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2010-07-09Fix compile error.russell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275309 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-09Include rdnis in msgXXXX.txt file.pabelanger1-0/+2
(closes issue #17566) Reported by: outcast Patches: voicemail-rdnis.patch uploaded by outcast (license 1071) Tested by: outcast git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275307 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-09Weird, no output and Bamboo still fails...tilghman1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275227 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-09Add some diagnostic feedback to our data teststilghman1-0/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275172 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-09Kill some startup warnings and errors and make some messages more helpful in ↵tilghman2-5/+5
tracking down the source. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275105 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-09Merged revisions 275027 via svnmerge from mnicholson1-0/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275027 | mnicholson | 2010-07-09 11:04:21 -0500 (Fri, 09 Jul 2010) | 8 lines Clear the AST_CDR_FLAG_DIALED flag for channels going into the pbx via the G option in app_dial (closes issue #17592) Reported by: jamicque Patches: G-flag-cdr-fix1.diff uploaded by mnicholson (license 96) Tested by: jamicque, mnicholson ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275028 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-08Add IPv6 to Asterisk.mmichelson1-3/+5
This adds a generic API for accommodating IPv6 and IPv4 addresses within Asterisk. While many files have been updated to make use of the API, chan_sip and the RTP code are the files which actually support IPv6 addresses at the time of this commit. The way has been paved for easier upgrading for other files in the near future, though. Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne for their hard work on this. (closes issue #17565) Reported by: russell Patches: asteriskv6-test-report.pdf uploaded by russell (license 2) Review: https://reviewboard.asterisk.org/r/743 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274783 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-08Implement AstData API data providers as part of the GSOC 2010 project,eliel3-132/+244
midterm evaluation. Review: https://reviewboard.asterisk.org/r/757/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274727 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-07Also run the externnotify script when the pollmailboxes thread notices a change.tilghman1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274491 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-02The switch fallthrough could create some errorneous situations, so best to ↵tilghman1-0/+5
force directly to the default case. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@273714 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-02Fix various typos reported by Lintiantzafrir3-13/+13
(Also fix the typos in the comments) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@273641 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-01Merged revisions 273474 via svnmerge from jpeeler1-2/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273474 | jpeeler | 2010-07-01 15:19:16 -0500 (Thu, 01 Jul 2010) | 14 lines Allow admin user to join conference without using admin mode and no user pin. Configuring the conference in meetme.conf like the following: conf => 2345,,6666 did not prompt for pin when used without admin mode. This meant that the conference could not be joined as an admin even if the user knew the correct pin. The original bug report was submitted claiming that the blank user pin should deny entry into the conference. I think a better way to handle this would be with a feature enhancement that used the following syntax: conf => 2345,X,6666 - where X denotes no acceptable pin allowed (closes issue #15704) Reported by: modelnine ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@273522 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-01Merged revisions 273354 via svnmerge from jpeeler1-0/+12
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273354 | jpeeler | 2010-07-01 10:05:43 -0500 (Thu, 01 Jul 2010) | 12 lines Ensure channel placed in meetme in ringing state is properly hung up. An outgoing channel placed in meetme while still ringing which was then hung up would not exit meetme and the channel was not properly destroyed. Specifically checking for this scenario by looking at the appropriate control frames resolves the issue. (closes issue #15871) Reported by: Ivan Patches: meetme_congestion_trunk_v2.patch uploaded by Ivan (license 229) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@273355 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-23Merged revisions 272367 via svnmerge from mnicholson1-1/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 This version of the patch only adds AgentComplete for attended transfers. It was already present for blind transfers. ........ r272367 | mnicholson | 2010-06-23 17:33:51 -0500 (Wed, 23 Jun 2010) | 8 lines Send AgentComplete manager events in the event of blind and attended transfers. (closes issue #16819) Reported by: elbriga Patches: app_queue.diff uploaded by elbriga (license 482) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272368 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-23Fix previous merge. ast_test_flag != ast_test_flag64pabelanger1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272259 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-23Merged revisions 272255 via svnmerge from pabelanger1-3/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r272255 | pabelanger | 2010-06-23 16:57:01 -0400 (Wed, 23 Jun 2010) | 12 lines First caller into a dynamic conference now enter pin once. If MeetMe is configured to use dynamic conference numbers, then the first caller (which creates the conference) had to enter the PIN number twice. (closes issue #15878) Reported by: shawkris Patches: issue15878.patch uploaded by pabelanger (license 224) Tested by: pabelanger ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272257 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-23Don't start the sla thread unless we realy need ittwilson1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272146 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-23Make sure reload updates SLA configtwilson1-2/+19
Even if there are no stations or trunks defined, we need to start the sla thread to make sure we get the reload event. Also, when doing a reload we need to remove the existing trunks and stations or they end up hanging around. (closes issue #16818) Reported by: mbonin Patches: sla_reload.patch uploaded by twilson (license 396) Tested by: twilson git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272109 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-21Add new application for declining counting words in multiple languages.tilghman1-0/+202
(closes issue #16869) Reported by: chappell Patches: app_say_counted-20100317.c uploaded by chappell (license 8) Tested by: chappell git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271520 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-17option w[(secs)] incorrectly capitalized in xmldocpabelanger1-1/+1
(closes issue #17516) Reported by: karlfife git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271089 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-08Don't pass null to manager_event()mnicholson1-2/+2
(closes issue #17087) Reported by: bklang Patches: app-fax-null-sprintf1.diff uploaded by mnicholson (license 96) Tested by: bklang git-svn-id: http://svn.digium.com/svn/asterisk/trunk@269083 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-08Fix some doxygen warnings.lmadsen2-6/+19
(closes issue #17336) Reported by: snuffy Patches: doxygen-fixes1.diff uploaded by snuffy (license 35) Tested by: russell git-svn-id: http://svn.digium.com/svn/asterisk/trunk@268969 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Generic Advice of Charge.rmudgett2-0/+60
Asterisk Generic AOC Representation - Generic AOC encode/decode routines. (Generic AOC must be encoded to be passed on the wire in the AST_CONTROL_AOC frame) - AST_CONTROL_AOC frame type to represent generic encoded AOC data - Manager events for AOC-S, AOC-D, and AOC-E messages Asterisk App Support - app_dial AOC-S pass-through support on call setup - app_queue AOC-S pass-through support on call setup AOC Unit Tests - AOC Unit Tests for encode/decode routines - AOC Unit Test for manager event representation. SIP AOC Support - Pass-through of generic AOC-D and AOC-E messages to snom phones via the snom AOC specification. - Creation of chan_sip page3 flags for the addition of the new 'snom_aoc_enabled' sip.conf option. IAX AOC Support - Natively supports AOC pass-through through the use of the new AST_CONTROL_AOC frame type DAHDI AOC Support - ETSI PRI full AOC Pass-through support - 'aoc_enable' chan_dahdi.conf option for independently enabling pass-through of AOC-S, AOC-D, AOC-E. - 'aoce_delayhangup' option for retrieving AOC-E on disconnect. - DAHDI A() dial string option for requesting AOC services. example usage: ;requests AOC-S, AOC-D, and AOC-E on call setup exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e)) Review: https://reviewboard.asterisk.org/r/552/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267096 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Silence a compiler warning.russell1-1/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267093 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-01Support setting locale per-mailbox (changes date/time languages for email, ↵tilghman1-7/+15
pager messages). (closes issue #14333) Reported by: klaus3000 Patches: 20090515__issue14333.diff.txt uploaded by tilghman (license 14) app_voicemail.c-svn-trunk-rev211675-patch.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@266828 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-01Set app and appdata fields when a Dial is redirectedtwilson1-0/+2
(closes issue #17204) Reported by: one47 Tested by: twilson, one47 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@266786 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-26Remove redundant ast_conntected_line_free call.mmichelson1-1/+0
This wouldn't cause any problems, but it's certainly not needed either. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@266098 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-25Merged revisions 265610 via svnmerge from mnicholson1-7/+0
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r265610 | mnicholson | 2010-05-25 11:48:19 -0500 (Tue, 25 May 2010) | 8 lines Don't mark the cdr records of unanswered queue calls with "NOANSWER". This restores the behavior prior to r258670. (closes issue #17334) Reported by: jvandal Patches: queue-cdr-fixes1.diff uploaded by mnicholson (license 96) Tested by: aragon, jvandal ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@265611 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-24Allow SendDTMF to play digits to a specified channel.mmichelson1-2/+18
Patch supplied by reporter was modified to use autoservice and prevent a potential channel ref leak but is otherwise as the reporter uploaded it. (closes issue #17182) Reported by: rcasas Patches: app_senddtmf.c.patch_trunk uploaded by rcasas (license 641) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@265453 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-24Make app_rpt.c able to compile again.rmudgett1-35/+35
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@265367 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-21Merged revisions 265089 via svnmerge from mmichelson1-0/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r265089 | mmichelson | 2010-05-21 15:59:14 -0500 (Fri, 21 May 2010) | 8 lines Don't hang up on a queue caller if the file we attempt to play does not exist. This also fixes a documentation mistake in file.h that made my original attempt to correct this problem not work correctly. (closes issue #17061) Reported by: RoadKill ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@265090 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-20Error message fix.tilghman1-1/+1
(closes issue #17356) Reported by: kenner Patches: app_stack.c.diff uploaded by kenner (license 1040) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@264752 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-20Dial and queue connected line update macro not always run when expected.rmudgett2-43/+66
The connected line update macro would not get run if the connected line number string was empty. The number could be empty if the connected line update did not update a number but the name. It should be run if there was an AST_CONTROL_CONNECTED_LINE frame received for pending dials and queues. Renamed and added some more comments for some confusing identifiers directly connected to the related code. Also fixed a memory leak in app_queue. Review: https://reviewboard.asterisk.org/r/669/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@264669 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-19Merged revisions 264334 via svnmerge from mnicholson1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r264334 | mnicholson | 2010-05-19 15:01:38 -0500 (Wed, 19 May 2010) | 5 lines Set quieted flag when receiving a dtmf tone during playback in speechbackground. (closes issue #16966) Reported by: asackheim ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@264335 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-18Merged revisions 263769 via svnmerge from jpeeler1-14/+22
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r263769 | jpeeler | 2010-05-18 13:54:58 -0500 (Tue, 18 May 2010) | 10 lines Modify directory name reading to be interrupted with operator or pound escape. In the case of accidentally entering the wrong first three letters for the reading, users could be very frustrated if the name listing is very long. This allows interrupting the reading by pressing 0 or #. 0 will attempt to execute a configured operator (o) extension and # will exit and proceed in the dialplan. ABE-2200 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@263807 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-17With IMAP backend, messages in INBOX were counted twice for MWI.tilghman1-20/+26
(closes issue #17135) Reported by: edhorton Patches: 20100513__issue17135.diff.txt uploaded by tilghman (license 14) 17135_2.diff uploaded by ebroad (license 878) Tested by: edhorton, ebroad git-svn-id: http://svn.digium.com/svn/asterisk/trunk@263589 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-17Enhancements to connected line and redirecting work.mmichelson2-8/+80
From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@263541 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-12Merged revisions 262662 via svnmerge from dvossel1-5/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r262662 | dvossel | 2010-05-12 12:00:04 -0500 (Wed, 12 May 2010) | 11 lines fixes app_meetme dsp error We attempted to detect silence after translating a frame from signed linear. This caused a flooding of errors. To resolve this the code to detect silence was moved before the translation. (closes issue #17133) Reported by: jsdyer ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@262744 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-12Ensure the arguments are initialized. Also miscellaneous CG cleanup.tilghman1-30/+39
(closes issue #16576) Reported by: uxbod Patches: 20100505__issue16576.diff.txt uploaded by tilghman (license 14) Tested by: uxbod git-svn-id: http://svn.digium.com/svn/asterisk/trunk@262656 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-11Merged revisions 262321 via svnmerge from tilghman1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r262321 | tilghman | 2010-05-11 12:22:07 -0500 (Tue, 11 May 2010) | 2 lines Fix issue #17302 a slightly different way (mad props to Qwell) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@262330 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-10fixes PickupChan applicationdvossel1-2/+2
(closes issue #16863) Reported by: schern Patches: app_directed_pickup.c.patch uploaded by schern (license 995) for_trunk.diff uploaded by cjacobsen (license 1029) Tested by: Graber, cjacobsen, lathama, rickead2000, dvossel git-svn-id: http://svn.digium.com/svn/asterisk/trunk@262240 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-07VoicemailMain and VMauthenticate, allow escape to the 'a' extension when a ↵alecdavis1-0/+44
single '*' is entered Where a site uses VoicemailMain(mailbox) the users have to be at their own extension to clear their voicemail, they have no way of escaping VoicemailMain to allow entry of new boxnumber. This patch, allows a site to include to 'a' priority in the VoicemailMain context, to allow an escape. If the 'a' priority doesn't exist in the context that VoicemailMain was called from then it acts as the old behaviour. Reported by: alecdavis Tested by: alecdavis Patch vm_a_extension.diff2.txt uploaded by alecdavis (license 585) Review: https://reviewboard.asterisk.org/r/489/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@262005 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-06Merged revisions 261735 via svnmerge from jpeeler1-1/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r261735 | jpeeler | 2010-05-06 15:10:59 -0500 (Thu, 06 May 2010) | 8 lines Only allow the operator key to be accepted after leaving a voicemail. Or rather disallow the operator key from being accepted when not offered, such as after finishing a recording from within the mailbox options menu. ABE-2121 SWP-1267 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261736 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-05'queue reset stats' erroneously clears wrapuptime configuration.pabelanger1-1/+1
Resets each member's lastcall to 0 now. (closes issue #17262) Reported by: rain Patches: wrapuptime_reset_fix.diff uploaded by rain (license 327) Tested by: rain git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261232 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-04Add new possible value to autopause option to allow members to be autopaused ↵mmichelson1-5/+54
in all queues. See the CHANGES file and queues.conf.sample for more details. (closes issue #17008) Reported by: jlpedrosa Patches: queues.autopause_en_review.diff uploaded by jlpedrosa (license 1002) Review: https://reviewboard.asterisk.org/r/581/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261051 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-04Merged revisions 260923 via svnmerge from jpeeler1-7/+8
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r260923 | jpeeler | 2010-05-04 13:46:46 -0500 (Tue, 04 May 2010) | 12 lines Voicemail transfer to operator should occur immediately, not after main menu. There were two scenarios in the advanced options that while using the operator=yes and review=yes options, the transfer occurred only after exiting the main menu (after sending a reply or leaving a message for an extension). Now after the audio is processed for the reply or message the transfer occurs immediately as expected. ABE-2107 ABE-2108 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@260924 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-03Add new admin features to meetme: Roll call, eject all, mute all, record ↵jpeeler1-3/+158
in-conf This patch adds the following in-conference admin DTMF features: *81 - Roll call (or simply user count if INTROUSER isn't enabled) *82 - Eject all non-admins *83 - Mute/unmute all non-admins *84 - Start recording the conference on the fly FWIW, this code uses newly recorded prompts. (closes issue #16379) Reported by: rfinnie Patches: meetme-enhancements-232771-v1.patch uploaded by rfinnie (license 940) modified slightly by me git-svn-id: http://svn.digium.com/svn/asterisk/trunk@260757 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-30Fix logic reversal error when queue callers join the queue.mmichelson1-1/+1
When a specific position is specified for the queue, the idea was that the caller cannot be placed ahead of higher-priority callers. Unfortunately, the logic was reversed so that the caller could ONLY be placed ahead of higher priority callers. Discovered while writing a unit test. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@260344 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-28Merged revisions 259664 via svnmerge from jpeeler1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r259664 | jpeeler | 2010-04-28 12:13:29 -0500 (Wed, 28 Apr 2010) | 4 lines Do not play goodbye prompt after timeout of message review. ABE-2124 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@259672 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-22Pass interactive = 0 and fix a compile error.eliel1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258595 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-17 15:16:40 +0000