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${ noResults }
3682 Commits (a6fd85250df68587f3042e49edbc515009f743b1)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Tilghman Lesher
|
a6fd85250d |
Properly document voicemail API documents. Also fix a crash reported via the -dev list.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249405 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
|
e20c28078e |
Cleanups to fix bugs in the VM count API functions.
- Urgent voicemails were not attached, because the attachment code looked in the wrong folder. - Urgent voicemails were sometimes counted twice when displaying the count of new messages. - Backends were inconsistent as to which voicemails each API counted. - Unit tests added to verify behavior in the future. (closes issue #15654) Reported by: tomo1657 Patches: 20100225__issue15654.diff.txt uploaded by tilghman (license 14) Tested by: tilghman (closes issue #16448) Reported by: hevad Review: https://reviewboard.asterisk.org/r/525/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249187 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
David Vossel
|
48134df655 |
fixes Queue with C option crash
(closes issue #16475) Reported by: okrief Patches: queue_crash.diff uploaded by dvossel (license 671) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247736 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Mark Michelson
|
c54f8ced1b |
Merged revisions 247168 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247168 | mmichelson | 2010-02-17 10:24:17 -0600 (Wed, 17 Feb 2010) | 3 lines Make sure that when autofill is disabled that callers not in the front of the queue cannot place calls. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247169 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
|
22b144cef4 |
Change the blanket rules to delete .lastclean on all CFLAGS menuselect targets to be more particular.
This change builds upon the recent change to menuselect to add 'touch_on_change' as an attribute of both categories and members. This should allow only the most invasive defines to cause a complete rebuild, while defines which only affect a subset of modules will only cause a rebuild of that smaller set. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246789 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
TransNexus OSP Development
|
034a79c303 |
Updated doc for OSP lookup application.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246382 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
David Vossel
|
fa156c067d |
Merged revisions 246115 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r246115 | dvossel | 2010-02-10 11:44:20 -0600 (Wed, 10 Feb 2010) | 8 lines fixes random deadlock in app_queue with use_weight during reload (closes issue #16677) Reported by: tim_ringenbach Patches: app_queue_use_weight_deadlock.diff uploaded by tim ringenbach (license 540) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246116 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
|
2d6f0c4607 |
Ensure frames are only freed once.
(closes issue #16361) Reported by: vlad Patches: 20100208__issue16361.diff.txt uploaded by tilghman (license 14) Tested by: kenny, bloodoff, misaksen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245729 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Kevin P. Fleming
|
3760672f40 |
Don't offer MMR or JBIG transcoding during T.38 negotiation.
After further discussion with Steve Underwood, we should not (yet) be offering to receive MMR or JBIG transcoded streams from T.38 endpoints. A future spandsp release will support those features, and then they can be enabled during negotiation git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245680 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
|
dd1c5f27ee |
Properly respect GOSUB_RESULT as to what to do with the master channel.
Previously, we would parse GOSUB_RESULT, but not actually do anything with it. Also, allow GOSUB_RETVAL to be inherited back across a peer/master channel. (closes issue #16687) Reported by: bklang Patches: app_dial-preserve-gosub_retval.patch uploaded by bklang (license 919) (with modifications) (closes issue #16686) Reported by: bklang Patches: app_dial-respect-gosub_result.patch uploaded by bklang (license 919) (with modifications) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244393 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
|
81762bf4c7 |
Merged revisions 244242 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r244242 | tilghman | 2010-02-01 17:13:44 -0600 (Mon, 01 Feb 2010) | 11 lines Backup and restore original textfile, for prosthesis (gerund of prepend). Also, fix menuselect such that changing voicemail build options correctly causes rebuild. (closes issue #16415) Reported by: tomo1657 Patches: prepention.patch uploaded by tomo1657 (license 484) (with modifications by me to backport to 1.4) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244243 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Jeff Peeler
|
0f7c1a8cc9 |
Merged revisions 243691 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r243691 | jpeeler | 2010-01-27 14:35:56 -0600 (Wed, 27 Jan 2010) | 5 lines Revert 243570, I should have looked at this closer. Will reopen the issue, but am leaving the review closed as the change was pointless. (issue #16488) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243693 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Jeff Peeler
|
7e20456f3a |
Merged revisions 243570 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r243570 | jpeeler | 2010-01-27 12:47:34 -0600 (Wed, 27 Jan 2010) | 9 lines Extend announcement URL used with Queue from 80 chars to PATH_MAX. (closes issue #16488) Reported by: syspert Patches: soundfilelen.pacth-2 uploaded by syspert (license 938) Review: https://reviewboard.asterisk.org/r/475/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243571 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
David Ruggles
|
0375f18c9f |
Code clean up in app_senddtmf
Pushes code clean up done in app_externalivr back into app_senddtmf Review: https://reviewboard.asterisk.org/r/473/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243346 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
David Ruggles
|
4c49e70ec5 |
Add send DTMF feature to ExternalIVR app
Implemented a new command 'D' that allows client IVRs to send DTMF digits to the channel. (closes issue #16615) Reported by: thedavidfactor Review: https://reviewboard.asterisk.org/r/465/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242357 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
|
873989db91 |
Enable SendText to send strings in encoded format.
See http://lists.digium.com/pipermail/asterisk-users/2010-January/243462.html git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241364 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
David Ruggles
|
174cd3c65c |
Add notification of interrupted file
Add file information to data element of T event so the file information is sent to the client when it is interrupted. Previously only notification of pending files that were dropped was sent (closes issue #16147) Reported by: thedavidfactor Tested by: thedavidfactor Review: https://reviewboard.asterisk.org/r/449/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240969 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
David Vossel
|
8d8800072e |
fixes spelling error. s/memeber/member
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240842 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
|
e8a6d2995e |
Add pickup event to AMI. Also, fix AMI documentation.
(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 |
16 years ago |
|
Tilghman Lesher
|
f94e723a27 |
Make sure that the limit is N, not N - 1.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240419 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
|
6bc1fc7240 |
Merged revisions 240414 via svnmerge from
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 |
16 years ago |
Sean Bright
|
e612d87695 |
Convert a few places to use ast_calloc_with_stringfields where applicable.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240368 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
David Vossel
|
03529837cc |
add silence gen to wait apps
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 |
16 years ago |
|
TransNexus OSP Development
|
912d4da476 |
Updated XML doc for OSP.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239624 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
David Vossel
|
0a6c0ee1f7 |
cli 'queue show' formatting fix. queue name was truncated over 12 characters
(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 |
16 years ago |
|
Jeff Peeler
|
c6e038ba16 |
Fix misreverting from 177158.
(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 |
16 years ago |
Russell Bryant
|
5d7b80248b |
Merged revisions 238009 via svnmerge from
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 |
16 years ago |
|
David Vossel
|
bfae8dca78 |
fixes holdtime playback issue in app_queue
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 |
16 years ago |
|
Mark Michelson
|
2fa64b3ad4 |
Mismerged a bit.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237882 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Mark Michelson
|
c9d1ffcae8 |
Add a missing part of the connected line work into trunk.
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 |
16 years ago |
Michiel van Baak
|
0c62434201 |
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: https://origsvn.digium.com/svn/asterisk/trunk@237656 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
David Vossel
|
688e1bbac6 |
app_queue segfaults if realtime field uniqueid is NULL
(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 |
16 years ago |
|
TransNexus OSP Development
|
fb4870a74a |
1. Added reporting operator names in AuthReq.
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 |
16 years ago |
Jason Parker
|
f93071483f |
Add app_voicemail and say.c support for Vietnamese.
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 |
16 years ago |
|
TransNexus OSP Development
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28d16a3cb1 |
1. Updated for OSP Toolkit 3.6.0.
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 |
16 years ago |
|
Tilghman Lesher
|
06dc97772e |
Use recommended option, not deprecated option.
(closes issue #16515) Reported by: ManChicken git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236667 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Sean Bright
|
82446789f3 |
Merged revisions 236509 via svnmerge from
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 |
16 years ago |
|
David Vossel
|
0a5d21e6c7 |
QUEUE_MEMBER(..., ready) counts only ready agents, not free agents wrapping up
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 |
16 years ago |
|
David Vossel
|
065fce7310 |
update CHANGES to reflect new 'R' app_queue option plus a minor optimization to the feature patch
(issue #16384) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236306 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
David Vossel
|
6892b103ab |
new parameter 'R' to the Queue application
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 |
16 years ago |
|
Tilghman Lesher
|
169b74c313 |
AGI may be invoked from outside the dialplan
(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 |
16 years ago |
|
Tilghman Lesher
|
1e0306a04b |
Actually use tmp for something (brings trunk back into sync with 1.6 branches).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236183 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Alec L Davis
|
7537d3c0cb |
app_dial optional parameter to option 'r' to allow play indication from 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 |
16 years ago |
|
Kevin P. Fleming
|
df1fc1f381 |
spandsp does in fact support V.17 modulation at 14.4 kilobits per second,
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 |
16 years ago |
|
Alec L Davis
|
13c3260c92 |
Support option 'n', as applications like Playback, Background etc.
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 |
16 years ago |
|
Alec L Davis
|
90be4cf5ef |
fixes escape to extensions 'o' and 'a', for digits '0' and '*'
(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 |
16 years ago |
|
Alec L Davis
|
19f8080654 |
ast_stream_and_wait(chan,dir-usingkeypad) didn't capture the dialled DTMF.
(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 |
16 years ago |
|
Tilghman Lesher
|
89a1af1d38 |
Allow greetings-only mailboxes for Voicemail.
(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 |
16 years ago |
|
Jason Parker
|
e52ee5c8e6 |
Allow tonelist as argument to ReadExten.
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 |
16 years ago |
|
Jeff Peeler
|
2923086daf |
Merged revisions 234379 via svnmerge from
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 |
16 years ago |