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${ noResults }
18540 Commits (544acf5a22b6411f924edc45b77fbab6e2d3832b)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Jeff Peeler
|
c846ad7d7e |
Blocked revisions 207902 via svnmerge
........ r207902 | jpeeler | 2009-07-21 17:02:25 -0500 (Tue, 21 Jul 2009) | 2 lines Fix my_is_off_hook to check rxbits only for FXS signaling ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@207905 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Jeff Peeler
|
98b1700959 |
Merged revisions 207854 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r207854 | jpeeler | 2009-07-21 15:26:02 -0500 (Tue, 21 Jul 2009) | 16 lines Merged revisions 207827 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207827 | jpeeler | 2009-07-21 15:16:55 -0500 (Tue, 21 Jul 2009) | 9 lines Wait for wink before dialing when using E&M wink signaling There was already code for other signaling types in dahdi_handle_event to handle dialing if a dial operation dial string was present. Simply add SIG_EMWINK to the list. (closes issue #14434) Reported by: araasch ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@207862 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Jeff Peeler
|
0c86a3f3bf |
Revert r207638, this approach could potentially block for an unacceptable
amount of time. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@207785 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Mark Michelson
|
08a09ead1c |
Merged revisions 207723 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r207723 | mmichelson | 2009-07-21 09:29:40 -0500 (Tue, 21 Jul 2009) | 11 lines Merged revisions 207714 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207714 | mmichelson | 2009-07-21 09:26:00 -0500 (Tue, 21 Jul 2009) | 5 lines Document default timeout for AMI originations. AST-224 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@207727 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Kevin P. Fleming
|
c049643f5f |
Merged revisions 207680 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r207680 | kpfleming | 2009-07-21 08:28:04 -0500 (Tue, 21 Jul 2009) | 18 lines Merged revisions 207647 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul 2009) | 12 lines Ensure that user-provided CFLAGS and LDFLAGS are honored. This commit changes the build system so that user-provided flags (in ASTCFLAGS and ASTLDFLAGS) are supplied to the compiler/linker *after* all flags provided by the build system itself, so that the user can effectively override the build system's flags if desired. In addition, ASTCFLAGS and ASTLDFLAGS can now be provided *either* in the environment before running 'make', or as variable assignments on the 'make' command line. As a result, the use of COPTS and LDOPTS is no longer necessary, so they are no longer documented, but are still supported so as not to break existing build systems that supply them when building Asterisk. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@207685 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Jeff Peeler
|
539bfbfb0e |
Wait for wink before dialing when using E&M wink signaling
This patch adds a new dahdi_wait function to specifically wait for the wink event. If the wink is not eventually received the channel is hung up. (closes issue #14434) Reported by: araasch Patches: emwinkmod uploaded by araasch (license 693) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@207638 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Mark Michelson
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725a152b81 |
Blocked revisions 207551 via svnmerge
........ r207551 | mmichelson | 2009-07-20 18:08:56 -0500 (Mon, 20 Jul 2009) | 3 lines Okay, that didn't fix the crash. It didn't really do anything useful. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@207552 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Mark Michelson
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def6d8d83c |
Blocked revisions 207522 via svnmerge
........ r207522 | mmichelson | 2009-07-20 17:13:34 -0500 (Mon, 20 Jul 2009) | 7 lines Initialize connected line instance when doing a directed pickup. This helps to prevent a crash which may occur due to our freeing garbage due to a struct being uninitialized. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@207524 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Mark Michelson
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82849e55d0 |
Merged revisions 207424 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r207424 | mmichelson | 2009-07-20 14:48:12 -0500 (Mon, 20 Jul 2009) | 39 lines Merged revisions 207423 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207423 | mmichelson | 2009-07-20 14:39:59 -0500 (Mon, 20 Jul 2009) | 33 lines Answer video SDP offers properly when videosupport is not enabled. Copied from Review board: In issue 12434, the reporter describes a situation in which audio and video is offered on the call, but because videosupport is disabled in sip.conf, Asterisk gives no response at all to the video offer. According to RFC 3264, all media offers should have a corresponding answer. For offers we do not intend to actually reply to with meaningful values, we should still reply with the port for the media stream set to 0. In this patch, we take note of what types of media have been offered and save the information on the sip_pvt. The SDP in the response will take into account whether media was offered. If we are not otherwise going to answer a media offer, we will insert an appropriate m= line with the port set to 0. It is important to note that this patch is pretty much a bandage being applied to a broken bone. The patch *only* helps for situations where video is offered but videosupport is disabled and when udptl_pt is disabled but T.38 is offered. Asterisk is not guaranteed to respond to every media offer. Notable cases are when multiple streams of the same type are offered. The 2 media stream limit is still present with this patch, too. In trunk and the 1.6.X branches, things will be a bit different since Asterisk also supports text in SDPs as well. (closes issue #12434) Reported by: mnnojd Review: https://reviewboard.asterisk.org/r/311 Review: https://reviewboard.asterisk.org/r/313 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@207523 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
David Vossel
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1b6019ec28 |
Blocked revisions 207484 via svnmerge
........ r207484 | dvossel | 2009-07-20 15:45:26 -0500 (Mon, 20 Jul 2009) | 16 lines reg->username is parsed only once on sip reload The registration string can contain an expanded user portion of the form user@domain. This expanded user portion was stored in reg->username and parsed each time there is a registration refresh. Now, the domain portion of the user is parsed and stored separately in the regdomain field. (closes issue #14331) Reported by: Nick_Lewis Patches: chan_sip.c.domainparse3.patch uploaded by Nick (license 657) Tested by: Nick_Lewis, dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@207486 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Russell Bryant
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801e732c24 |
Merged revisions 207361 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r207361 | russell | 2009-07-20 11:36:15 -0500 (Mon, 20 Jul 2009) | 16 lines Merged revisions 207360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207360 | russell | 2009-07-20 11:26:24 -0500 (Mon, 20 Jul 2009) | 9 lines Only do the chan->fdno check in ast_read() in a developer build. I changed this check to only happen in a dev-mode build. I also added a comment explaining what is going on. I also made it so that detection of this situation does not affect ast_read() operation. (closes issue #14723) Reported by: seadweller ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@207364 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Tilghman Lesher
|
57d354321c |
Merged revisions 207317 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r207317 | tilghman | 2009-07-17 23:16:44 -0500 (Fri, 17 Jul 2009) | 3 lines Flag field in wrong position. Reported by "Hoggins!" on asterisk-dev list. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@207327 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Richard Mudgett
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e0e14ba72a |
Merged revisions 145293,158010 from
https://origsvn.digium.com/svn/asterisk/branches/1.4 to make merging easier. These changes are already on trunk. ................ r145293 | rmudgett | 2008-09-30 18:55:24 -0500 (Tue, 30 Sep 2008) | 54 lines channels/chan_misdn.c channels/misdn/isdn_lib.c * Miscellaneous other fixes from trunk to make merging easier later. ........ r145200 | rmudgett | 2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) | 7 lines * Miscellaneous formatting changes to make v1.4 and trunk more merge compatible in the mISDN area. channels/chan_misdn.c * Eliminated redundant code in cb_events() EVENT_SETUP ........ r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008) | 9 lines improved helptext of misdn_set_opt. ........ r142181 | rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line Cleaned up comment ........ r138738 | rmudgett | 2008-08-18 16:07:28 -0500 (Mon, 18 Aug 2008) | 30 lines channels/chan_misdn.c * Made bearer2str() use allowed_bearers_array[] * Made use the causes.h defines instead of hardcoded numbers. * Made use Asterisk presentation indicator values if either of the mISDN presentation or screen options are negative. * Updated the misdn_set_opt application option descriptions. * Renamed the awkward Caller ID presentation misdn_set_opt application option value not_screened to restricted. Deprecated the not_screened option value. channels/misdn/isdn_lib.c * Made use the causes.h defines instead of hardcoded numbers. * Fixed some spelling errors and typos. * Added all defined facility code strings to fac2str(). channels/misdn/isdn_lib.h * Added doxygen comments to struct misdn_bchannel. channels/misdn/isdn_lib_intern.h * Added doxygen comments to struct misdn_stack. channels/misdn_config.c configs/misdn.conf.sample * Updated the mISDN presentation and screen parameter descriptions. doc/misdn.txt (doc/tex/misdn.tex) * Updated the misdn_set_opt application option descriptions. * Fixed some spelling errors and typos. ................ r158010 | rmudgett | 2008-11-19 19:46:09 -0600 (Wed, 19 Nov 2008) | 9 lines Merged revision 157977 from https://origsvn.digium.com/svn/asterisk/team/group/issue8824 ........ Fixes JIRA ABE-1726 The dial extension could be empty if you are using MISDN_KEYPAD to control ISDN provider features. ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@207315 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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ddfc4070c8 |
Merged revisions 207255 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r207255 | tilghman | 2009-07-17 17:29:50 -0500 (Fri, 17 Jul 2009) | 2 lines Add flag here, too (as requested by jsmith) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@207257 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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a274d055a5 |
Merged revisions 207224 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r207224 | tilghman | 2009-07-17 17:04:43 -0500 (Fri, 17 Jul 2009) | 2 lines Document the "flag" field in the voicemessages table. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@207226 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Jeff Peeler
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3b08cf189a |
Merged revisions 207156 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r207156 | jpeeler | 2009-07-17 14:37:38 -0500 (Fri, 17 Jul 2009) | 14 lines Merged revisions 207155 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207155 | jpeeler | 2009-07-17 14:36:19 -0500 (Fri, 17 Jul 2009) | 7 lines Fix format specifier to print out an unsigned long long. Yep, it's even ifdefed out code. But it made it to the RR list... (closes issue #14726) Reported by: lmadsen ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@207159 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Jeff Peeler
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53533451ae |
Merged revisions 207095 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r207095 | jpeeler | 2009-07-17 14:16:35 -0500 (Fri, 17 Jul 2009) | 2 lines Update some missing allowed options for overlapdial ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@207104 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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05f171d7a3 |
Merged revisions 207029 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r207029 | dvossel | 2009-07-17 12:51:44 -0500 (Fri, 17 Jul 2009) | 6 lines sip option flags handled incorrectly (closes issue #15376) Reported by: Takehiko Ooshima Tested by: dvossel, Takehiko_Ooshima ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@207030 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Jeff Peeler
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9b65ab2be5 |
Blocked revisions 206998 via svnmerge
........ r206998 | jpeeler | 2009-07-17 12:02:44 -0500 (Fri, 17 Jul 2009) | 14 lines Fix segfault in sig_analog when using callwaiting, respect callwaiting options Sig_analog handles allocating the sub channel for callwaiting, so no longer try to do it in chan_dahdi. Modified analog_alloc_sub to only mark the sub as allocated upon success of the alloc_sub callback, which was responsible for the segfault. Also, the callwaiting and callwaitingcallerid options were being unconditionally set to true. Now, the options are properly set from chan_dahdi.conf. (closes issue #15508) Reported by: elguero Tested by: elguero ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@207001 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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39bbea3a6a |
Merged revisions 206939 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r206939 | dvossel | 2009-07-17 11:13:22 -0500 (Fri, 17 Jul 2009) | 20 lines Merged revisions 206938 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206938 | dvossel | 2009-07-17 11:05:06 -0500 (Fri, 17 Jul 2009) | 14 lines SIP incorrect From: header information when callpres is prohib Some ITSP make use of the "Anonymous" display name to detect a requirement to withhold caller id across the PSTN. This does not work if the display name is "Unknown". (closes issue #14465) Reported by: Nick_Lewis Patches: chan_sip.c-callerpres.patch uploaded by Nick (license 657) chan_sip.c-callerpres_trunk.patch uploaded by dvossel (license 671) Tested by: Nick_Lewis, dvossel ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@206940 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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693a287cb5 |
Merged revisions 206877 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r206877 | dvossel | 2009-07-16 16:45:14 -0500 (Thu, 16 Jul 2009) | 6 lines TIMEOUT(absolute) returned negative value. (closes issue #15513) Reported by: ys ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@206878 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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8ff8696c38 |
Merged revisions 206873 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r206873 | dvossel | 2009-07-16 16:33:51 -0500 (Thu, 16 Jul 2009) | 12 lines Merged revisions 206872 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206872 | dvossel | 2009-07-16 16:33:19 -0500 (Thu, 16 Jul 2009) | 6 lines error in iax.conf related IP-based access control (closes issue #15518) Reported by: pkempgen ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@206874 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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8f32f97da1 |
Merged revisions 206868 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r206868 | dvossel | 2009-07-16 16:25:22 -0500 (Thu, 16 Jul 2009) | 14 lines Merged revisions 206867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206867 | dvossel | 2009-07-16 16:24:16 -0500 (Thu, 16 Jul 2009) | 8 lines avoid segfault caused by user error If the CALLERPRES() dialplan function is set to nothing, a segfault occurs. This is user error to begin with, but I'd rather see a cli warning message than have Asterisk crash on me. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@206869 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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88cfc407c2 |
Merged revisions 206808 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r206808 | tilghman | 2009-07-16 11:51:05 -0500 (Thu, 16 Jul 2009) | 13 lines Merged revisions 206807 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206807 | tilghman | 2009-07-16 11:27:35 -0500 (Thu, 16 Jul 2009) | 6 lines Fix a memory leak. (closes issue #15517) Reported by: adomjan Patches: func_realtime.c-ast_variable_destroy.diff uploaded by adomjan (license 487) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@206811 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Jeff Peeler
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0a02e6053d |
Blocked revisions 206767 via svnmerge
........ r206767 | jpeeler | 2009-07-15 17:02:55 -0500 (Wed, 15 Jul 2009) | 10 lines The dialing flag was mistakingly removed from sig_pri. This readds the proper setting of the flag and is really a continuation of r205731. The flag was being set properly in sig_analog, but use of the newly added set_dialing callback allowed for some simplification in chan_dahdi. (closes issue #15486) Reported by: rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@206773 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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0cc5fb40f0 |
Merged revisions 206768 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r206768 | dvossel | 2009-07-15 17:04:13 -0500 (Wed, 15 Jul 2009) | 8 lines Session timer were not activated if Supported header field in INVITE had both "timer" and other options. (closes issue #15403) Reported by: makoto Patches: sip-session-timer.patch uploaded by makoto (license ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@206770 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Richard Mudgett
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606324d08e |
Merged revisions 206707 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r206707 | rmudgett | 2009-07-15 16:14:41 -0500 (Wed, 15 Jul 2009) | 33 lines Merged revisions 206706 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r206706 | rmudgett | 2009-07-15 15:44:55 -0500 (Wed, 15 Jul 2009) | 26 lines Merged revision 206700 from https://origsvn.digium.com/svn/asterisk/be/branches/C.2-... .......... Fixed chan_misdn crash because mISDNuser library is not thread safe. With Asterisk the mISDNuser library is driven by two threads concurrently: 1. channels/misdn/isdn_lib.c::manager_event_handler() 2. channels/misdn/isdn_lib.c::misdn_lib_isdn_event_catcher() Calls into the library are done concurrently and recursively from isdn_lib.c. Both threads can fiddle with the master/child layer3_proc_t lists. One thread may traverse the list when the other interrupts it and then removes the list element which the first thread was currently handling. This is exactly what caused the crash. About 60 calls were needed to a Gigaset CX475 before it occurred once. This patch adds locking when calling into the mISDNuser library. This also fixes some cb_log calls with wrong port parameter. JIRA ABE-1913 Patches: misdn-locking.patch (Modified with mostly cosmetic changes) .......... ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@206765 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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dbd935dc5e |
Merged revisions 206702 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r206702 | dvossel | 2009-07-15 15:20:01 -0500 (Wed, 15 Jul 2009) | 10 lines callerid(num) is wrong when username is missing A domain only sip uri <sip:123.123.123.123> would return 123.123.123.123 as callid num. Now, if the username is missing from a uri, the callerid num field is left empty. (closes issue #15476) Reported by: viraptor ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@206703 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Sean Bright
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01cf201708 |
Merged revisions 206636 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r206636 | seanbright | 2009-07-15 12:00:24 -0400 (Wed, 15 Jul 2009) | 9 lines Merged revisions 206635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206635 | seanbright | 2009-07-15 11:57:51 -0400 (Wed, 15 Jul 2009) | 1 line Only print debug info in codec_dahdi if we are asking for it. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@206639 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Jeff Peeler
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ac818f19d4 |
Blocked revisions 206603 via svnmerge
........ r206603 | jpeeler | 2009-07-14 15:38:56 -0500 (Tue, 14 Jul 2009) | 1 line fix a typo in sample config file for option change ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@206604 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Jeff Peeler
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9fa4e02720 |
Blocked revisions 206566 via svnmerge
........ r206566 | jpeeler | 2009-07-14 15:01:10 -0500 (Tue, 14 Jul 2009) | 8 lines Restore some missing functionality to sig_analog. The main purpose of this commit is to restore missing functionality present in the ss_thread before all the sig related work was done. Two of the biggest missing things were distinctive ring detection and cid handling for V23. fxsoffhookstate and associated mwi variables have been moved inside sig_analog as they were not being set properly as well. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@206602 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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d4a4a57735 |
Merged revisions 206567 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r206567 | tilghman | 2009-07-14 15:14:45 -0500 (Tue, 14 Jul 2009) | 6 lines Document all meetme realtime fields, and in the process, make some field lengths more consistent. (closes issue #15493) Reported by: lasko Patches: meetme.diff uploaded by lasko (license 833) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@206598 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Richard Mudgett
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552f1cdfee |
Merged revisions 206489 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r206489 | rmudgett | 2009-07-14 12:01:48 -0500 (Tue, 14 Jul 2009) | 35 lines Merged revisions 206487 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206487 | rmudgett | 2009-07-14 11:44:47 -0500 (Tue, 14 Jul 2009) | 28 lines Fixes several call transfer issues with chan_misdn. * issue #14355 - Crash if attempt to transfer a call to an application. Masquerade the other pair of the four asterisk channels involved in the two calls. The held call already must be a bridged call (not an applicaton) or it would have been rejected. * issue #14692 - Held calls are not automatically cleared after transfer. Allow the core to initate disconnect of held calls to the ISDN port. This also fixes a similar case where the party on hold hangs up before being transferred or taken off hold. * JIRA ABE-1903 - Orphaned held calls left in music-on-hold. Do not simply block passing the hangup event on held calls to asterisk core. * Fixed to allow held calls to be transferred to ringing calls. Previously, held calls could only be transferred to connected calls. * Eliminated unused call states to simplify hangup code. * Eliminated most uses of "holded" because it is not a word. (closes issue #14355) (closes issue #14692) Reported by: sodom Patches: misdn_xfer_v14_r205839.patch uploaded by rmudgett (license 664) Tested by: rmudgett ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@206565 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Mark Michelson
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ea1d53f20e |
Blocked revisions 206490 via svnmerge
........ r206490 | mmichelson | 2009-07-14 12:03:58 -0500 (Tue, 14 Jul 2009) | 3 lines I AM A TERRIBLE PERSON ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@206491 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Mark Michelson
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95731c1c8b |
Blocked revisions 206455 via svnmerge
........ r206455 | mmichelson | 2009-07-14 11:09:38 -0500 (Tue, 14 Jul 2009) | 9 lines Reset the sentringing indication when redirects occur. If a redirecting control frame is processed or a call forward occurs, we need to reset the sentringing flag so that we can send another ringing indication to the phone that may contain a connected line update. AST-164 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@206456 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Russell Bryant
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b64abeeeb1 |
Merged revisions 206386 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r206386 | russell | 2009-07-14 09:51:44 -0500 (Tue, 14 Jul 2009) | 20 lines Merged revisions 206385 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r206385 | russell | 2009-07-14 09:48:00 -0500 (Tue, 14 Jul 2009) | 13 lines Merged revisions 206384 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r206384 | russell | 2009-07-14 09:45:47 -0500 (Tue, 14 Jul 2009) | 6 lines Ensure apathetic replies are sent out on the proper socket. chan_iax2 supports multiple address bindings. The send_apathetic_reply() function did not attempt to send its response on the same socket that the incoming message came in on. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@206389 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Richard Mudgett
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06ddc261f9 |
Merged revisions 206341 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r206341 | rmudgett | 2009-07-13 19:48:59 -0500 (Mon, 13 Jul 2009) | 11 lines Merged revisions 206284 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206284 | rmudgett | 2009-07-13 19:17:28 -0500 (Mon, 13 Jul 2009) | 4 lines Fix some memory leaks in chan_misdn. JIRA ABE-1911 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@206373 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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ce7719531e |
Merged revisions 206280 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r206280 | dvossel | 2009-07-13 18:26:51 -0500 (Mon, 13 Jul 2009) | 9 lines dns lookup of peername rather than peer's host in transmit_register() (closes issue #15052) Reported by: fsantulli Patches: chan_sip_bug_15052_[20090626204511].patch uploaded by fsantulli (license 818) Tested by: fsantulli ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@206281 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Sean Bright
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2b859c28e9 |
Blocked revisions 206225 via svnmerge
........ r206225 | seanbright | 2009-07-13 14:46:47 -0400 (Mon, 13 Jul 2009) | 5 lines Make sure that since we are passing -c to asterisk that we have a console. Without this line, Asterisk will busy-loop trying to read and write to /dev/null (woops... my bad). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@206226 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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474204522c |
Merged revisions 206185 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r206185 | tilghman | 2009-07-13 11:23:07 -0500 (Mon, 13 Jul 2009) | 2 lines Remove reference to non-existent help file ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@206187 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Russell Bryant
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b4f77e5d86 |
Blocked revisions 206021,206049 via svnmerge
........ r206021 | russell | 2009-07-11 14:15:03 -0500 (Sat, 11 Jul 2009) | 27 lines Add an API for reporting security events, and a security event logging module. This commit introduces the security events API. This API is to be used by Asterisk components to report events that have security implications. A simple example is when a connection is made but fails authentication. These events can be used by external tools manipulate firewall rules or something similar after detecting unusual activity based on security events. Inside of Asterisk, the events go through the ast_event API. This means that they have a binary encoding, and it is easy to write code to subscribe to these events and do something with them. One module is provided that is a subscriber to these events - res_security_log. This module turns security events into a parseable text format and sends them to the "security" logger level. Using logger.conf, these log entries may be sent to a file, or to syslog. One service, AMI, has been fully updated for reporting security events. AMI was chosen as it was a fairly straight forward service to convert. The next target will be chan_sip. That will be more complicated and will be done as its own project as the next phase of security events work. For more information on the security events framework, see the documentation generated from doc/tex/. "make asterisk.pdf" Review: https://reviewboard.asterisk.org/r/273/ ........ r206049 | russell | 2009-07-11 14:30:19 -0500 (Sat, 11 Jul 2009) | 1 line note the security events API in CHANGES ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@206050 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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15bdfdb4cf |
Merged revisions 205985 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r205985 | dvossel | 2009-07-10 16:42:10 -0500 (Fri, 10 Jul 2009) | 16 lines SIP register not using peer's outbound proxy If callbackextension is defined for a peer it successfully causes a registration to occur, but the registration ignores the outboundproxy settings for the peer. This patch allows the peer to be passed to obproxy_get() in transmit_register(). (closes issue #14344) Reported by: Nick_Lewis Patches: callbackextension_peer_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel Review: https://reviewboard.asterisk.org/r/294/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@205986 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Kevin P. Fleming
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e6d43d45e1 |
Merged revisions 205939 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r205939 | kpfleming | 2009-07-10 13:44:09 -0500 (Fri, 10 Jul 2009) | 1 line Update comments about the level of T.38 support in Asterisk. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@205942 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Mark Michelson
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519e8360fc |
Merged revisions 205878 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r205878 | mmichelson | 2009-07-10 12:39:57 -0500 (Fri, 10 Jul 2009) | 30 lines Merged revisions 205877 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r205877 | mmichelson | 2009-07-10 12:39:13 -0500 (Fri, 10 Jul 2009) | 23 lines Merged revisions 205776 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205776 | mmichelson | 2009-07-10 10:56:45 -0500 (Fri, 10 Jul 2009) | 16 lines Merged revisions 205775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul 2009) | 10 lines Ensure that outbound NOTIFY requests are properly routed through stateful proxies. With this change, we make note of Record-Route headers present in any SUBSCRIBE request that we receive so that our outbound NOTIFY requests will have the proper Route headers in them. (closes issue #14725) Reported by: ibc ........ ................ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@205882 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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59120cbfd7 |
Merged revisions 205840 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r205840 | dvossel | 2009-07-10 11:42:04 -0500 (Fri, 10 Jul 2009) | 37 lines Merged revisions 205804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205804 | dvossel | 2009-07-10 11:23:59 -0500 (Fri, 10 Jul 2009) | 31 lines SIP registration auth loop caused by stale nonce If an endpoint sends two registration requests in a very short period of time with the same nonce, both receive 401 responses from Asterisk, each with a different nonce (the second 401 containing the current nonce and the first one being stale). If the endpoint responds to the first 401, it does not match the current nonce so Asterisk sends a third 401 with a newly generated nonce (which updates the current nonce)... Now if the endpoint responds to the second 401, it does not match the current nonce either and Asterisk sends a fourth 401 with a newly generated nonce... This loop goes on and on. There appears to be a simple fix for this. If the nonce from the request does not match our nonce, but is a good response to a previous nonce, instead of sending a 401 with a newly generated nonce, use the current one instead. This breaks the loop as the nonce is not updated until a response is received. Additional logic has been added to make sure no nonce can be responded to twice though. (closes issue #15102) Reported by: Jamuel Patches: patch-bug_0015102 uploaded by Jamuel (license 809) nonce_sip.diff uploaded by dvossel (license 671) Tested by: Jamuel Review: https://reviewboard.asterisk.org/r/289/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@205841 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Kevin P. Fleming
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ae1252bf1e |
Merged revisions 205780 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r205780 | kpfleming | 2009-07-10 11:00:44 -0500 (Fri, 10 Jul 2009) | 11 lines Eliminate extraneous LOG_DEBUG messages generated by app_fax. The transmit_audio() and transmit_t38() functions in app_fax have processing loops that are supposed to wait for frames to arrive on the channel and then handle them, but they also have short timeouts so that the loops can have watchdog timers and do other required processing. This commit changes the loops to not actually call ast_read() and attempt to process the returned frame unless a frame actually arrived, eliminating hundreds of LOG_DEBUG messages and slightly improving performance. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@205781 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Mark Michelson
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0d092a6558 |
Merged revisions 205776 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r205776 | mmichelson | 2009-07-10 10:56:45 -0500 (Fri, 10 Jul 2009) | 16 lines Merged revisions 205775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205775 | mmichelson | 2009-07-10 10:51:36 -0500 (Fri, 10 Jul 2009) | 10 lines Ensure that outbound NOTIFY requests are properly routed through stateful proxies. With this change, we make note of Record-Route headers present in any SUBSCRIBE request that we receive so that our outbound NOTIFY requests will have the proper Route headers in them. (closes issue #14725) Reported by: ibc ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@205779 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Kevin P. Fleming
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14483d9fc3 |
Merged revisions 205770 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r205770 | kpfleming | 2009-07-10 10:28:11 -0500 (Fri, 10 Jul 2009) | 12 lines Fix some remaining T.38 negotiation problems in app_fax. Revision 205696 did not quite fix all the issues with the T.38 negotiation changes and app_fax; this patch corrects them, along with a couple of other minor issues. (closes issue #15480) Reported by: dimas Patches: test2-15480.patch uploaded by dimas (license 88) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@205773 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Richard Mudgett
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64989a9206 |
Merged revisions 205728 via svn merge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205728 | rmudgett | 2009-07-09 18:37:53 -0500 (Thu, 09 Jul 2009) | 21 lines No audio on calls from Asterisk to various ISDN devices until DTMF sent by caller. Add missing clearing of the dialing flag when the ISDN call is CONNECTED. (i.e. When libpri generates the event PRI_EVENT_ANSWER.) (closes issue #15420) Reported by: scottbmilne Patches: bug15420-1.4.25.1-diff2.txt uploaded by alecdavis (license 585) Tested by: scottbmilne, alecdavis (closes issue #15416) Reported by: avinoash (closes issue #15389) Reported by: alecdavis This patch should also fix the following issue: (issue #15205) Reported by: vinsik ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@205731 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Kevin P. Fleming
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b4ad95ecf3 |
Merged revisions 205696 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r205696 | kpfleming | 2009-07-09 16:20:23 -0500 (Thu, 09 Jul 2009) | 16 lines Repair ability of SendFAX/ReceiveFAX to respond to T.38 switchover. Recent changes in T.38 negotiation in Asterisk caused these applications to not respond when the other endpoint initiated a switchover to T.38; this resulted in the T.38 switchover failing, and the FAX attempt to be made using an audio connection, instead of T.38 (which would usually cause the FAX to fail completely). This patch corrects this problem, and the applications will now correctly respond to the T.38 switchover request. In addition, the response will include the appopriate T.38 session parameters based on what the other end offered and what our end is capable of. (closes issue #14849) Reported by: afosorio ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@205699 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |