mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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17581 Commits (8d5fab9e52ef89e0dc318af2ceb3bb4bb8a5ffa4)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Joshua Colp
|
8d5fab9e52 |
Fix a crash caused by freeing a dialog directly instead of using dialog_unref.
(closes issue #16097) Reported by: steinwej Patches: no_RTP.diff uploaded by steinwej (license 841) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@228415 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Joshua Colp
|
c24d9bdb6d |
Merged revisions 228410 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r228410 | file | 2009-11-06 12:42:23 -0400 (Fri, 06 Nov 2009) | 14 lines Merged revisions 228409 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228409 | file | 2009-11-06 12:41:20 -0400 (Fri, 06 Nov 2009) | 7 lines Fix a bug caused by a partially invalid frame (from the jitterbuffer) passing through the Asterisk core. (closes issue #15560) Reported by: jvandal (closes issue #15709) Reported by: covici ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@228411 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
David Vossel
|
b7d3dbe86b |
Merged revisions 228339 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r228339 | dvossel | 2009-11-06 09:42:46 -0600 (Fri, 06 Nov 2009) | 12 lines Merged revisions 228338 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228338 | dvossel | 2009-11-06 09:41:41 -0600 (Fri, 06 Nov 2009) | 5 lines fixes crash in astfd.c (closes issue #15981) Reported by: slavon ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@228342 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
David Vossel
|
703dde7da8 |
Merged revisions 228268 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r228268 | dvossel | 2009-11-06 09:04:24 -0600 (Fri, 06 Nov 2009) | 9 lines fixes memory leak in func_audiohookinherit.c (closes issue #15394) Reported by: boroda Patches: bug15394_memoryleak_diff2.txt uploaded by dbrooks (license 790) Tested by: dbrooks, boroda ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@228271 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Jeff Peeler
|
da7e463795 |
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/1.6.0@228190 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
David Brooks
|
f56910147b |
Merged revisions 228145 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r228145 | dbrooks | 2009-11-05 13:34:50 -0600 (Thu, 05 Nov 2009) | 16 lines Merged revisions 228078 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228078 | dbrooks | 2009-11-05 12:59:41 -0600 (Thu, 05 Nov 2009) | 9 lines chan_misdn Asterisk 1.4.27-rc2 crash Crash related to chan_misdn connection. Patch submitted by gknispel_proformatique, tested by francesco_r. "I have many crash since i have upgraded to Asterisk 1.4.27-rc2. Attached a full bt." This patch zeros out an ast_frame. (closes issue #16041) Reported by: francesco_r ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@228146 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Jason Parker
|
6799f73d2e |
Merged revisions 228080 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r228080 | qwell | 2009-11-05 13:16:29 -0600 (Thu, 05 Nov 2009) | 15 lines Merged revisions 228079 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228079 | qwell | 2009-11-05 13:14:25 -0600 (Thu, 05 Nov 2009) | 8 lines Fix crash on VPB exception when no hardware is present. (closes issue #14970) Reported by: tzafrir Patches: vpb_exception.diff uploaded by tzafrir (license 46) Tested by: markwaters ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@228081 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Jeff Peeler
|
e25812a5b8 |
Merged revisions 227945 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r227945 | jpeeler | 2009-11-04 17:50:59 -0600 (Wed, 04 Nov 2009) | 21 lines Merged revisions 227944 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227944 | jpeeler | 2009-11-04 17:47:08 -0600 (Wed, 04 Nov 2009) | 14 lines Fix incorrect filename comparsion after monitor file change The logic to detect if a requested file is indeed a different file from the current file was incorrect. The main issue being confusion of the use of filename_base which was previously set without pathing information and then compared to another full path. Robust file comparison logic has been added to properly check if two files are the same even if symlinks are used. (closes issue #15313) Reported by: caspy Patches: 20091103__issue15313__1.4.diff.txt uploaded by jpeeler (license 325) but mostly tilghman's work ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@227946 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Matthew Nicholson
|
a101929502 |
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/1.6.0@227833 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Matthew Nicholson
|
e1a49d1c33 |
Modify the SDP parsing code to parse session and media level items separately.
With the new code, media level proprieties should no longer be confused with session level proprieties. This change also reorganizes some of the SDP parsing code which should make it easier to manage in the future. (closes issue #14994) Reported by: frawd git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@227763 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Joshua Colp
|
26ba777c94 |
Merged revisions 227739 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r227739 | file | 2009-11-04 15:26:19 -0400 (Wed, 04 Nov 2009) | 12 lines Merged revisions 227735 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227735 | file | 2009-11-04 15:25:37 -0400 (Wed, 04 Nov 2009) | 5 lines Fix a security issue where it may be possible for someone to execute a cross-site AJAX request exploit. (AST-2009-009) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@227743 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Joshua Colp
|
7cf3d5c6dd |
Merged revisions 227712 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r227712 | file | 2009-11-04 15:20:46 -0400 (Wed, 04 Nov 2009) | 12 lines Merged revisions 227700 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227700 | file | 2009-11-04 15:17:39 -0400 (Wed, 04 Nov 2009) | 5 lines Fix a security issue where sending a REGISTER with a differing username in the From URI and Authorization header would reveal whether it was valid or not. (AST-2009-008) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@227717 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Jason Parker
|
acf67fef9a |
Merged revisions 227372 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r227372 | qwell | 2009-11-03 13:59:46 -0600 (Tue, 03 Nov 2009) | 9 lines Fix some build issues on Solaris. (closes issue #14517) (SWP-109) Reported by: asgaroth Patches: bug_14517.diff uploaded by snuffy (license 35) Tested by: asgaroth, snuffy, dougm, qwell ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@227373 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Leif Madsen
|
08fc66db6a |
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/1.6.0@227369 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Leif Madsen
|
4344040d52 |
Merged revisions 227361 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r227361 | lmadsen | 2009-11-03 13:25:18 -0600 (Tue, 03 Nov 2009) | 11 lines Additional fixes to the extensions.conf.sample file. Update the extensions.conf.sample [stdexten] context so that we use the variable instead of requiring it to be passed explicitly. Also updated uses of the [stdexten] context throughout. (closes issue #15858) Reported by: pprindeville Patches: stdexten-context-update.txt uploaded by lmadsen (license 10) Tested by: pprindeville ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@227362 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Richard Mudgett
|
2bc3964ec9 |
Merged revisions 227275 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227275 | rmudgett | 2009-11-03 11:55:47 -0600 (Tue, 03 Nov 2009) | 4 lines Make sure the outgoing flag is cleared if a new channel fails to get created for outgoing calls. This is the relevant portion of asterisk/trunk -r226648 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@227278 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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David Vossel
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b6560b3586 |
Blocked revisions 227238 via svnmerge
........ r227238 | dvossel | 2009-11-03 11:12:52 -0600 (Tue, 03 Nov 2009) | 5 lines user.conf entries in SIP were not having their peer type set. (closes issue #16120) Reported by: jsmith ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@227242 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Joshua Colp
|
c1cc4d0833 |
Merged revisions 227167 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r227167 | file | 2009-11-03 11:37:08 -0400 (Tue, 03 Nov 2009) | 12 lines Merged revisions 227166 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227166 | file | 2009-11-03 11:36:16 -0400 (Tue, 03 Nov 2009) | 5 lines Fix a bug where an RPID header could be generated with a blank username in the URI. (closes issue #15909) Reported by: kobaz ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@227168 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Leif Madsen
|
5002373306 |
Merged revisions 227162 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r227162 | lmadsen | 2009-11-03 09:19:47 -0600 (Tue, 03 Nov 2009) | 7 lines Update extensions.conf.sample file to fix incorrect extensions. (closes issue #15857) Reported by: pprindeville Patches: stdexten.patch#2 uploaded by pprindeville (license 347) Tested by: pprindeville ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@227163 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Olle Johansson
|
2400c2cc61 |
Merged revisions 227091 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r227091 | oej | 2009-11-03 12:11:15 +0100 (Tis, 03 Nov 2009) | 15 lines Merged revisions 227088 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227088 | oej | 2009-11-03 11:29:59 +0100 (Tis, 03 Nov 2009) | 7 lines Use proper response code when violating Contact ACL's. https://reviewboard.asterisk.org/r/415/ Thanks kpfleming for a quick review. (EDVX-003) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@227102 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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David Brooks
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3264cb328c |
SIP channel name uniqueness
SIP channel names were supposed to be unique by way of a name suffix derived from the pointer to the channel's private data. Uniqueness was preserved on 32-bit systems, but not on 64-bit systems. This patch, as suggested by kpfleming, replaces this suffix with a simple incremented unsigned int. (closes issue #15152) Reported by: palbrecht Review: https://reviewboard.asterisk.org/r/420/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@226976 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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David Brooks
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42d27c4390 |
SIP channel name uniqueness
SIP channel names were supposed to be unique by way of a name suffix derived from the pointer to the channel's private data. Uniqueness was preserved on 32-bit systems, but not on 64-bit systems. This patch, as suggested by kpfleming, replaces this suffix with a simple incremented unsigned int. (closes issue #15152) Reported by: palbrecht Review: https://reviewboard.asterisk.org/r/420/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@226975 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Joshua Colp
|
ae224e9a75 |
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/1.6.0@226891 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Tilghman Lesher
|
29a2e11d90 |
Merged revisions 226812 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r226812 | tilghman | 2009-11-02 11:15:31 -0600 (Mon, 02 Nov 2009) | 15 lines Merged revisions 226811 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226811 | tilghman | 2009-11-02 11:14:20 -0600 (Mon, 02 Nov 2009) | 8 lines Don't allow two separate instances of safe_asterisk when restarting from the init script. (closes issue #14562) Reported by: davidw Patches: Initially 20091022__issue14562.diff.txt uploaded by tilghman (license 14) Modified to 20091030__Issue14562_diff.txt uploaded by davidw (license 780) Tested by: davidw ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@226813 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Joshua Colp
|
12b962ac19 |
Merged revisions 226532 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r226532 | file | 2009-10-29 15:13:42 -0300 (Thu, 29 Oct 2009) | 13 lines Merged revisions 226531 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226531 | file | 2009-10-29 15:11:26 -0300 (Thu, 29 Oct 2009) | 6 lines Add an option to enabling passing music on hold start and stop requests through instead of acting on them in chan_local. (closes issue #14709) Reported by: dimas ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@226533 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Leif Madsen
|
d1c3865674 |
Merged revisions 226384 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r226384 | lmadsen | 2009-10-28 15:11:07 -0500 (Wed, 28 Oct 2009) | 17 lines Merged revisions 226382 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226382 | lmadsen | 2009-10-28 15:06:13 -0500 (Wed, 28 Oct 2009) | 9 lines Update documentation in sip.conf.sample. Update the documentation in sip.conf.sample in order to make it more clear that directmedia/canreinvite do not cause Asterisk to ignore reINVITEs. It is only used to stop Asterisk from generating a reINVITE, but does not stop it from accepting them if necessary. (closes issue #15644) Reported by: lmadsen ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@226387 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Leif Madsen
|
0c583527bc |
Merged revisions 226378 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r226378 | lmadsen | 2009-10-28 14:50:00 -0500 (Wed, 28 Oct 2009) | 15 lines Merged revisions 226377 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226377 | lmadsen | 2009-10-28 14:48:29 -0500 (Wed, 28 Oct 2009) | 7 lines Update CALLINGSUBADDR channel variable documentation. (closes issue #15734) Reported by: alecdavis Patches: channelvariables.tex.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@226381 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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69b4362386 |
Merged revisions 226305 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r226305 | tilghman | 2009-10-28 13:04:05 -0500 (Wed, 28 Oct 2009) | 9 lines Merged revisions 226304 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226304 | tilghman | 2009-10-28 13:02:25 -0500 (Wed, 28 Oct 2009) | 2 lines Fix documentation (pointed out by TheDavidFactor on #-dev) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@226306 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
|
1c7cbc50a7 |
Merged revisions 226159 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r226159 | tilghman | 2009-10-27 15:22:07 -0500 (Tue, 27 Oct 2009) | 14 lines Merged revisions 226138 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226138 | tilghman | 2009-10-27 15:16:49 -0500 (Tue, 27 Oct 2009) | 7 lines Manager output is not always NULL-terminated, so force a NULL at the end of the filestream. (closes issue #15495) Reported by: pdf Patches: 20090916__issue15495.diff.txt uploaded by tilghman (license 14) Tested by: pdf ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@226167 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Tzafrir Cohen
|
4a76a51630 |
detect ARM Linux EABI OSARCH as linux-gnu instead of linux-gnueabi
* Set OSARCH to linux-gnu even if host_os is linux-gnueabi
* When checking if we are Linux, check OSARCH rather than host_os
The newer ARM ABI ("EABI") shows the OS name 'linux-gnueabi' rather than
'linux-gnu' . This patch sets OSARCH to be 'linux-gnu' even in such a case.
OSARCH is tested for the value of 'linux-gnu' in one or two places in the
tree. This patch also fixes the check libcap to check for $OSARCH rather
than $host_os .
See also: http://wiki.debian.org/ArmEabiPort
Merged revisions 225957 via svnmerge from
http://svn.digium.com/svn/asterisk/branches/1.4
Merged revisions 226018 via svnmerge from
http://svn.digium.com/svn/asterisk/trunk
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@226019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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17 years ago |
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Jeff Peeler
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7c19321b0d |
Blocked revisions 225912 via svnmerge
........ r225912 | jpeeler | 2009-10-26 14:40:26 -0500 (Mon, 26 Oct 2009) | 12 lines ACL check not present for verifying SIP INVITEs The ACL check in check_peer_ok was missing and has now been restored. The missing check allowed for calls to be made on prohibited networks where an ACL was defined in sip.conf and the allowguest option was set to off. See the AST security advisory below for more information. Merge code associated with AST-2009-007. (closes issue #16091) Reported by: thom4fun ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@225916 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Kevin P. Fleming
|
d84d683dbd |
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/1.6.0@225869 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Kevin P. Fleming
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d54d256d78 |
Merged revisions 225582 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r225582 | kpfleming | 2009-10-23 09:02:42 -0500 (Fri, 23 Oct 2009) | 17 lines Merged revisions 225581 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225581 | kpfleming | 2009-10-23 09:00:01 -0500 (Fri, 23 Oct 2009) | 10 lines Don't force menuselect.makeopts to be rebuilt on every build. For some reason the menuselect.makeopts file was listed as PHONY in the Makefile, resulting in 'make' needing to rebuild it for every build. This then resulted in the embedded module rules being rebuilt on every build, which can be slow and is unnecessary. This patch fixes the problem by properly allowing 'make' to know when the menuselect.makeopts file needs to be rebuilt (defining the proper dependencies). ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@225583 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Leif Madsen
|
9e5c03795d |
Merged revisions 225485 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r225485 | lmadsen | 2009-10-22 16:52:30 -0500 (Thu, 22 Oct 2009) | 19 lines Merged revisions 225484 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225484 | lmadsen | 2009-10-22 16:51:52 -0500 (Thu, 22 Oct 2009) | 11 lines Clean valgrind output by suppressing false errors. Update valgrind.txt documentation and add valgrind.supp file in order to allow those who are creating valgrind output to have less false errors in the logfile. (closes issue #16007) Reported by: atis Patches: valgrind.txt.diff uploaded by atis (license 242) asterisk2.supp uploaded by atis (license 242) Tested by: atis, amorsen ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@225486 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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b8b4e92f99 |
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/1.6.0@225361 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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David Vossel
|
1a0047d161 |
Merged revisions 225307 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r225307 | dvossel | 2009-10-21 16:58:46 -0500 (Wed, 21 Oct 2009) | 20 lines Merged revisions 225243 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225243 | dvossel | 2009-10-21 15:58:08 -0500 (Wed, 21 Oct 2009) | 13 lines IAX2: VNAK loop caused by signaling frames with no destination call number It is possible for the PBX thread to queue up signaling frames before a destination call number is received. This can result in signaling frames being sent out with no destination call number. Since recent versions of Asterisk require accurate destination callnumbers for all Full Frames, this can cause a VNAK loop to occur. To resolve this no signaling frames are sent until a destination callnumber is received, and destination call numbers are now only required for iax_pvt matching when the frame is an ACK. Review: https://reviewboard.asterisk.org/r/413/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@225311 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
David Vossel
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31c282574b |
Merged revisions 225033 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r225033 | dvossel | 2009-10-21 09:39:10 -0500 (Wed, 21 Oct 2009) | 27 lines Merged revisions 225032 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009) | 20 lines IAX/SIP shrinkcallerid option The shrinking of caller id removes '(', ' ', ')', non-trailing '.', and '-' from the string. This means values such as 555.5555 and test-test result in 555555 and testtest. There are instances, such as Skype integration, where a specific value is passed via caller id that must be preserved unmodified. This patch makes the shrinking of caller id optional in chan_sip and chan_iax in order to support such cases. By default this option is on to preserve previous expected behavior. (closes issue #15940) Reported by: dimas Patches: v2-15940.patch uploaded by dimas (license 88) 15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671) Tested by: dvossel Review: https://reviewboard.asterisk.org/r/408/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@225310 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Russell Bryant
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ffe8346471 |
Merged revisions 224932 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r224932 | russell | 2009-10-20 22:09:04 -0500 (Tue, 20 Oct 2009) | 12 lines Merged revisions 224931 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224931 | russell | 2009-10-20 21:59:54 -0500 (Tue, 20 Oct 2009) | 5 lines Isolate frames returned from a DSP instance or codec translator. The reasoning for these changes are the same as what I wrote in the commit message for rev 222878. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@224933 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
0f2eec5ab0 |
Merged revisions 224856 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r224856 | tilghman | 2009-10-20 17:09:07 -0500 (Tue, 20 Oct 2009) | 12 lines Merged revisions 224855 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224855 | tilghman | 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009) | 5 lines Pay attention to the return value of the manipulate function. While this looks like an optimization, it prevents a crash from occurring when used with certain audiohook callbacks (diagnosed with SVN trunk, backported to 1.4 to keep the source consistent across versions). ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@224857 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Joshua Colp
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2edc76e63c |
Merged revisions 224774 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r224774 | file | 2009-10-20 14:47:34 -0300 (Tue, 20 Oct 2009) | 12 lines Merged revisions 224773 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224773 | file | 2009-10-20 14:46:37 -0300 (Tue, 20 Oct 2009) | 5 lines Add support for relaying early media in the features attended transfer option. (closes issue #14828) Reported by: licedey ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@224775 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Kevin P. Fleming
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43805797d7 |
Merged revisions 224671 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r224671 | kpfleming | 2009-10-19 18:47:39 -0500 (Mon, 19 Oct 2009) | 14 lines Merged revisions 224670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224670 | kpfleming | 2009-10-19 18:44:07 -0500 (Mon, 19 Oct 2009) | 7 lines Correct timestamp calculations when RTP sample rates over 8kHz are used. While testing some endpoints that support 16kHz and 32kHz sample rates, some log messages were generated due to calc_rxstamp() computing timestamps in a way that produced odd results, so this patch sanitizes the result of the computations. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@224672 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Joshua Colp
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e23fe41857 |
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/1.6.0@224568 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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db344967da |
Merged revisions 224448 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r224448 | tilghman | 2009-10-18 19:05:56 -0500 (Sun, 18 Oct 2009) | 3 lines Allow ODBC storage to be queried with multiple mailboxes. This corrects an issue reported on the -users list. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@224449 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Jeff Peeler
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474544e477 |
fix typo, sorry
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@224337 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Jeff Peeler
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b597aca06a |
Merged revisions 224331 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r224331 | jpeeler | 2009-10-16 20:36:08 -0500 (Fri, 16 Oct 2009) | 20 lines Merged revisions 224330 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224330 | jpeeler | 2009-10-16 20:32:47 -0500 (Fri, 16 Oct 2009) | 13 lines Fix stale caller id data from being reported in AMI NewChannel event The problem here is that chan_dahdi is designed in such a way to set certain values in the dahdi_pvt only once. One of those such values is the configured caller id data in chan_dahdi.conf. For PRI, the configured caller id data could be overwritten during a call. Instead of saving the data and restoring, it was decided that for all non-analog channels it was simply best to not set the configured caller id in the first place and also clear it at the end of the call. (closes issue #15883) Reported by: jsmith ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@224332 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Richard Mudgett
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e2c27b5cd0 |
Merged revisions 224261 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r224261 | rmudgett | 2009-10-16 15:40:57 -0500 (Fri, 16 Oct 2009) | 25 lines Merged revisions 224260 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224260 | rmudgett | 2009-10-16 15:25:23 -0500 (Fri, 16 Oct 2009) | 18 lines Never released PRI channels when using Busy() or Congestion() dialplan apps. When the Busy() or Congestion() application is used towards ISDN (an ISDN progress is sent), the responding ISDN Disconnect or Release may contain the ISDN cause user busy or one of the congestion causes. In chan_dahdi.c these causes will only set the needbusy or needcongestion flags and not activate the softhangup procedure. Unfortunately only the latter can interrupt the endless wait loop of Busy()/Congestion(). Result: PRI channels staying in state busy for the rest of asterisk life or until the other end times out and forces the call to clear. (in issue 0014292) Reported by: tomaso Patches: disc_rel_userbusy.patch uploaded by tomaso (license 564) (This patch is unrelated to the issue.) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@224262 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Jeff Peeler
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7473efd695 |
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/1.6.0@224179 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Jeff Peeler
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e0dad69187 |
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/1.6.0@223833 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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David Vossel
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e61e6c49f8 |
Merged revisions 223756 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r223756 | dvossel | 2009-10-12 15:58:27 -0500 (Mon, 12 Oct 2009) | 5 lines Clarifies trunkmaxsize, trunkfreq, and trunkmtu iax2 options SWP-151 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@223759 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Kevin P. Fleming
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09fe2f94e6 |
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/1.6.0@223653 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |