mirror of https://github.com/asterisk/asterisk
master
20
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${ noResults }
3427 Commits (cebb241a4463e063721f27279552718a35a07ac7)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
b4e2771bb2 |
Fixed compile error with OSP Toolkit 3.6.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@233688 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
|
be35aaf361 |
Merged revisions 233121 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r233121 | dvossel | 2009-12-04 11:22:31 -0600 (Fri, 04 Dec 2009) | 12 lines Merged revisions 233116 via svnmerge from 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/branches/1.6.1@233166 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 232854 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r232854 | tilghman | 2009-12-03 14:47:07 -0600 (Thu, 03 Dec 2009) | 15 lines Merged revisions 232820 via svnmerge from 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/branches/1.6.1@232865 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 232587 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r232587 | diruggles | 2009-12-02 17:17:22 -0500 (Wed, 02 Dec 2009) | 12 lines Prevent double closing of FDs by EIVR 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/branches/1.6.1@232812 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 232356 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r232356 | file | 2009-12-02 13:06:54 -0400 (Wed, 02 Dec 2009) | 12 lines Merged revisions 232355 via svnmerge from 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/branches/1.6.1@232358 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 231688 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r231688 | mnicholson | 2009-11-30 15:31:55 -0600 (Mon, 30 Nov 2009) | 15 lines Merged revisions 231614 via svnmerge from 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/branches/1.6.1@231690 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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03f3ad606e |
Turn off debug mode in 1.6.1; fix such that debug mode and non-debug mode functions return the same types.
(Fixes an issue brought up in chat by twilson) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@231608 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 231556 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r231556 | dvossel | 2009-11-30 12:55:07 -0600 (Mon, 30 Nov 2009) | 11 lines app_queue crashes randomly, often during call-transfers 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/branches/1.6.1@231559 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 231134 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r231134 | tilghman | 2009-11-24 14:31:28 -0600 (Tue, 24 Nov 2009) | 7 lines Found a few places where queue refcounts were counted incorrectly. Also add debug statements. (closes issue #15982, closes issue #15984) Reported by: atis Patches: 20091111__issue15982.diff.txt uploaded by tilghman (license 14) Tested by: atis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@231135 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 230509 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r230509 | dvossel | 2009-11-19 15:26:21 -0600 (Thu, 19 Nov 2009) | 17 lines Merged revisions 230508 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r230508 | dvossel | 2009-11-19 15:22:46 -0600 (Thu, 19 Nov 2009) | 10 lines fixes MixMonitor thread not exiting when StopMixMonitor is used (closes issue #16152) Reported by: AlexMS Patches: stopmixmonitor_1.4.diff uploaded by dvossel (license 671) Tested by: dvossel, AlexMS Review: https://reviewboard.asterisk.org/r/424/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@230511 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 230381 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r230381 | kpfleming | 2009-11-16 10:40:25 -0600 (Mon, 16 Nov 2009) | 1 line Fix another buglet in T.38 session teardown at the end of FAX sessions. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@230383 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 230343 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r230343 | kpfleming | 2009-11-16 06:51:59 -0600 (Mon, 16 Nov 2009) | 2 lines Ensure that only one end of a T.38 session initiates teardown at completion. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@230345 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 229966 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r229966 | file | 2009-11-13 11:20:26 -0600 (Fri, 13 Nov 2009) | 13 lines Merged revisions 229965 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229965 | file | 2009-11-13 11:19:59 -0600 (Fri, 13 Nov 2009) | 6 lines Document a limitation in the AVAILSTATUS variable from ChanIsAvail and provide a workaround for it that does not change existing behavior. (closes issue #14426) Reported by: macli ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@229968 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 229460 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r229460 | dbrooks | 2009-11-11 12:13:56 -0600 (Wed, 11 Nov 2009) | 7 lines Flags not initialized in app_softhangup.c, causing undefined behavior Trivial patch [kobaz] to initialize an ast_flags = {0} (closes issue #16129) Reported by: kobaz ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@229491 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 229351 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r229351 | tilghman | 2009-11-10 15:22:50 -0600 (Tue, 10 Nov 2009) | 7 lines When GOSUB is invoked within an AGI, it may not exit correctly. (closes issue #16216) Reported by: atis Patches: 20091110__atis_work.diff.txt uploaded by tilghman (license 14) Tested by: atis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@229353 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 228196 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r228196 | tilghman | 2009-11-05 16:12:45 -0600 (Thu, 05 Nov 2009) | 2 lines Yet another error message in the dialplan (thanks, rmudgett/russellb) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@228197 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 228191 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r228191 | tilghman | 2009-11-05 15:24:21 -0600 (Thu, 05 Nov 2009) | 7 lines MEETME_INFO should not return a literal error message to the dialplan. (closes issue #15450) Reported by: JimVanM Patches: meetmeinfopatch.diff.txt uploaded by dbrooks (license 790) Tested by: JimVanM ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@228193 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 228189 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r228189 | jpeeler | 2009-11-05 15:23:06 -0600 (Thu, 05 Nov 2009) | 11 lines Fix the fix for chanspy option o In 224178, I assumed the uploaded patch was correct as it had received positive feedback. The flags were being checked in the incorrect location. Upon testing the fix this time it was also found that the flags from the dialplan weren't being copied to the chanspy_translation_helper. (closes issue #16167) Reported by: marhbere ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@228192 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 228015 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r228015 | tilghman | 2009-11-05 11:08:02 -0600 (Thu, 05 Nov 2009) | 4 lines Don't crash if no arguments are passed. (closes issue #16119) Reported by: thedavidfactor ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@228016 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 227829 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r227829 | mnicholson | 2009-11-04 15:03:33 -0600 (Wed, 04 Nov 2009) | 17 lines Merged revisions 227827 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227827 | mnicholson | 2009-11-04 14:52:27 -0600 (Wed, 04 Nov 2009) | 10 lines This patch modifies the Dial application to monitor the calling channel for hangups while playing back announcements. (closes issue #16005) Reported by: falves11 Patches: dial-announce-hangup-fix1.diff uploaded by mnicholson (license 96) Tested by: mnicholson, falves11 Review: https://reviewboard.asterisk.org/r/407/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@227832 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 227368 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r227368 | lmadsen | 2009-11-03 13:48:53 -0600 (Tue, 03 Nov 2009) | 8 lines Change warning message to debug message. app_controlplayback outputs a warning, when in fact it is normal. (closes issue #16071) Reported by: atis Patches: controlplayback_warning.patch uploaded by atis (license 242) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@227370 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 226890 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r226890 | file | 2009-11-02 14:08:54 -0400 (Mon, 02 Nov 2009) | 18 lines Merged revisions 226889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226889 | file | 2009-11-02 14:08:11 -0400 (Mon, 02 Nov 2009) | 11 lines Fix a bug where the recorded privacy introduction file would not get removed if the caller hung up while the called party had not yet answered. This was fixed by introducing an argument to the 'n' option which, when enabled, removes the introduction file under all scenarios. This was done to preserve the behavior that has existed for quite some time. (closes issue #14674) Reported by: ulogic Patches: bug14674.patch uploaded by jpeeler (license 325) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@226892 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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e236b8ad49 |
Backport audio handling loop fixes from trunk version of app_fax.
This backport resolves some issues handling audio frames during FAX processing, and ensures that the FAX application doesn't accidentally get notified of a T.38 switchover at the end of a successful FAX. (issue #16127) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@225870 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 225445 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r225445 | dvossel | 2009-10-22 14:55:51 -0500 (Thu, 22 Oct 2009) | 50 lines SIP TCP/TLS: move client connection setup/write into tcp helper thread, various related locking/memory fixes. What this patch fixes 1.Moves sip TCP/TLS connection setup into the TCP helper thread: Connection setup takes awhile and before this it was being done while holding the monitor lock. 2.Moves TCP/TLS writing to the TCP helper thread: Through the use of a packet queue and an alert pipe, the TCP helper thread can now be woken up to write data as well as read data. 3.Locking error: sip_xmit returned an XMIT_ERROR without giving up the tcptls_session lock. This lock has been completely removed from sip_xmit and placed in the new sip_tcptls_write() function. 4.Memory leak: When creating a tcptls_client the tls_cfg was alloced but never freed unless the tcptls_session failed to start. Now the session_args for a sip client are an ao2 object which frees the tls_cfg on destruction. 5.Pointer to stack variable: During sip_prepare_socket the creation of a client's ast_tcptls_session_args was done on the stack and stored as a pointer in the newly created tcptls_session. Depending on the events that followed, there was a slight possibility that pointer could have been accessed after the stack returned. Given the new changes, it is always accessed after the stack returns which is why I found it. Notable code changes 1.I broke tcptls.c's ast_tcptls_client_start() function into two functions. One for creating and allocating the new tcptls_session, and a separate one for starting and handling the new connection. This allowed me to create the tcptls_session, launch the helper thread, and then establish the connection within the helper thread. 2.Writes to a tcptls_session are now done within the helper thread. This is done by using an alert pipe to wake up the thread if new data needs to be sent. The thread's sip_threadinfo object contains the alert pipe as well as the packet queue. 3.Since the threadinfo object contains the alert pipe, it must now be accessed outside of the helper thread for every write (queuing of a packet). For easy lookup, I moved the threadinfo objects from a linked list to an ao2_container. (closes issue #13136) Reported by: pabelanger Tested by: dvossel, whys (closes issue #15894) Reported by: dvossel Tested by: dvossel Review: https://reviewboard.asterisk.org/r/380/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@225490 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 225360 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r225360 | tilghman | 2009-10-22 12:11:23 -0500 (Thu, 22 Oct 2009) | 11 lines Merged revisions 225105 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009) | 4 lines Fix documentation for ast_softhangup() and correct the misuse thereof. (closes issue #16103) Reported by: majorbloodnok ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@225362 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 224567 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r224567 | file | 2009-10-19 16:49:09 -0300 (Mon, 19 Oct 2009) | 12 lines Merged revisions 224565 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224565 | file | 2009-10-19 16:47:50 -0300 (Mon, 19 Oct 2009) | 5 lines Do not attempt early media bridging (ie: direct RTP setup) if options are enabled that should prevent it. (closes issue #14763) Reported by: cupotka ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@224570 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 224178 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r224178 | jpeeler | 2009-10-15 10:57:14 -0500 (Thu, 15 Oct 2009) | 11 lines Readd removed ability to allow listening to one side of the call in app_chanspy (Option o) (closes issue #15675) Reported by: john8675309 Patches: issue15675patchtrunk.txt uploaded by dbrooks (license 790) Tested by: jgutierrez on users list: http://lists.digium.com/pipermail/asterisk-users/2009-October/239155.html ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@224180 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 223832 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r223832 | jpeeler | 2009-10-12 18:48:09 -0500 (Mon, 12 Oct 2009) | 15 lines Merged revisions 223804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223804 | jpeeler | 2009-10-12 18:12:50 -0500 (Mon, 12 Oct 2009) | 8 lines Ensure ringing continues for branched calls after progress is received While waiting for an answer, don't send progress for branched calls for which ringing was sent. (closes issue #15028) Reported by: fnordian ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@223834 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 223652 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r223652 | kpfleming | 2009-10-12 09:25:29 -0500 (Mon, 12 Oct 2009) | 13 lines Remove automatic switching from T.38 to voice mode in chan_sip. chan_sip has some code to automatically switch from T.38 mode to voice mode when a voice frame is written to the channel while it is in T.38 mode; this was intended to handle the situation when a FAX transmission has ended and the channel is not yet hung up, but is causing problems at the beginning of FAX sessions as well when there are still voice frames 'in flight' at the time the T.38 negotiation completes. This patch removes the automatic switchover, and changes app_fax to explicitly switch off T.38 mode when the FAX transmission process ends. (closes issue #16025) Reported by: jamicque ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@223654 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 223330 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r223330 | kpfleming | 2009-10-09 15:58:44 -0500 (Fri, 09 Oct 2009) | 10 lines Initiate T.38 switchover when acting as called party, regardless of FAX direction. SendFAX() and ReceiveFAX() can be given options to indicate whether they should act as the calling or called party; this mode should be used to decide whether to initiate a switchover to T.38, not the direction that the FAX transfer will take place. (closes issue #16039) Reported by: jamicque ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@223332 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 223215 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r223215 | mmichelson | 2009-10-09 13:17:34 -0500 (Fri, 09 Oct 2009) | 9 lines Recorded merge of revisions 223213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223213 | mmichelson | 2009-10-09 13:17:12 -0500 (Fri, 09 Oct 2009) | 3 lines Fix potential memory leak in app_dial.c ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@223241 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 222176 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r222176 | kpfleming | 2009-10-05 20:24:24 -0500 (Mon, 05 Oct 2009) | 27 lines Recorded merge of revisions 222152 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct 2009) | 20 lines Fix ao2_iterator API to hold references to containers being iterated. See Mantis issue for details of what prompted this change. Additional notes: This patch changes the ao2_iterator API in two ways: F_AO2I_DONTLOCK has become an enum instead of a macro, with a name that fits our naming policy; also, it is now necessary to call ao2_iterator_destroy() on any iterator that has been created. Currently this only releases the reference to the container being iterated, but in the future this could also release other resources used by the iterator, if the iterator implementation changes to use additional resources. (closes issue #15987) Reported by: kpfleming Review: https://reviewboard.asterisk.org/r/383/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@222186 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 221436 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r221436 | mnick | 2009-09-30 16:15:01 -0500 (Wed, 30 Sep 2009) | 2 lines Prevents from division by zero ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@221470 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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7e2bac719e |
Merged revisions 221085 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r221085 | seanbright | 2009-09-30 10:47:58 -0400 (Wed, 30 Sep 2009) | 9 lines Clarify documentation for VoiceMailMain()'s a() option. We require box numbers, not names as the documentation implies. (issue #14740) Reported by: pj Patches: __20090729-app_voicemail-documentation.patch uploaded by lmadsen (license 10) Tested by: seanbright, lmadsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@221088 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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185e4321a7 |
Avoid a deadlock in chanspy, just in case the spyee is masqueraded and chanspy_ds_chan_fixup() is called with the channel locked.
(closes issue #15965) Reported by: atis Patches: chanspy-deadlock-fix1.diff uploaded by mnicholson (license 96) Tested by: atis git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@220938 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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75ab04c1cd |
Merged revisions 220833 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r220833 | jpeeler | 2009-09-29 11:58:29 -0500 (Tue, 29 Sep 2009) | 12 lines Make deletion of temporary greetings work properly with IMAP_STORAGE When imapgreetings was set to yes, the message was being deleted but wasn't actually being expunged. When imapgreetings was set to no, the file based message was not being deleted at all. All good now! (closes issue #14949) Reported by: noahisaac Patches: vm_tempgreeting_removal.patch uploaded by noahisaac (license 748), modified by me ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@220835 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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698a9ede87 |
Merged revisions 220289 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r220289 | tilghman | 2009-09-24 14:41:02 -0500 (Thu, 24 Sep 2009) | 13 lines Merged revisions 220288 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220288 | tilghman | 2009-09-24 14:39:41 -0500 (Thu, 24 Sep 2009) | 6 lines Implicitly sending a progress signal breaks some applications. Call Progress() in your dialplan if you explicitly want progress to be sent. (Reverts change 216430, closes issue #15957) Reported by: Pavel Troller on the Asterisk-Dev mailing list http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@220291 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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f0d39f6478 |
Merged revisions 219987 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r219987 | tilghman | 2009-09-24 02:39:44 -0500 (Thu, 24 Sep 2009) | 8 lines Fix two possible crashes, one only in 1.6.1 and one in 1.6.1 forward. (closes issue #15739) Reported by: DLNoah, jeffg Patches: 20090914__issue15739.diff.txt uploaded by tilghman (license 14) 20090922__issue15739.diff.txt uploaded by tilghman (license 14) Tested by: DLNoah, jeffg ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@219988 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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5a26f37df8 |
Merged revisions 219818 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r219818 | tilghman | 2009-09-22 16:43:22 -0500 (Tue, 22 Sep 2009) | 17 lines Merged revisions 219816 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219816 | tilghman | 2009-09-22 16:37:03 -0500 (Tue, 22 Sep 2009) | 10 lines When IMAP variables were changed during a reload, Voicemail did not use the new values. This change introduces a configuration version variable, which ensures that connections with the old values are not reused but are allowed to expire normally. (closes issue #15934) Reported by: viniciusfontes Patches: 20090922__issue15934.diff.txt uploaded by tilghman (license 14) Tested by: viniciusfontes ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@219820 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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f7ea3005d2 |
Merged revisions 219412 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r219412 | tilghman | 2009-09-18 08:54:51 -0500 (Fri, 18 Sep 2009) | 6 lines Missing value setting line for maxsecs/maxmessage (closes issue #15696) Reported by: fhackenberger Patches: maxsecs.patch uploaded by fhackenberger (license 592) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@219414 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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28f7949944 |
Merged revisions 218731 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r218731 | tilghman | 2009-09-15 17:33:10 -0500 (Tue, 15 Sep 2009) | 13 lines Merged revisions 218730 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218730 | tilghman | 2009-09-15 17:27:41 -0500 (Tue, 15 Sep 2009) | 6 lines If the user enters the same password as before, don't signal an error when the change does nothing. (closes issue #15492) Reported by: cbbs70a Patches: 20090713__issue15492.diff.txt uploaded by tilghman (license 14) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@218734 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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c4865fe228 |
Merged revisions 218579 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r218579 | tilghman | 2009-09-15 11:04:41 -0500 (Tue, 15 Sep 2009) | 16 lines Merged revisions 218577 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218577 | tilghman | 2009-09-15 11:01:17 -0500 (Tue, 15 Sep 2009) | 9 lines Ensure FollowMe sets language in channels it creates. Also, not in the original bug report, but related fields are accountcode and musicclass, and the inheritance of datastores. (closes issue #15372) Reported by: Romik Patches: 20090828__issue15372.diff.txt uploaded by tilghman (license 14) Tested by: cervajs ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@218581 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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7a56903be3 |
Merged revisions 218361 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r218361 | tilghman | 2009-09-14 14:29:48 -0500 (Mon, 14 Sep 2009) | 11 lines Recorded merge of revisions 218331 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218331 | tilghman | 2009-09-14 14:16:35 -0500 (Mon, 14 Sep 2009) | 4 lines Don't say "Please try again" if we don't give the user another chance to try again. (issue #15055, SWP-129) Reported by: jthurman ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@218363 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Merged revisions 218224 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r218224 | mnicholson | 2009-09-14 09:57:23 -0500 (Mon, 14 Sep 2009) | 14 lines Merged revisions 218223 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218223 | mnicholson | 2009-09-14 09:53:57 -0500 (Mon, 14 Sep 2009) | 8 lines Ensure we don't pickup ourselves when doing pickup by exten. (closes issue #15100) Reported by: lmsteffan Patches: (modified) pickup.patch uploaded by lmsteffan (license 779) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@218227 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Merged revisions 217990 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r217990 | tilghman | 2009-09-10 18:54:51 -0500 (Thu, 10 Sep 2009) | 10 lines Merged revisions 217989 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r217989 | tilghman | 2009-09-10 18:52:22 -0500 (Thu, 10 Sep 2009) | 3 lines Don't ring another channel, if there's not enough time for a queue member to answer. (Fixes AST-228) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@218053 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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be3c80c535 |
Merged revisions 217286 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r217286 | seanbright | 2009-09-08 18:17:08 -0400 (Tue, 08 Sep 2009) | 4 lines Fix compilation of app_meetme. Reported by ebroad in #asterisk-bugs ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@217295 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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1cdd533d6a |
Merged revisions 217199 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r217199 | tilghman | 2009-09-08 15:28:41 -0500 (Tue, 08 Sep 2009) | 14 lines Merged revisions 217156 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r217156 | tilghman | 2009-09-08 15:01:45 -0500 (Tue, 08 Sep 2009) | 7 lines When MOH is playing on the channel, announcements sent through the conference are not heard. (closes issue #14588) Reported by: voipas Patches: 20090716__issue14588__2.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen, twisted, tilghman ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@217213 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Merged revisions 216438 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r216438 | oej | 2009-09-04 16:02:34 +0200 (Fre, 04 Sep 2009) | 35 lines Merged revisions 216430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 lines Make apps send PROGRESS control frame for early media and fix too early media issue in SIP The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI links *before* any call progress. The SIP channel receives these frames and by default signals 183 Session progress and starts sending media. This will cause phones to play silence and ignore the later 180 ringing message. A bad user experience. The fix is twofold: - We discovered that asterisk apps that support early media ("noanswer") did not send any PROGRESS frame to indicate early media. Fixed. - We introduce a setting in chan_sip so that users can disable any relay of media frames before the outbound channel actually indicates any sort of call progress. In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions of Asterisk, this will be enabled. We don't assume that it will change your Asterisk phone experience - only for the better. We encourage third-party application developers to make sure that if they have applications that wants to send early media, add a PROGRESS control frame transmission to make sure that all channel drivers actually will start sending early media. This has not been the default in Asterisk previous to this patch, so if you got inspiration from our code, you need to update accordingly. Sorry for the trouble and thanks for your support. This code has been running for a few months in a large scale installation (over 250 servers with PRI and/or BRI links to old PBX systems). That's no proof that this is an excellent patch, but, well, it's tested :-) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@216646 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Merged revisions 216593 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r216593 | seanbright | 2009-09-04 15:29:02 -0400 (Fri, 04 Sep 2009) | 1 line Use ast_free() instead of free(). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@216596 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |