asterisk

Commit Graph

Author	SHA1	Message	Date
Kevin P. Fleming	bbf4a08fe7	Merged revisions 210190 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r210190 \| kpfleming \| 2009-08-03 15:48:48 -0500 (Mon, 03 Aug 2009) \| 11 lines Rename 'canreinvite' option to 'directmedia', with backwards compatibility. It is clear from multiple mailing list, forum, wiki and other sorts of posts that users don't really understand the effects that the 'canreinvite' config option actually has, and that in some cases they think that setting it to 'no' will actually cause various other features (T.38, MOH, etc.) to not work properly, when in fact this is not the case. This patch changes the proper name of the option to what it should have been from the beginning ('directmedia'), but preserves backwards compatibility for existing configurations. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@210191 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	0cad342de7	Merged revisions 209760-209761 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r209760 \| kpfleming \| 2009-07-31 20:03:07 -0500 (Fri, 31 Jul 2009) \| 13 lines Merged revisions 209759 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r209759 \| kpfleming \| 2009-07-31 19:52:00 -0500 (Fri, 31 Jul 2009) \| 7 lines Minor changes inspired by testing with latest GCC. The latest GCC (what will become 4.5.x) has a few new warnings, that in these cases found some either downright buggy code, or at least seriously poorly designed code that could be improved. ........ ................ r209761 \| kpfleming \| 2009-07-31 20:04:06 -0500 (Fri, 31 Jul 2009) \| 1 line Revert accidental Makefile change. ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@209816 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Brooks	503bb8de9c	Merged revisions 209554 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209554 \| dbrooks \| 2009-07-30 11:07:05 -0500 (Thu, 30 Jul 2009) \| 6 lines Fixes numerous spelling errors. Patch submitted by alecdavis. (closes issue #15595) Reported by: alecdavis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@209594 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	b4e1b718bb	Merged revisions 209516 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209516 \| mmichelson \| 2009-07-30 09:38:21 -0500 (Thu, 30 Jul 2009) \| 8 lines Fix a crash that can result if text codecs are allowed but textsupport is disabled. (closes issue #15596) Reported by: fabled Patches: sip-red.patch uploaded by fabled (license 448) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@209518 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Brooks	4921462e88	Merged revisions 209098 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r209098 \| dbrooks \| 2009-07-27 11:33:50 -0500 (Mon, 27 Jul 2009) \| 6 lines Fixing typos. Replaces "recieved" with "received" and "initilize" with "initialize" (closes issue #15571) Reported by: alecdavis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@209234 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	2d99be05a5	Merged revisions 208924 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208924 \| jpeeler \| 2009-07-26 20:20:37 -0500 (Sun, 26 Jul 2009) \| 9 lines Merged revisions 208923 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208923 \| jpeeler \| 2009-07-26 20:18:31 -0500 (Sun, 26 Jul 2009) \| 2 lines Fix logic errors from 208746 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@208927 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	b42e55db37	Merged revisions 208749 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208749 \| jpeeler \| 2009-07-25 01:23:18 -0500 (Sat, 25 Jul 2009) \| 13 lines Merged revisions 208746 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208746 \| jpeeler \| 2009-07-25 01:19:50 -0500 (Sat, 25 Jul 2009) \| 7 lines Fix compiling under dev-mode with gcc 4.4.0. Mostly trivial changes, but I did not know of any other way to fix the "dereferencing type-punned pointer will break strict-aliasing rules" error without creating a tmp variable in chan_skinny. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@208755 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	17fcb39e1a	Merged revisions 208588 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208588 \| mmichelson \| 2009-07-24 13:31:04 -0500 (Fri, 24 Jul 2009) \| 16 lines Merged revisions 208587 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208587 \| mmichelson \| 2009-07-24 13:26:50 -0500 (Fri, 24 Jul 2009) \| 10 lines Only send a BYE when hanging up a channel that is up. For cases where Asterisk sends an INVITE and receives a non 2XX final response, Asterisk would follow the INVITE transaction by immediately sending a BYE, which was unnecessary. (closes issue #14575) Reported by: chris-mac ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@208591 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	b7e75a4a56	Merged revisions 208548 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208548 \| kpfleming \| 2009-07-24 10:02:53 -0500 (Fri, 24 Jul 2009) \| 8 lines Resolve a T.38 negotiation issue left over from the udptl-updates merge. The udptl-updates branch that was merged yesterday failed to properly send back T.38 SDP responses with the correct error correction mode, if the incoming SDP from the other end caused us to change error correction modes. This patch corrects that situation. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@208551 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	03d73280b3	Merged revisions 208464 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r208464 \| kpfleming \| 2009-07-23 16:57:24 -0500 (Thu, 23 Jul 2009) \| 46 lines Rework of T.38 negotiation and UDPTL API to address interoperability problems Over the past couple of months, a number of issues with Asterisk negotiating (and successfully completing) T.38 sessions with various endpoints have been found. This patch attempts to address many of them, primarily focused around ensuring that the endpoints' MaxDatagram size is honored, and in addition by ensuring that T.38 session parameter negotiation is performed correctly according to the ITU T.38 Recommendation. The major changes here are: 1) T.38 applications in Asterisk (app_fax) only generate/receive IFP packets, they do not ever work with UDPTL packets. As a result of this, they cannot be allowed to generate packets that would overflow the other endpoints' MaxDatagram size after the UDPTL stack adds any error correction information. With this patch, the application is told the maximum IFP size it can generate, based on a calculation using the far end MaxDatagram size and the active error correction mode on the T.38 session. The same is true for sending our MaxDatagram size to the remote endpoint; it is computed from the value that the application says it can accept (for a single IFP packet) combined with the active error correction mode. 2) All treatment of T.38 session parameters as 'capabilities' in chan_sip has been removed; these parameters are not at all like audio/video stream capabilities. There are strict rules to follow for computing an answer to a T.38 offer, and chan_sip now follows those rules, using the desired parameters from the application (or channel) that wants to accept the T.38 negotiation. 3) chan_sip now stores and forwards ast_control_t38_parameters structures for tracking 'our' and 'their' T.38 session parameters; this greatly simplifies negotiation, especially for pass-through calls. 4) Since T.38 negotiation without specifying parameters or receiving the final negotiated parameters is not very worthwhile, the AST_CONTROL_T38 control frame has been removed. A note has been added to UPGRADE.txt about this removal, since any out-of-tree applications that use it will no longer function properly until they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review: https://reviewboard.asterisk.org/r/310/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@208501 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	5e3c4e498f	Merged revisions 208388 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208388 \| mmichelson \| 2009-07-23 14:34:49 -0500 (Thu, 23 Jul 2009) \| 24 lines Merged revisions 208386 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208386 \| mmichelson \| 2009-07-23 14:24:21 -0500 (Thu, 23 Jul 2009) \| 17 lines Fix a problem where a 491 response could be sent out of dialog. This generalizes the fix for issue 13849. The initial fix corrected the problem that Asterisk would reply with a 491 if a reinvite were received from an endpoint and we had not yet received an ACK from that endpoint for the initial INVITE it had sent us. This expansion also allows Asterisk to appropriately handle an INVITE with authorization credentials if Asterisk had not received an ACK from the previous transaction in which Asterisk had responded to an unauthorized INVITE with a 407. (closes issue #14239) Reported by: klaus3000 Patches: 14239.patch uploaded by mmichelson (license 60) Tested by: klaus3000 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@208391 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	92bec24919	Merged revisions 208383 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208383 \| jpeeler \| 2009-07-23 14:21:50 -0500 (Thu, 23 Jul 2009) \| 12 lines Merged revisions 208380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208380 \| jpeeler \| 2009-07-23 14:19:53 -0500 (Thu, 23 Jul 2009) \| 6 lines Only set the priindication setting when not performing a reload (closes issue #14696) Reported by: fdecher ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@208387 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	da4229cf04	Merged revisions 208314 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208314 \| mmichelson \| 2009-07-23 11:29:37 -0500 (Thu, 23 Jul 2009) \| 9 lines Merged revisions 208312 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208312 \| mmichelson \| 2009-07-23 11:29:18 -0500 (Thu, 23 Jul 2009) \| 3 lines Remove inaccurate XXX comment. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@208320 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	3e40075a0a	Merged revisions 208263 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r208263 \| mmichelson \| 2009-07-23 10:46:34 -0500 (Thu, 23 Jul 2009) \| 15 lines Merged revisions 208262 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208262 \| mmichelson \| 2009-07-23 10:43:07 -0500 (Thu, 23 Jul 2009) \| 8 lines Properly handle 183 responses which do not contain an SDP. (closes issue #15442) Reported by: ffloimair Patches: 15442.patch uploaded by mmichelson (license 60) Tested by: tkarl, ffloimair ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@208266 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 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	17 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	17 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	17 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	17 years ago
Mark Michelson	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	17 years ago
Richard Mudgett	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	17 years ago
Jeff Peeler	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	17 years ago
David Vossel	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	17 years ago
David Vossel	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	17 years ago
David Vossel	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	17 years ago
Richard Mudgett	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	17 years ago
David Vossel	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	17 years ago
Richard Mudgett	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	17 years ago
Russell Bryant	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	17 years ago
Richard Mudgett	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	17 years ago
David Vossel	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	17 years ago
David Vossel	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	17 years ago
Mark Michelson	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	17 years ago
David Vossel	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	17 years ago
Mark Michelson	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	17 years ago
Richard Mudgett	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	17 years ago
Kevin P. Fleming	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	17 years ago
David Vossel	4088c54d14	Merged revisions 205479 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r205479 \| dvossel \| 2009-07-08 18:19:09 -0500 (Wed, 08 Jul 2009) \| 16 lines Merged revisions 205471 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205471 \| dvossel \| 2009-07-08 18:15:54 -0500 (Wed, 08 Jul 2009) \| 10 lines Fixes 8khz assumptions Many calculations assume 8khz is the codec rate. This is not always the case. This patch only addresses chan_iax.c and res_rtp_asterisk.c, but I am sure there are other areas that make this assumption as well. Review: https://reviewboard.asterisk.org/r/306/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@205595 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	49f8d109bf	Merged revisions 204835 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r204835 \| rmudgett \| 2009-07-02 17:01:28 -0500 (Thu, 02 Jul 2009) \| 17 lines Merged revisions 204834 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204834 \| rmudgett \| 2009-07-02 16:59:43 -0500 (Thu, 02 Jul 2009) \| 10 lines Removed confusing warning message "Got Busy in Connected State" If an incoming mISDN call is answered with the Answer application and a subsequent Dial gets a busy endpoint then it is valid for that already connected channel to get the busy indication. Asterisk will play the busy tones until the dialplan plays something else or hangs up the call. (closes issue #11974) Reported by: fvdb ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@204838 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	88b7cce6cb	Merged revisions 204301 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r204301 \| mmichelson \| 2009-06-29 17:50:35 -0500 (Mon, 29 Jun 2009) \| 15 lines Merged revisions 204300 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204300 \| mmichelson \| 2009-06-29 17:45:34 -0500 (Mon, 29 Jun 2009) \| 9 lines Add error message so that it is clear why a SIP peer was not processed when a DNS lookup fails on a host or outboundproxy. (closes issue #13432) Reported by: p_lindheimer Patches: outboundproxy.patch uploaded by p (license 558) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@204304 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	0c3fa0cb94	Merged revisions 204247 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r204247 \| mmichelson \| 2009-06-29 16:48:54 -0500 (Mon, 29 Jun 2009) \| 32 lines Merged revisions 204243,204246 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204243 \| mmichelson \| 2009-06-29 16:23:43 -0500 (Mon, 29 Jun 2009) \| 22 lines Fix a problem where chan_sip would ignore "old" but valid responses. chan_sip has had a problem for quite a long time that would manifest when Asterisk would send multiple SIP responses on the same dialog before receiving a response. The problem occurred because chan_sip only kept track of the highest outgoing sequence number used on the dialog. If Asterisk sent two requests out, and a response arrived for the first request sent, then Asterisk would ignore the response. The result was that Asterisk would continue retransmitting the requests and ignoring the responses until the maximum number of retransmissions had been reached. The fix here is to rearrange the code a bit so that instead of simply comparing the sequence number of the response to our latest outgoing sequence number, we walk our list of outstanding packets and determine if there is a match. If there is, we continue. If not, then we ignore the response. In doing this, I found a few completely useless variables that I have now removed. (closes issue #11231) Reported by: flefoll Review: https://reviewboard.asterisk.org/r/298 ........ r204246 \| mmichelson \| 2009-06-29 16:37:05 -0500 (Mon, 29 Jun 2009) \| 3 lines Fix build oops. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@204250 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	fb7229c56f	Merged revisions 203909 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r203909 \| rmudgett \| 2009-06-26 20:07:52 -0500 (Fri, 26 Jun 2009) \| 23 lines Merged revisions 203908 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203908 \| rmudgett \| 2009-06-26 19:55:12 -0500 (Fri, 26 Jun 2009) \| 16 lines The ISDN CPE side should not exclusively pick B channels normally. Before this patch, Asterisk unconditionally picked B channels exclusively on the CPE side and normally allowed alternative B channels on the network side. Now Asterisk does the opposite. Reasons for the CPE side to normally not pick B channels exclusively: * For CPE point-to-multipoint mode (i.e. phone side), the CPE side does not have enough information to exclusively pick B channels. (There may be other devices on the line.) * Q.931 gives preference to the network side picking B channels. * Some telcos require the CPE side to not pick B channels exclusively. (closes issue #14383) Reported by: mbrancaleoni ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@203941 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	02b001519b	Merged revisions 203853 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r203853 \| jpeeler \| 2009-06-26 17:11:31 -0500 (Fri, 26 Jun 2009) \| 12 lines Merged revisions 203848 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203848 \| jpeeler \| 2009-06-26 17:09:19 -0500 (Fri, 26 Jun 2009) \| 5 lines Make sure to recreate the dahdi pseudo channel after dahdi restart (closes issue #14477) Reported by: timking ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@203857 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	679544f8dd	Merged revisions 203779 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203779 \| russell \| 2009-06-26 15:45:00 -0500 (Fri, 26 Jun 2009) \| 5 lines Ensure the TCP read buffer is fully initialized before handling each packet. (closes issue #14452) Reported by: umberto71 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@203782 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	576540abec	Merged revisions 203710 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203710 \| dvossel \| 2009-06-26 14:47:11 -0500 (Fri, 26 Jun 2009) \| 7 lines moving debug message from level 0 to 1. (closes issue #15404) Reported by: leobrown Patches: iax_codec_debug.patch uploaded by leobrown (license 541) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@203712 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	c33b1aa707	Merged revisions 203672 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203672 \| jpeeler \| 2009-06-26 14:03:25 -0500 (Fri, 26 Jun 2009) \| 16 lines Check if polarityonanswerdelay has elapsed before setting a channel as answered after a polarity reversal. Previously on a polarity switch event chan_dahdi would set the channel immediately as answered. This would cause problems if a polarity reversal occurred when the line was picked up as the dial would not have yet occurred. Now if the polarity reversal occurs before delay has elapsed after coming off hook or an answer, it is ignored. Also, some refactoring was done in _handle_event. (closes issue #13917) Reported by: alecdavis Patches: chan_dahdi.bug13917.feb09.diff2.txt uploaded by alecdavis (license 585) Tested by: alecdavis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@203709 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	45d9ad47ca	Merged revisions 203699 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203699 \| file \| 2009-06-26 16:27:24 -0300 (Fri, 26 Jun 2009) \| 2 lines Improve T.38 negotiation by exchanging session parameters between application and channel. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@203705 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jason Parker	a87390de51	Merged revisions 203258 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r203258 \| qwell \| 2009-06-25 14:22:46 -0500 (Thu, 25 Jun 2009) \| 10 lines Unmute when we get a dtmfup (we muted on dtmfdown) event. This would occasionally cause one-way audio when using hardware DTMF detection. (closes issue #14761) Reported by: tzafrir Patches: v1-14761.patch uploaded by dimas (license 88) Tested by: tzafrir, dimas ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@203276 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	b3e3e5add6	Merged revisions 203116 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r203116 \| russell \| 2009-06-25 11:04:10 -0500 (Thu, 25 Jun 2009) \| 18 lines Merged revisions 203115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203115 \| russell \| 2009-06-25 11:02:16 -0500 (Thu, 25 Jun 2009) \| 11 lines Resolve a crash related to a T.38 reinvite race condition. This change resolves a crash observed locally during some T.38 testing. A call was set up using a call file, and when the T.38 reinvite came in, the channel state was still AST_STATE_DOWN. The reason is explained by a comment in the code that previously lived in the handling of AST_STATE_RINGING. This change modifies the logic to handle the same race condition for any channel state that is not UP. (closes ABE-1895) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@203119 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	70d80ebe62	Merged revisions 203037 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r203037 \| rmudgett \| 2009-06-24 16:08:55 -0500 (Wed, 24 Jun 2009) \| 15 lines Merged revisions 203036 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203036 \| rmudgett \| 2009-06-24 16:01:43 -0500 (Wed, 24 Jun 2009) \| 8 lines Improved chan_dahdi.conf pritimer error checking. Valid format is: pritimer=timer_name,timer_value * Fixed segfault if the ',' is missing. * Completely check the range returned by pri_timer2idx() to prevent possible access outside array bounds. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@203077 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	f3ecab7e23	Merged revisions 202967 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r202967 \| mmichelson \| 2009-06-24 13:29:10 -0500 (Wed, 24 Jun 2009) \| 9 lines Merged revisions 202966 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202966 \| mmichelson \| 2009-06-24 13:28:47 -0500 (Wed, 24 Jun 2009) \| 3 lines Use the handy UNLINK macro instead of hand-coding the same thing in-line. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@202970 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago

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