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2010-01-15Add pickup event to AMI. Also, fix AMI documentation.Tilghman Lesher
(closes issue #16431) Reported by: syspert Patches: 20100112__issue16431.diff.txt uploaded by tilghman (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240421 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-15Make sure that the limit is N, not N - 1.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240419 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-15Merged revisions 240414 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r240414 | tilghman | 2010-01-15 14:52:27 -0600 (Fri, 15 Jan 2010) | 15 lines Disallow leaving more than maxmsg voicemails. This is a possibility because our previous method assumed that no messages are left in parallel, which is not a safe assumption. Due to the vmu structure duplication, it was necessary to track in-process messages via a separate structure. If at some point, we switch vmu to an ao2-reference-counted structure, which would eliminate the prior noted duplication of structures, then we could incorporate this new in-process structure directly into vmu. (closes issue #16271) Reported by: sohosys Patches: 20100108__issue16271.diff.txt uploaded by tilghman (license 14) 20100108__issue16271__trunk.diff.txt uploaded by tilghman (license 14) 20100108__issue16271__1.6.0.diff.txt uploaded by tilghman (license 14) Tested by: jsutton ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240415 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-15Convert a few places to use ast_calloc_with_stringfields where applicable.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-13add silence gen to wait appsDavid Vossel
asterisk.conf's 'transmit_silence' option existed before this patch, but was limited to only generating silence while recording and sending DTMF. Now enabling the transmit_silence option generates silence during wait times as well. To achieve this, ast_safe_sleep has been modified to generate silence anytime no other generators are present and transmit_silence is enabled. Wait apps not using ast_safe_sleep now generate silence when transmit_silence is enabled as well. (closes issue #16524) Reported by: kobaz (closes issue #16523) Reported by: kobaz Tested by: dvossel Review: https://reviewboard.asterisk.org/r/456/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239712 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-13Updated XML doc for OSP.TransNexus OSP Development
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239624 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-07cli 'queue show' formatting fix. queue name was truncated over 12 charactersDavid Vossel
(closes issue #16078) Reported by: RoadKill Patches: quequename_limit.patch uploaded by ppyy (license 906) Tested by: dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@238361 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-06Fix misreverting from 177158.Jeff Peeler
(closes issue #15725) Reported by: shanermn Patches: v1-15725.patch uploaded by dimas (license 88) Tested by: shanermn git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@238181 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-06Merged revisions 238009 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r238009 | russell | 2010-01-06 09:18:22 -0600 (Wed, 06 Jan 2010) | 7 lines Resolve a crash due to an ast_frame not being fully initialized. (closes issue #16531) Reported by: john8675309 (closes SWP-615) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@238010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-05fixes holdtime playback issue in app_queueDavid Vossel
When reporting hold time, the number of seconds should be mod 60. Otherwise audio playback could be something like "2 minutes 123 seconds" rather than "2 minutes 3 seconds". Also, the "minute" sound file is missing, so for the moment until that file can be created the "minutes" file is used instead. (closes issue #16168) Reported by: nickilo Patches: patch-unified-trunk-rev-222176 uploaded by nickilo (license ) Tested by: nickilo, wonderg git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237920 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-05Mismerged a bit.Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237882 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-05Add a missing part of the connected line work into trunk.Mark Michelson
Part of the work done for connected line was to add an optional argument to the 'f' option to allow for the connected party information of the outgoing channel to be set to the argument provided. This was overlooked during the merge of the work to trunk and is being added back now. The CHANGES file has also been updated to note this change. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237803 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-05Make CLI command 'mixmonitor start|stop <channel> work again.Michiel van Baak
(closes issue #16534) Reported by: jlaguilar Fix as suggested by jlaguilar in the bugreport git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237656 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-04app_queue segfaults if realtime field uniqueid is NULLDavid Vossel
(closes issue #16385) Reported by: haakon Patches: app_queue.c.patch uploaded by haakon (license 880) app_queue.c.patch_v2 uploaded by dvossel (license 671) Tested by: haakon git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237327 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-041. Added reporting operator names in AuthReq.TransNexus OSP Development
2. Added retrieving operator names from AuthRsp and exporting them. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237250 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-30Add app_voicemail and say.c support for Vietnamese.Jason Parker
Also add an XXX comment that I'm baffled nobody has ever complained about. We say "first message", and then we go into language-specific stuff where we proceed to say..."first message". (closes issue #15053) Reported by: dinhtrung Patches: vietnamese.ods uploaded by dinhtrung (license 776) app_voicemail.c.diff uploaded by dinhtrung (license 776) (closes issue #15626) Reported by: dinhtrung Patches: say.c.diff uploaded by dinhtrung (license 776) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-291. Updated for OSP Toolkit 3.6.0.TransNexus OSP Development
2. Added service type ported number query. 3. Formated code. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236756 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-28Use recommended option, not deprecated option.Tilghman Lesher
(closes issue #16515) Reported by: ManChicken git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236667 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-28Merged revisions 236509 via svnmerge from Sean Bright
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236509 | seanbright | 2009-12-28 07:43:36 -0500 (Mon, 28 Dec 2009) | 12 lines Avoid a crash with large numbers of MeetMe conferences. Similar to changes made to Queue(), when we have large numbers of conferences in meetme.conf (1000s) and we use alloca()/strdupa(), we can blow out the stack and crash, so instead just use a single fixed buffer. (closes issue #16509) Reported by: Kashif Raza Patches: 20091223_16509.patch uploaded by seanbright (license 71) Tested by: seanbright ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236510 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-23QUEUE_MEMBER(..., ready) counts only ready agents, not free agents wrapping upDavid Vossel
The QUEUE_MEMBER dialplan function can return total members, logged-in members and "free" members count. A member is counted as "free" immediately after his call ends, even though its wrap-up time, if specified in queues.conf, has not yet expired, and the queue will not actually route a call to it. This Patch introduces a new "ready" option that only counts free agents no longer in the wrap up time period. (closes issue #16240) Reported by: kkm Patches: appqueue-memberfun-readyoption-trunk.diff uploaded by kkm (license 888) Tested by: kkm, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236308 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-23update CHANGES to reflect new 'R' app_queue option plus a minor optimization ↵David Vossel
to the feature patch (issue #16384) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236306 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-23new parameter 'R' to the Queue applicationDavid Vossel
The 'R' argument stops moh and indicates ringing once the agent is ringing. This allows the person in the queue to know their call is potentially about to be answered. (closes issue #16384) Reported by: haakon Patches: new_app_queue.c.patch uploaded by haakon (license 880) Tested by: haakon, loloski, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-23AGI may be invoked from outside the dialplanTilghman Lesher
(closes issue #16510) Reported by: atis Patches: 20091223__issue16510.diff.txt uploaded by tilghman (license 14) Tested by: atis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236300 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-23Actually use tmp for something (brings trunk back into sync with 1.6 branches).Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236183 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-19app_dial optional parameter to option 'r' to allow play indication from ↵Alec L Davis
indications.conf (closes issue #14504) Reported by: alecdavis Tested by: alecdavis,jsmith Patch app_dial.play_ring_indications.diff7.txt uploaded by alecdavis (license 585) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-15spandsp does in fact support V.17 modulation at 14.4 kilobits per second,Kevin P. Fleming
so we should generate T38MaxBitRate of 14400 (even though that doesn't really affect the FAX transmission much at all) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-15Support option 'n', as applications like Playback, Background etc.Alec L Davis
Suggested on asterisk-dev as trivial application change. Reported by: alecdavis Tested by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234976 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-15fixes escape to extensions 'o' and 'a', for digits '0' and '*'Alec L Davis
(closes issue #16437) Reported by: alecdavis Tested by: alecdavis Patch extension_o_a_fix.diff.txt uploaded by alecdavis (license 585) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-15ast_stream_and_wait(chan,dir-usingkeypad) didn't capture the dialled DTMF.Alec L Davis
(closes issue #16409) Reported by: alecdavis Tested by: alecdavis Patch bug_16409.diff.txt uploaded by alecdavis (license 585) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234855 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-14Allow greetings-only mailboxes for Voicemail.Tilghman Lesher
(closes issue #15132) Reported by: floletarmo Patches: voicemail_changes.patch uploaded by floletarmo (license 784) (with some additional changes by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234820 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-14Allow tonelist as argument to ReadExten.Jason Parker
ReadExten already supported playing a tonezone from indications.conf. It now has the ability to use a tonelist like 440+480/2000|0/4000 (closes issue #15185) Reported by: jcovert Patches: app_readexten.c.patch uploaded by jcovert (license 551) Tested by: qwell Patch modified by me, to maintain backwards compatibility. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234776 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-11Merged revisions 234379 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r234379 | jpeeler | 2009-12-11 16:37:21 -0600 (Fri, 11 Dec 2009) | 11 lines Fix talking detection status after conference user is muted. This patch ensures that when a conference user is muted that the accompanying AMI Meetme talking off event is sent. Also, the meetme list output is updated to show the muted user as unmonitored. (closes issue #16247) Reported by: dimas Patches: v3-16247.patch uploaded by dimas (license 88) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234380 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-10Add audio announcement option to app_pageJeff Peeler
As described in the CHANGES file: * MeetMe has a new option 'G' to play an announcement before joining a conference. * Page has a new option 'A(x)' which will playback an announcement simultaneously to all paged phones (and optionally excluding the caller's one using the new option 'n') before the call is bridged. To add the new option to meetme, the conference flag options had to be extended to 64 bits. (closes issue #14365) Reported by: dferrer Patches: page_announce.patch uploaded by dferrer (license 525) modified by me Review: https://reviewboard.asterisk.org/r/188/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234173 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-07Fix TCP Client interfaceDavid Ruggles
Fix a couple of very minor bugs that prevent the socket client from working. The wrong set of properties were used in one place and the size of the address variable isn't set if the host name is an ip address. Also includes a fix for a bug that was introduced previously. (closes issue #16121) Reported by: thedavidfactor Tested by: thedavidfactor Review: https://reviewboard.asterisk.org/r/439/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233545 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-04.m3u support for Mp3Player appDavid Vossel
(closes issue #14823) Reported by: macli Patches: app_mp3.diff1 uploaded by macli (license ) Tested by: macli, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233234 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-04changes penaltymemberslimit to use scanf for config value parsingDavid Vossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233197 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-04new queue option, penaltymemberslimit, disregards penalty on too few queue ↵David Vossel
members when enabled (closes issue #14559) Reported by: fiddur Patches: trunk-199584-1.diff uploaded by fiddur (license 678) Tested by: fiddur, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233196 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-04Merged revisions 233116 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r233116 | dvossel | 2009-12-04 11:21:34 -0600 (Fri, 04 Dec 2009) | 6 lines document and rename strip_control() in app_voicemail (closes issue #16291) Reported by: wdoekes ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-03Add pagerdateformat, to allow shorter dates for SMS messages.Tilghman Lesher
(closes issue #16263) Reported by: andrew Patches: pagerdate.patch uploaded by andrew (license 240) (with a slight modification by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232916 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-03Merged revisions 232820 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232820 | tilghman | 2009-12-03 14:10:19 -0600 (Thu, 03 Dec 2009) | 8 lines Deprecate "cz" in favor of "cs". Also, change the use of language codes so that language registers as a prefix, rather than an exact match. (closes issue #16272) Reported by: patrol-cz Patches: 20091203__issue16272.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232854 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-03Replaced two deprecated functions of OSP Toolkit.TransNexus OSP Development
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232771 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-03Added custom info support.TransNexus OSP Development
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232738 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-03Extend voicemail to allow IMAP folders to be specified per mailbox.Jeff Peeler
Previously only possible per context, new option called imapfolder. (closes issue #14298) Reported by: jablko Patches: patch-200906202 uploaded by jablko (license 675) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232700 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-02Prevent double closing of FDs by EIVRDavid Ruggles
This caused a problem when asterisk was under heavy load and running both AGI and EIVR applications. EIVR would close an FD at which point it would be considered freed and be used by a new AGI instance the second close would then close the FD now in use by AGI. (closes issue #16305) Reported by: diLLec Tested by: thedavidfactor, diLLec Review: https://reviewboard.asterisk.org/r/436/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232587 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-02Add an option to Record which enables a mode where any DTMF digit will ↵Joshua Colp
terminate recording. (closes issue #15436) Reported by: Vince Patches: app_record.diff uploaded by Vince (license 823) Tested by: dbrooks git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232442 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-02Merged revisions 232355 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r232355 | file | 2009-12-02 13:04:52 -0400 (Wed, 02 Dec 2009) | 5 lines Fix a bug where if you hung up very quickly after calling AMD it would overwrite the AMDSTATUS of HANGUP with TOOLONG. (closes issue #16239) Reported by: CGMChris ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232356 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-30Merged revisions 231614 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov 2009) | 8 lines Remove duplicate entries from voicemail format lists. This prevents app_voicemail from entering an infinite loop when the same format is specified twice in the format list. (closes issue #15625) Reported by: Shagg63 Tested by: mnicholson Review: https://reviewboard.asterisk.org/r/429/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231688 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-30Reverted 231616Matthew Nicholson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-30Merged revisions 231614 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov 2009) | 8 lines Remove duplicate entries from voicemail format lists. This prevents app_voicemail from entering an infinite loop when the same format is specified twice in the format list. (closes issue #15625) Reported by: Shagg63 Tested by: mnicholson Review: https://reviewboard.asterisk.org/r/429/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231616 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-30app_queue crashes randomly, often during call-transfersDavid Vossel
This patch adds a ref to the queue_ent object's parent call_queue in queue_exec() so the call_queue won't be destroyed while the the queue_ent still holds a pointer to it. (closes issue 0015686) Tested by: dvossel, aragon git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231556 65c4cc65-6c06-0410-ace0-fbb531ad65f3