svnmerge from dvossel1-23/+70
https://origsvn.digium.com/svn/asterisk/trunk ................ r230509 | dvossel | 2009-11-19 15:26:21 -0600 (Thu, 19 Nov 2009) | 17 lines Merged revisions 230508 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r230508 | dvossel | 2009-11-19 15:22:46 -0600 (Thu, 19 Nov 2009) | 10 lines fixes MixMonitor thread not exiting when StopMixMonitor is used (closes issue #16152) Reported by: AlexMS Patches: stopmixmonitor_1.4.diff uploaded by dvossel (license 671) Tested by: dvossel, AlexMS Review: https://reviewboard.asterisk.org/r/424/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@230510 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-19Merged revisions 213113 via svnmerge from dvossel1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r213113 | dvossel | 2009-08-19 16:21:00 -0500 (Wed, 19 Aug 2009) | 14 lines Merged revisions 213103 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r213103 | dvossel | 2009-08-19 16:18:37 -0500 (Wed, 19 Aug 2009) | 8 lines Fixes memory leak caused by incorrectly freeing mixmonitor (closes issue #15699) Reported by: edantie Patches: mixmonitor.patch uploaded by edantie (license 862) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@213128 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-10AST-2009-005tilghman1-3/+3
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@211580 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-17Merged revisions 201445 via svnmerge from dvossel1-23/+59
https://origsvn.digium.com/svn/asterisk/trunk ................ r201445 | dvossel | 2009-06-17 14:45:35 -0500 (Wed, 17 Jun 2009) | 25 lines Merged revisions 201423 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201423 | dvossel | 2009-06-17 14:28:12 -0500 (Wed, 17 Jun 2009) | 19 lines StopMixMonitor race condition (not giving up file immediately) StopMixMonitor only indicates to the MixMonitor thread to stop writing to the file. It does not guarantee that the recording's file handle is available to the dialplan immediately after execution. This results in a race condition. To resolve this, the filestream pointer is placed in a datastore on the channel. When StopMixMonitor is called, the datastore is retrieved from the channel and the filestream is closed immediately before returning to the dialplan. Documentation indicating the use of StopMixMonitor to free files has been updated as well. (closes issue #15259) Reported by: travisghansen Tested by: dvossel Review: https://reviewboard.asterisk.org/r/283/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@201447 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-16Merged revisions 201056 via svnmerge from kpfleming1-4/+9
https://origsvn.digium.com/svn/asterisk/trunk ................ r201056 | kpfleming | 2009-06-16 13:54:30 -0500 (Tue, 16 Jun 2009) | 18 lines Merged revisions 200991 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun 2009) | 11 lines Improve support for media paths that can generate multiple frames at once. There are various media paths in Asterisk (codec translators and UDPTL, primarily) that can generate more than one frame to be generated when the application calling them expects only a single frame. This patch addresses a number of those cases, at least the primary ones to solve the known problems. In addition it removes the broken TRACE_FRAMES support, fixes a number of bugs in various frame-related API functions, and cleans up various code paths affected by these changes. https://reviewboard.asterisk.org/r/175/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@201097 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-28Merged revisions 197828 via svnmerge from lmadsen1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r197828 | lmadsen | 2009-05-28 18:04:00 -0400 (Thu, 28 May 2009) | 8 lines Update documentation in MixMonitor. Updated the MixMonitor documentation for the 'b' option so that it is more obvious that you must not optimize away the Local channel when using this option. (closes issue #14829) Reported by: licedey Tested by: mmichelson, licedey, lmadsen ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@197894 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-05Merged revisions 173592 via svnmerge from mmichelson1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173592 | mmichelson | 2009-02-05 12:47:24 -0600 (Thu, 05 Feb 2009) | 3 lines Add some missing cleanup to app_mixmonitor ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@173593 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-05Merged revisions 173559 via svnmerge from mmichelson1-5/+88
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173559 | mmichelson | 2009-02-05 11:34:33 -0600 (Thu, 05 Feb 2009) | 25 lines Fix a problem where a channel pointer becomes invalid due to masquerading or hanging up. app_mixmonitor runs its own thread to monitor the channel's activity and write the mixed audio to a file. Since this thread runs independently of the channel, it is possible that the mixmonitor thread's channel pointer will point to freed memory when the channel either is masqueraded or hangs up (technically, both cases are hangups, but we need to handle the cases slightly differently). The solution for this is to employ a datastore, which has the nice benefit of allowing us to hook into channel masquerades and hangups and update our pointer as necessary. If this looks familiar, this same technique is employed in app_chanspy. app_chanspy is a bit more involved since it does a lot more operations on the channel that is being spied upon. app_mixmonitor does have an extra touch that app_chanspy doesn't have, though. Since there is a thread race between the channel's thread and the mixmonitor thread on a hangup, we em- ploy a condition-and-boolean combination to ensure that the channel thread finishes with our structure before the mixmonitor thread attempts to free it. No crashes! (closes issue #14374) Reported by: aragon Patches: 14374.patch uploaded by putnopvut (license 60) Tested by: aragon, putnopvut ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@173589 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-05Janitor, use ARRAY_LEN() when possible.eliel1-2/+2
(closes issue #13990) Reported by: eliel Patches: array_len.diff uploaded by eliel (license 64) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@161218 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-05Add more [see also] references based on TFOT.eliel1-0/+9
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@154798 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-01Merge changes from team/group/appdocsxmlrussell1-29/+69
This commit introduces the first phase of an effort to manage documentation of the interfaces in Asterisk in an XML format. Currently, a new format is available for applications and dialplan functions. A good number of conversions to the new format are also included. For more information, see the following message to asterisk-dev: http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153365 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-21Default file modes should always be full read and write, to allow the systemtilghman1-1/+1
administrator to make the decision of what permissions will actually be given, through the use of the process umask. (Closes issue# 13751) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@151371 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-28Merge the cli_cleanup branch.mvanbaak1-1/+1
This work is done by lmadsen, junky and mvanbaak during AstriDevCon. This is the second audit the CLI got, and this time lmadsen made sure he had _ALL_ modules loaded that have CLI commands in them. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@145121 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-12Merged revisions 108083 via svnmerge from file1-1/+1
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/trunk@108084 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-14Convert ast_verbose to ast_verb.tilghman1-6/+3
Reported by: snuffy Patch by: snuffy (Closes issue #11547) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@92913 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-12Conversions of free to ast_free, where applicable, and several other ↵tilghman1-1/+1
formatting fixes. Reported by: eliel Patch by: eliel,tilghman (Closes issue #11209) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@92594 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26Merged revisions 89587 via svnmerge from file1-3/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89587 | file | 2007-11-26 13:20:58 -0400 (Mon, 26 Nov 2007) | 6 lines Close the audio file before sending it to the post processing application. (closes issue #11357) Reported by: reformed Patches: mixmonitor.patch uploaded by reformed (license 330) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89589 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-22shuffle a little bit the content of header files to reduce dependencies.rizzo1-0/+1
In this commit: - move the ast_register/unregister_app functions to module.h to avoid the need to include pbx.h for the simpler apps; - move the ast_group structure to channel.h to remove the dependency of app.h on linkedlists.h Note, this is a long process that I am doing in small steps. The main difficulty is that now for each subsystem we have a single header (e.g. channel.h) included by the subsystem provider (usually one file, e.g. channel.c) and by its clients (dozens of them, e.g. we have some 70+ apps and 30+ functions). This requires the clients to include all the extra headers required by the provider (eg. lock.h, linkedlists.h, definitions of substructures...) even though many of the clients would be just happy with opaque struct declarations and function prototypes. The long term plan is to eventually rectify this structure so that the compilation can become faster, and also APIs are more stable. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89522 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-22more removal of redundant headersrizzo1-4/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89519 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-21remove another set of redundant #include "asterisk/options.h"rizzo1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89512 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-20move asterisk/paths.h outside asterisk.h and into those filesrizzo1-0/+1
who really need it. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89466 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-19include "logger.h" and errno.h from asterisk.h - usage shows that theyrizzo1-1/+0
were included almost everywhere. Remove some of the instances. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89424 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-16Start untangling header inclusion in a way that does not affectrizzo1-5/+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
2007-11-13Merged revisions 89241 via svnmerge from mmichelson1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89241 | mmichelson | 2007-11-13 10:02:02 -0600 (Tue, 13 Nov 2007) | 5 lines Reverting commit made in revision 89205 since it is unnecessary. Thanks to Kevin for pointing this out ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89242 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-13There is the potential to copy uninitialized memory into the ↵mmichelson1-1/+1
mixmonitor->post_process string. This fix prevents that. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89207 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-13Merged revisions 89205 via svnmerge from mmichelson1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89205 | mmichelson | 2007-11-12 18:56:46 -0600 (Mon, 12 Nov 2007) | 5 lines Some sanity checking for MixMonitor. If only 1 argument is given, then the args.options and args.post_process strings are uninitialized and could contain garbage. This change handles this situation properly by only using arguments that we have parsed. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89206 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-12Based on a note in asterisk-dev by Brian Capouch, I determined I too ↵murf1-0/+1
agressive in not initializing arrays passed to pbx_substitute_variables_xxxx; I reviewed the code (again) and hopefully found every possible spot where substitute_variables is called conditionally, and made sure the char array involved was set to a null string. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89186 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06"show application <foo>" changes for clarity.mmichelson1-2/+2
(closes issue #11171, reported and patched by blitzrage) Many thanks! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89044 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-01This commits the performance mods that give the priority processing engine ↵murf1-2/+1
in the pbx, a 25-30% speed boost. The two updates used, are, first, to merge the ast_exists_extension() and the ast_spawn_extension() where they are called sequentially in a loop in the code, into a slightly upgraded version of ast_spawn_extension(), with a few extra args; and, second, I modified the substitute_variables_helper_full, so it zeroes out the byte after the evaluated string instead of demanding you pre-zero the buffer; I also went thru the code and removed the code that zeroed this buffer before every call to the substitute_variables_helper_full. The first fix provides about a 9% speedup, and the second the rest. These figures come from the 'PIPS' benchmark I describe in blogs, conf. reports, etc. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88166 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-31Add volume adjustment in.file1-0/+5
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@87851 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-31Restore operation of the option that only writes when the channel is bridged.file1-17/+21
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@87850 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-22Switch from AST_CLI (formerly NEW_CLI) to AST_CLI_DEFINE, since the former ↵qwell1-1/+1
didn't make much sense git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86820 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-19Convert NEW_CLI to AST_CLI.qwell1-1/+1
Closes issue #11039, as suggested by seanbright. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86536 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-11Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :)russell1-25/+23
(closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85460 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-08Merge audiohooks branch into trunk. This is a new API for developers to ↵file1-80/+55
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
2007-07-31Mostly cleanup of documentation to substitute the pipe with the comma, but a ↵tilghman1-1/+1
few other formatting cleanups, too. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@77808 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-19After some study, thought, comparing, etc. I've backed out the previous ↵murf1-2/+2
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: http://svn.digium.com/svn/asterisk/trunk@75983 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-17via 10206, I have added an option (e) to Dial to allow the h exten to get ↵murf1-2/+2
run on peer. Had to upgrade ast_flag stuff to 64 bits to do this. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75400 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-16Applications no longer need to call ast_module_user_add and ↵file1-13/+0
ast_module_user_remove. This is now taken care of in the pbx_exec function outside of the application. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75200 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-16It is no longer required for each module that deals with a channel to call ↵file1-2/+0
ast_module_user_hangup_all in it's unload function. The loader will automatically perform this action for it. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75183 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-27Merged revisions 72381 via svnmerge from file1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r72381 | file | 2007-06-27 19:25:12 -0400 (Wed, 27 Jun 2007) | 10 lines Merged revisions 72378 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r72378 | file | 2007-06-27 19:24:01 -0400 (Wed, 27 Jun 2007) | 2 lines Update documentation to clarify variable usage with MixMonitor. (issue #9494 reported by netoguy) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@72382 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-24Issue 9970 - Ensure directory exists before trying to write an output filetilghman1-1/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@71268 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-06Issue 9869 - replace malloc and memset with ast_calloc, and other coding ↵tilghman1-3/+3
guidelines changes git-svn-id: http://svn.digium.com/svn/asterisk/trunk@67864 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-04Add support for autocompleting start/stop options of the mixmonitor CLI ↵file1-0/+5
command. (issue #9862 reported by eliel) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@66998 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-24Add a new API call for creating detached threads. Then, go replace all of therussell1-5/+1
places in the code where the same block of code for creating detached threads was replicated. (patch from bbryant) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@65968 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-29Merged revisions 52717 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r52717 | file | 2007-01-29 18:43:40 -0500 (Mon, 29 Jan 2007) | 10 lines Merged revisions 52716 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r52716 | file | 2007-01-29 18:39:39 -0500 (Mon, 29 Jan 2007) | 2 lines Now that filename is part of the structure and since it comes before postprocess... we have to add it to our postprocess line. (reported on asterisk-dev by Boris Bakchiev) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@52718 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-25Merged revisions 52163 via svnmerge from file1-1/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r52163 | file | 2007-01-24 20:51:35 -0500 (Wed, 24 Jan 2007) | 10 lines Merged revisions 52162 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r52162 | file | 2007-01-24 20:48:52 -0500 (Wed, 24 Jan 2007) | 2 lines Add another note about audio files being played back to each bridged party. (issue #8718 reported by ppyy) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@52168 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-22Merged revisions 51407 via svnmerge from file1-27/+32
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r51407 | file | 2007-01-22 14:13:44 -0500 (Mon, 22 Jan 2007) | 10 lines Merged revisions 51406 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51406 | file | 2007-01-22 14:08:52 -0500 (Mon, 22 Jan 2007) | 2 lines Move filestream creation to Mixmonitor loop. This will prevent a blank file from being created if no frames ever pass through to be recorded. (issue #7589 reported by steve_mcneil) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51408 f38db490-d61c-443f-a65b-d21fe96a405b
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