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2007-12-04Merged revisions 90735 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03 Dec 2007) | 22 lines A big one... This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop. This is accomplished by creating a datastore on the calling channel which has a linked list of all devices dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore is detached from the channel and destroyed. This change also introduces some side effects to the code which I shall enumerate here: 1. Datastore inheritance has been backported from trunk into 1.4 2. A large chunk of code has been removed from app_dial. This chunk is the section of code which handles the call forward case after the channel has been requested but before it has been called. This was removed because call-forwarding still works fine without it, it makes the code less error-prone should it need changing, and it made this set of changes much less painful to just have the forwarding handled in one place in each module. 3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore which is attached to the channel may be created and attached in either app_dial or app_queue, so they need a common place to find the datastore info. This approach was taken in case similar datastores are needed in the future, there will be a common place to add them. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-03Remove the file descriptors from the main poll channel when the channel is ↵Joshua Colp
hung up during the dialing attempt, and make sure a channel exists before trying to remove it at the end. (closes issue #11441) Reported by: blitzrage git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-30Adding support for the "automixmonitor" dial and queue options.Mark Michelson
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: https://origsvn.digium.com/svn/asterisk/trunk@90388 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27Merged revisions 89622 via svnmerge from Steve Murphy
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89622 | murf | 2007-11-26 23:24:02 -0700 (Mon, 26 Nov 2007) | 1 line closes issue #11379; OK, this is an attempt to make both sides happy. To the cdr.conf file, I added the option 'unanswered', which defaults to 'no'. In this mode, you will see a cdr for a call, whether it was answered or not. The disposition will be NO ANSWER or ANSWERED, as appropriate. The src is as you'd expect, the destination channel will be one of the channels from the Dial() call, usually the last in the list if more than one chan was specified. With unanswered set to 'yes', you will still see this cdr entry in both cases. But in the case where the dial timed out, you will also see a cdr for each line attempted, marked NO ANSWER, with no destination channel name. The new option defaults to 'no', so you don't see the pesky extra cdr's by default, and you will not see the irritating 'not posted' messages. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-21closes issue #11285, where an unload of a module that creates a dialplan ↵Steve Murphy
context, causes a crash when you do a 'dialplan show' of that context. This is because the registrar string is defined in the module, and the stale pointer is traversed. The reporter offered a patch that would always strdup the registrar string, which is practical, but I preferred to destroy the created contexts in each module where one is created. That seemed more symmetric. There were only 6 place in asterisk where this is done: chan_sip, chan_iax2, chan_skinny, res_features, app_dial, and app_queue. The two apps destroyed the context, but left the contexts. All is fixed now and unloads should be dialplan friendly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89513 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-21remove another set of redundant #include "asterisk/options.h"Luigi Rizzo
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89512 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-20move asterisk/paths.h outside asterisk.h and into those filesLuigi Rizzo
who really need it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-19include "logger.h" and errno.h from asterisk.h - usage shows that theyLuigi Rizzo
were included almost everywhere. Remove some of the instances. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89424 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-16Start untangling header inclusion in a way that does not affectLuigi Rizzo
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: https://origsvn.digium.com/svn/asterisk/trunk@89333 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-14Use BEGIN_OPTIONS / END_OPTIONS to make the syntax highlighting in my editor ↵Russell Bryant
happy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89264 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-01This commits the performance mods that give the priority processing engine ↵Steve Murphy
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: https://origsvn.digium.com/svn/asterisk/trunk@88166 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11Make sure we propogate ANI2 to the outbound channelMatthew Fredrickson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85499 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-09Remove redundant includes (patch by snuffy) (Closes issue #10922)Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85140 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-01Merged revisions 84166 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84166 | russell | 2007-10-01 09:24:49 -0500 (Mon, 01 Oct 2007) | 2 lines Simplify the CAN_EARLY_BRIDGE macro a bit. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@84167 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-01Merged revisions 84158 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84158 | file | 2007-10-01 10:49:36 -0300 (Mon, 01 Oct 2007) | 4 lines Only attempt early bridging if the options given to Dial() permit it. (closes issue #10861) Reported by: peekyb ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@84159 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-17Make the MALLOC_DEBUG output for free() useful again. After changing calls toRussell Bryant
free to be ast_free, astmm said all calls to free were coming from utils.h git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82628 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-31Merged revisions 81412 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #10621) ........ r81412 | qwell | 2007-08-31 13:44:44 -0500 (Fri, 31 Aug 2007) | 4 lines Re-order dial options to be in line with the existing alpha order. Issue 10621, initial patch by junky ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@81413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-08Add support for using epoll instead of poll. This should increase ↵Joshua Colp
scalability and is done in such a way that we should be able to add support for other poll() replacements. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78683 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
2007-08-01Convert code that checks the _softhangup member of ast_channel directory to useRussell Bryant
the ast_check_hangup() funciton. This function takes scheduled hangups into account. (closes issue #10230, patch by Juggie) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-27These fixes take care of two problems: a complaint in asterisk-dev that ↵Steve Murphy
goto's aren't working in trunk, a side effect of the move to commas as arg seps in apps and funcs; and a problem I spotted myself with dial's 'e' option, where gotos were off by one, because I forgot to set the AUTOLOOP flag in the peer channel. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77520 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
2007-07-23Merge the dialplan_aesthetics branch. Most of this patch simply converts ↵Tilghman Lesher
applications using old methods of parsing arguments to using the standard macros. However, the big change is that the really old way of specifying application and arguments separated by a comma will no longer work (e.g. NoOp,foo|bar). Instead, the way that has been recommended since long before 1.0 will become the only method available (e.g. NoOp(foo,bar). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76703 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
2007-07-17Merged revisions 75405 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r75405 | mmichelson | 2007-07-17 15:03:48 -0500 (Tue, 17 Jul 2007) | 6 lines Fixing an error I made earlier. ast_fileexists can return -1 on failure, so I need to be sure that we only enter the if statement if it is successful. Related to my fix to issue #10186 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75406 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
2007-07-17Fix an incorrect parenthesization (TODO: Find a better word) in app_dialJason Parker
Pointed out by Fanzhou Zhao Closes issue #10216 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75351 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-16Merged revisions 75253 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r75253 | mmichelson | 2007-07-16 13:16:15 -0500 (Mon, 16 Jul 2007) | 8 lines Restoring functionality from 1.2 wherein Retrydial will not exit if there is no announce file specified. This change makes it so that if there is no announce file specified, the application will continue until finished (or caller hangs up). If a bogus announce file is specified, then a warning message will be printed saying that the file could not be found, but execution will still continue. (closes issue #10186, reported by jon, patched by me) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75254 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-16Applications no longer need to call ast_module_user_add and ↵Joshua Colp
ast_module_user_remove. This is now taken care of in the pbx_exec function outside of the application. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-16It is no longer required for each module that deals with a channel to call ↵Joshua Colp
ast_module_user_hangup_all in it's unload function. The loader will automatically perform this action for it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75183 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-09Implementation of a feature that will disable "missed calls" counters on SIP ↵Olle Johansson
phones. If the call is answered by another phone, other phones won't display the call as "missed". You can also add an option to the dial command so that you can have a "followme" scenario and not count the calls as "missed" when you cancel the call. Thanks to Ramon and Frank for feedback on this feature. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@74024 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-03Merged revisions 73053 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r73053 | tilghman | 2007-07-03 07:38:53 -0500 (Tue, 03 Jul 2007) | 10 lines Merged revisions 73052 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r73052 | tilghman | 2007-07-03 07:34:14 -0500 (Tue, 03 Jul 2007) | 2 lines RetryDial should accept a 0 argument, but it does not, because atoi does not distinguish between 0 and error (closes issue #10106) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@73054 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-22Issue 9990 - New API ast_mkdir, which creates parent directories as ↵Tilghman Lesher
necessary (and is faster than an outcall to mkdir -p) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@71040 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-20Cleaning up a small disaster I created earlierSteve Murphy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@70607 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-20As per 9228, now app_queue should have the proper machinery to do gosubs.Steve Murphy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@70531 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-20Merged revisions 70445 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r70445 | tilghman | 2007-06-20 14:29:23 -0500 (Wed, 20 Jun 2007) | 10 lines Merged revisions 70444 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r70444 | tilghman | 2007-06-20 14:25:54 -0500 (Wed, 20 Jun 2007) | 2 lines Issue 9997 - Timelimit times out the wrong channel ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@70446 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-20Merge work to make U(...) option work for DialTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@70358 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-19Via bug9228, no way to create macros via AEL, and some of the apps allow you ↵Steve Murphy
to call macros..., I modded the apps that allow macro calls to allow gosubs calls also, to make them AEL compliant. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@70161 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
2007-06-12Completely remove all of the code related to jumping to priority n + 101. yay!Russell Bryant
(issue #9926, caio1982) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@68970 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-07Merged revisions 68071 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r68071 | file | 2007-06-07 10:21:59 -0400 (Thu, 07 Jun 2007) | 10 lines Merged revisions 68070 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r68070 | file | 2007-06-07 10:19:40 -0400 (Thu, 07 Jun 2007) | 2 lines Allow the 'g' option to work if used with the 'S' option. (issue #9888 reported by gasparz) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@68072 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
2007-06-04Merged revisions 67066 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r67066 | file | 2007-06-04 13:59:14 -0400 (Mon, 04 Jun 2007) | 2 lines Initialize cidname variable to nothing since it may be used without having been touched. (issue #9661 reported by dimas) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-18Merged revisions 65200 via svnmerge from Steve Murphy
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r65200 | murf | 2007-05-18 16:06:27 -0600 (Fri, 18 May 2007) | 9 lines Merged revisions 65172 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r65172 | murf | 2007-05-18 14:56:20 -0600 (Fri, 18 May 2007) | 1 line This update will fix the situation that occurs as described by 9717, where when several targets are specified for a dial, if any one them reports FAIL, the whole call gets FAIL, even though others were ringing OK. I rearranged the priorities, so that a new disposition, NULL, is at the lowest level, and the disposition get init'd to NULL. Then, next up is FAIL, and next up is BUSY, then NOANSWER, then ANSWERED. All the related set routines will only do so if the disposition value to be set to is greater than what's already there. This gives the intended effect. So, if all the targets are busy, you'd get BUSY for the call disposition. If all get BUSY, but one, and that one rings is not answered, you get NOANSWER. If by some freak of nature, the NULL value doesn't get overridden, then the disp2str routine will report NOANSWER as before. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65202 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-17Merged revisions 64756 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64756 | russell | 2007-05-17 11:47:29 -0500 (Thu, 17 May 2007) | 3 lines Increase the size of a buffer to support longer dial strings for channels. (issue #9291, reported and fix suggested by meni) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64757 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-13Merged revisions 61656 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61656 | file | 2007-04-13 15:17:08 -0400 (Fri, 13 Apr 2007) | 10 lines Merged revisions 61655 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61655 | file | 2007-04-13 15:15:12 -0400 (Fri, 13 Apr 2007) | 2 lines Add OUTBOUND_GROUP_ONCE variable to app_dial. This behaves the same as OUTBOUND_GROUP except it will get unset after use so it won't get accidentally inherited. (issue #BE-140) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61657 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-09Remove unused instances of unnamed enums.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61116 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-09Merged revisions 60798 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r60798 | file | 2007-04-08 21:03:14 -0400 (Sun, 08 Apr 2007) | 10 lines Merged revisions 60797 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r60797 | file | 2007-04-08 20:59:29 -0400 (Sun, 08 Apr 2007) | 2 lines When calling a device that then forwards us elsewhere... we have to make our channels compatible if it is the only channel being dialed. (issue #9445 reported by marcelbarbulescu) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60799 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-27Properly hangup the original dialed channel, not the new channel that ↵Joshua Colp
appeared from the forwarding. (issue #9161 reported by PhilSmith) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57011 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-17Merged revisions 55154 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r55154 | file | 2007-02-16 22:55:30 -0500 (Fri, 16 Feb 2007) | 10 lines Merged revisions 55153 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r55153 | file | 2007-02-16 22:53:45 -0500 (Fri, 16 Feb 2007) | 2 lines Answer the channel before recording privacy information. (issue #8926 reported by lmamane) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55155 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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