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2010-08-10Merged revisions 281567 via svnmerge from russell2-2/+9
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r281567 | russell | 2010-08-10 12:47:13 -0500 (Tue, 10 Aug 2010) | 15 lines Merged revisions 281566 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r281566 | russell | 2010-08-10 12:45:45 -0500 (Tue, 10 Aug 2010) | 8 lines Reset visible indication after answer. (closes issue #17641) Reported by: klaus3000 Patches: ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by klaus3000 (license 65) Tested by: schmidts ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@281568 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-10Fixed the issue caused by EXTEN including user parameters.transnexus1-1/+7
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@281497 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-02Merged revisions 280671 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r280671 | tilghman | 2010-08-02 16:26:11 -0500 (Mon, 02 Aug 2010) | 2 lines Allow the pipe, but also allow the comma ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@280672 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-29Merged revisions 280345 via svnmerge from jeang1-0/+3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r280345 | jeang | 2010-07-29 11:01:35 -0500 (Thu, 29 Jul 2010) | 10 lines Merged revisions 280341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r280341 | jeang | 2010-07-29 10:52:31 -0500 (Thu, 29 Jul 2010) | 2 lines Fix a dsp structure leak occuring when a local channel is put into a meetme conference, then masquaraded away. ABE-2422 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@280346 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-28Merged revisions 280160 via svnmerge from seanbright1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r280160 | seanbright | 2010-07-28 12:51:11 -0400 (Wed, 28 Jul 2010) | 8 lines Plug a reference leak in app_queue when adding members dynamically. (closes issue #17738) Reported by: bobwienholt Patches: issue17738.patch uploaded by bobwienholt (license 950) Tested by: bobwienholt, seanbright ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@280161 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-23Merged revisions 279207 via svnmerge from rmudgett2-6/+12
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r279207 | rmudgett | 2010-07-23 17:11:23 -0500 (Fri, 23 Jul 2010) | 14 lines Merged revisions 279206 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r279206 | rmudgett | 2010-07-23 16:56:44 -0500 (Fri, 23 Jul 2010) | 7 lines SIP promiscuous redirect could fail to dial the redirect. The ast_channel was created with one variable to ast_request() but the call to ast_call() that initiates the outgoing call was using a different variable. The two variables are not equivalent if the call_forward string included a channel technology specifier. e.g., SIP/200 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@279227 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-21Ensure realtime conferences are treated the same as static conferences when ↵tilghman1-7/+52
trying to find an empty one. Also, parse the useropts properly, when retrieving from realtime, and add them to the existing flags. (closes issue #17502) Reported by: kenji Patches: 20100720__issue17502.diff.txt uploaded by tilghman (license 14) Tested by: kenji git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278463 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-20Merged revisions 278261 via svnmerge from tilghman1-38/+44
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r278261 | tilghman | 2010-07-20 17:23:13 -0500 (Tue, 20 Jul 2010) | 7 lines Delete IMAP messages in reverse order, to ensure reordering after each expunge does not cause deletion of the wrong message. (closes issue #16350) Reported by: noahisaac Patches: 20100623__issue16350.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278275 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-20Add load priority order, such that preload becomes unnecessary in most casestilghman3-3/+9
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278132 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Fix reporting estimated queue hold time.jpeeler1-1/+1
Just say the number of seconds (after minutes) rather than doing some incorrect calculation with respect to minutes. (closes issue #17498) Reported by: corruptor Patches: holdesecs_bug.diff uploaded by corruptor (license 253) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277488 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Add missing handling for ringing state for use with queue empty options.jpeeler1-0/+5
(closes issue #17471) Reported by: jazzy Patches: app_queue.c.diff uploaded by jazzy (license 1056) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277366 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Merged revisions 277182 via svnmerge from pabelanger1-3/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277182 | pabelanger | 2010-07-16 13:10:36 -0400 (Fri, 16 Jul 2010) | 8 lines Total analysis time error with SIP and silence suppression When using app_amd with SIP providers that have silence suppression on, the iTotalTime count increases exponentially. (closes issue #17656) Reported by: juls ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277183 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Add a dialplan function to check if a queue exists: QUEUE_EXISTSoej1-0/+41
Review: https://reviewboard.asterisk.org/r/777/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276950 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14Expand the caller ANI field to an ast_party_idrmudgett1-1/+1
Expand the ani field in ast_party_caller and ast_party_connected_line to an ast_party_id. This is an extension to the ast_callerid restructuring patch in review: https://reviewboard.asterisk.org/r/702/ Review: https://reviewboard.asterisk.org/r/744/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276393 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14ast_callerid restructuringrmudgett24-312/+483
The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-13Merged revisions 275773 via svnmerge from jpeeler1-160/+255
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275773 | jpeeler | 2010-07-12 15:34:51 -0500 (Mon, 12 Jul 2010) | 12 lines Make user removals and traversals thread safe in meetme. Race conditions present in meetme involving the user list where a lack of locking has the potential for a user to be removed during a traversal or as in the case of the reporter after checking if the list is empty could cause a crash. Fixing this was done by convering the userlist to an ao2 container. (closes issue #17390) Reported by: Vince Review: https://reviewboard.asterisk.org/r/746/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276074 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-12Added support for indirect work mode.transnexus1-8/+45
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275551 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-10When creating a conference for a unit test, it is not mandatory to open aeliel1-6/+10
dahdi pseudo channel, so if we fail doing it, continue creating the conference. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275509 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-09Get more information about the Bamboo test failurestilghman1-11/+16
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275312 f38db490-d61c-443f-a65b-d21fe96a405b
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
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