5642 Commits (d0bee2f7ba543604dd0a6a73e96d9f3c6f63177f)

Author	SHA1	Message	Date
Richard Mudgett	0de3f2833b	Merged revisions 210640 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r210640 | rmudgett | 2009-08-05 14:40:03 -0500 (Wed, 05 Aug 2009) | 21 lines Merged revisions 210575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r210575 | rmudgett | 2009-08-05 14:18:56 -0500 (Wed, 05 Aug 2009) | 14 lines Dialplan starts execution before the channel setup is complete. * Issue 15655: For the case where dialing is complete for an incoming call, dahdi_new() was asked to start the PBX and then the code set more channel variables. If the dialplan hungup before these channel variables got set, asterisk would likely crash. * Fixed potential for overlap incoming call to erroneously set channel variables as global dialplan variables if the ast_channel structure failed to get allocated. * Added missing set of CALLINGSUBADDR in the dialing is complete case. (closes issue #15655) Reported by: alecdavis ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@210647 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	db0581c7f0	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/1.6.0@209762 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Brooks	40c0cfaff8	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/1.6.0@209587 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Brooks	fef52dce32	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/1.6.0@209555 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Brooks	6d177693e0	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/1.6.0@209221 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	ab117ba86d	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/1.6.0@208925 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	bf0a2c9fa5	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/1.6.0@208752 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	6aa63436ab	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/1.6.0@208589 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	f43a65fd21	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/1.6.0@208549 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	791d4f0478	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/1.6.0@208468 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	db6c757a3d	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/1.6.0@208389 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	8d49fb4502	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/1.6.0@208384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	a9ad08042d	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/1.6.0@208316 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	0a6ccac217	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/1.6.0@208264 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	4ab9bff204	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/1.6.0@207860 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	ab6510ebf1	Revert r207636, this approach could potentially block for an unacceptable ... amount of time. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207783 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	69255bd210	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/1.6.0@207683 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	55a51b9194	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/1.6.0@207636 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	935f33e481	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/1.6.0@207425 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	c305a6d0a3	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/1.6.0@207286 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	08fb833859	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/1.6.0@207157 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	5f6fa4990f	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/1.6.0@207032 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	263df0044d	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/1.6.0@206947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	0faed3d459	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/1.6.0@206775 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	57f664c8f4	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/1.6.0@206762 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	f84624e23d	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/1.6.0@206705 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	587c202b8c	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/1.6.0@206555 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	f54c70ea66	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/1.6.0@206387 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	3d3e165752	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/1.6.0@206369 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	23705acc5e	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/1.6.0@206017 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	d2c214e042	Fix build. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205880 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	b3c7b4fa2d	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/1.6.0@205879 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	6e6557cb04	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/1.6.0@205843 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	966a316fac	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/1.6.0@205777 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	35dbf93676	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/1.6.0@205729 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	b2e3c3e436	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/1.6.0@205697 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	f22cf5c484	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/1.6.0@205597 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	6547190dd4	SIP Dialog ref counting ... This patch adds reference counting for sip dialogs into 1.6.0. When proc_session_timer() is called from the scheduler thread it has no guarantee the session timer's dialog won't be freed from underneath it. Now the session timer holds a reference to the dialog, preventing it from being destroyed during the middle of proc_session_timer(). (closes issue #13623) Reported by: Nik Soggia Review: https://reviewboard.asterisk.org/r/302/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205117 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	0a0144c4a0	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/1.6.0@204836 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	8d1643655b	removes fake dialog_unref and dialog_ref function calls. ... dialog_unref() and dialog_ref() in 1.6.0 where only place holders for reference counting once it was implemented. The functions did nothing but return the pointer on ref and NULL on unref. These calls have been removed to make way for a patch that actually does dialog ref counting. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@204652 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	ae065d0125	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/1.6.0@204302 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	0889af49c6	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/1.6.0@204248 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	4730b56a50	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/1.6.0@203910 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	21af4c4acb	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/1.6.0@203855 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	3be09ad7e9	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/1.6.0@203780 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	cb69985027	reverse whitespace change 203711 that was based on looking at sig_analog (which has about a 1000 line indentation change that is not worth doing here) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@203717 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	9fa99fc5b7	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/1.6.0@203716 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	a184348797	whitespace fix ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@203711 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	fc33f7b57e	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/1.6.0@203701 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	7f0af9cf29	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/1.6.0@203698 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	8ee2d538bd	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/1.6.0@203117 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	db7b94fafb	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/1.6.0@203044 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	a228e74faa	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/1.6.0@202968 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	03914b9a4a	Merged revisions 202925 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r202925 | file | 2009-06-24 15:08:17 -0300 (Wed, 24 Jun 2009) | 2 lines Ensure the default settings are applied for T.38 when we set it up for a peer. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@202926 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Fredrickson	d9ab6e17c1	Merged revisions 202761 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r202761 | mattf | 2009-06-23 17:08:43 -0500 (Tue, 23 Jun 2009) | 1 line I could have sworn I committed this patch ages ago, but... bug fix with setting NAI properly on linksets in certain situations. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@202763 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	cbdbf23bdc	Merged revisions 202672 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r202672 | dvossel | 2009-06-23 11:31:30 -0500 (Tue, 23 Jun 2009) | 18 lines Merged revisions 202671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202671 | dvossel | 2009-06-23 11:28:46 -0500 (Tue, 23 Jun 2009) | 12 lines MWI NOTIFY contains a wrong URI if Asterisk listens to non-standard port and transport (closes issue #14659) Reported by: klaus3000 Patches: patch_chan_sip_fixMWIuri_1.4.txt uploaded by klaus3000 (license 65) mwi_port-transport_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel, klaus3000 Review: https://reviewboard.asterisk.org/r/288/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@202675 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	b535dda70c	Merged revisions 202603 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r202603 | mmichelson | 2009-06-23 10:23:00 -0500 (Tue, 23 Jun 2009) | 8 lines Blocked revisions 202601 via svnmerge ........ r202601 | mmichelson | 2009-06-23 10:22:35 -0500 (Tue, 23 Jun 2009) | 3 lines Fix more memory leaks that may result if rtp is not successfully allocated. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@202612 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	436dce6109	Recorded merge of revisions 202574 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r202574 | mmichelson | 2009-06-23 10:11:47 -0500 (Tue, 23 Jun 2009) | 8 lines Blocked revisions 202572 via svnmerge ........ r202572 | mmichelson | 2009-06-23 10:08:27 -0500 (Tue, 23 Jun 2009) | 3 lines Fix potential memory leak in chan_sip when video rtp is not allocated properly. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@202577 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	eca12f6e13	Merged revisions 202415 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r202415 | russell | 2009-06-22 11:05:08 -0500 (Mon, 22 Jun 2009) | 9 lines Merged revisions 202414 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202414 | russell | 2009-06-22 11:00:00 -0500 (Mon, 22 Jun 2009) | 2 lines Make Polycom subscription type override check more explicit. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@202416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	25b0edc60a	Merged revisions 202343 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r202343 | mmichelson | 2009-06-22 09:58:24 -0500 (Mon, 22 Jun 2009) | 36 lines Merged revisions 202341-202342 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202341 | mmichelson | 2009-06-22 09:42:55 -0500 (Mon, 22 Jun 2009) | 26 lines Fix a situation in which Asterisk would not stop retransmitting 487s. If a CANCEL were received by Asterisk, we would send a 487 in response to the original INVITE and a 200 OK for the CANCEL. If there were a network hiccup which caused the 200 OK and the 487 to be lost, then the UA communicating with Asterisk may try to retransmit its CANCEL. Asterisk's response to this used to be to try sending another 487 to the canceled INVITE and another 200 OK to the CANCEL. The problem here is that the originally-sent 487 was sent "reliably" meaning that it will be retransmitted until it is received properly. So when we receive the second CANCEL it is likely that the first batch of 487s we sent is still going strong and reaches the UA. The result was that the second set of 487s would be retransmitted constantly until the maximum number of retries had been reached. The fix for this is that if we receive a second CANCEL for an INVITE, then we cancel the retransmission of the first set of 487s and start a second set. This causes the dialog to be terminated reasonably. (closes issue #14584) Reported by: klaus3000 Patches: 14584_v2.patch uploaded by mmichelson (license 60) Tested by: klaus3000 ........ r202342 | mmichelson | 2009-06-22 09:44:58 -0500 (Mon, 22 Jun 2009) | 3 lines Remove an extra debug line left from previous commit. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@202344 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	03f46e7a81	Merged revisions 202337 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r202337 | mmichelson | 2009-06-22 09:35:09 -0500 (Mon, 22 Jun 2009) | 31 lines Merged revisions 202336 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202336 | mmichelson | 2009-06-22 09:34:05 -0500 (Mon, 22 Jun 2009) | 25 lines Fix a possible infinite loop in SDP parsing during glare situation. There was a while loop in get_ip_and_port_from_sdp which was controlled by a call to get_sdp_iterate. The loop would exit either if what we were searching for was found or if the return was NULL. The problem is that get_sdp_iterate never returns NULL. This means that if what we were searching for was not present, the loop would run infinitely. This modification of the loop fixes the problem. (closes issue #15213) Reported by: schmidts (closes issue #15349) Reported by: samy (closes issue #14464) Reported by: pj (closes issue #15345) Reported by: aragon Patches: sip_inf_loop.patch uploaded by mmichelson (license 60) Tested by: aragon ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@202338 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	93017afcc8	Added deadlock protection to try_suggested_sip_codec in chan_sip.c. ... Review: https://reviewboard.asterisk.org/r/287/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@202006 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	ac6ab2899d	Merged revisions 201994 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r201994 | dvossel | 2009-06-19 15:24:37 -0500 (Fri, 19 Jun 2009) | 14 lines Merged revisions 201993 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201993 | dvossel | 2009-06-19 15:22:02 -0500 (Fri, 19 Jun 2009) | 8 lines timestamp was being converted to host order as a short rather than a long (closes issue #15361) Reported by: ffloimair Patches: ts_issue.diff uploaded by dvossel (license 671) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@201997 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	986be7d2a2	Merged revisions 201678 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r201678 | dvossel | 2009-06-18 11:37:42 -0500 (Thu, 18 Jun 2009) | 11 lines fixes some memory leaks and redundant conditions (closes issue #15269) Reported by: contactmayankjain Patches: patch.txt uploaded by contactmayankjain (license 740) memory_leak_stuff.trunk.diff uploaded by dvossel (license 671) Tested by: contactmayankjain, dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@201682 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	82f2aa293d	Merged revisions 201462 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r201462 | mmichelson | 2009-06-17 15:10:01 -0500 (Wed, 17 Jun 2009) | 12 lines Fix problem with no audio due to ignoring the SDP. A recent change to our SDP version comparison made audio not function on some calls. This was because of a test wherein we were trying to see if an unsigned value was less than 0. This is a dumb comparison and arguably the compiler should have warned about it. Alas, though, it slipped past. Now it's fixed by changing the variable to be a signed type. Found by several developers. Tested by mnicholson and dbrooks. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@201463 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Brooks	ca7b9b9fe4	Merged revisions 201381 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r201381 | dbrooks | 2009-06-17 14:15:07 -0500 (Wed, 17 Jun 2009) | 16 lines Merged revisions 201380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201380 | dbrooks | 2009-06-17 13:45:50 -0500 (Wed, 17 Jun 2009) | 9 lines Checks for NULL sip_pvt pointer in chan_sip.c->acf_channel_read() Zombie channels could be passed, and chan_sip.c wasn't checking for it. Could crash Asterisk. Now checking for NULL pointer. (closes issue #15330) Reported by: okrief Tested by: dbrooks ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@201443 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	6cbe57b730	Merged revisions 201223 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r201223 | dvossel | 2009-06-16 17:29:30 -0500 (Tue, 16 Jun 2009) | 2 lines fix issue with build_contact introduced by the "SIP trasnport type issues" commit ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@201226 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	c2d79c89bb	Merged revisions 200946 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r200946 | dvossel | 2009-06-16 11:03:30 -0500 (Tue, 16 Jun 2009) | 32 lines SIP transport type issues What this patch addresses: 1. ast_sip_ouraddrfor() by default binds to the UDP address/port reguardless if the sip->pvt is of type UDP or not. Now when no remapping is required, ast_sip_ouraddrfor() checks the sip_pvt's transport type, attempting to set the address and port to the correct TCP/TLS bindings if necessary. 2. It is not necessary to send the port number in the Contact header unless the port is non-standard for the transport type. This patch fixes this and removes the todo note. 3. In sip_alloc(), the default dialog built always uses transport type UDP. Now sip_alloc() looks at the sip_request (if present) and determines what transport type to use by default. 4. When changing the transport type of a sip_socket, the file descriptor must be set to -1 and in some cases the tcptls_session's ref count must be decremented and set to NULL. I've encountered several issues associated with this process and have created a function, set_socket_transport(), to handle the setting of the socket type. (closes issue #13865) Reported by: st Patches: dont_add_port_if_tls.patch uploaded by Kristijan (license 753) 13865.patch uploaded by mmichelson (license 60) tls_port_v5.patch uploaded by vrban (license 756) transport_issues.diff uploaded by dvossel (license 671) Tested by: mmichelson, Kristijan, vrban, jmacz, dvossel Review: https://reviewboard.asterisk.org/r/278/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@200992 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	40757d599e	Merged revisions 165180,200689 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r165180 | mnicholson | 2008-12-17 12:49:12 -0600 (Wed, 17 Dec 2008) | 14 lines This patch adds a new 'ignoresdpversion' option to sip.conf. When this is enabled (either globally or for a specific peer), chan_sip will treat any SDP data it receives as new data and update the media stream accordingly. By default, Asterisk will only modify the media stream if the SDP session version received is different from the current SDP session version. This option is required to interoperate with devices that have non-standard SDP session version implementations (observed by toc on the bug tracker with Microsoft OCS which always uses 0 as the session version). http://reviewboard.digium.com/r/94/ (closes issue #13958) Reported by: toc Tested by: toc ........ r200689 | kpfleming | 2009-06-15 15:42:38 -0500 (Mon, 15 Jun 2009) | 12 lines Accept T.38 re-INVITE responses with invalid SDP versions. This commit changes the 'incoming SDP version' check logic a bit more; when 'ignoresdpversion' is *not* set for a peer, if we initiate a re-INVITE to switch to T.38, we'll always accept the peer's SDP response, even if they don't properly increment the SDP version number as they should. If this situation occurs, a warning message will be generated suggesting that the peer's configuration be changed to include the 'ignoresdpversion' configuration option (although ideally they'd fix their SIP implementation to be RFC compliant). AST-221 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@200724 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	5f0b3e489f	Merged revisions 200514 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r200514 | mmichelson | 2009-06-15 10:22:11 -0500 (Mon, 15 Jun 2009) | 11 lines Merged revisions 200513 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200513 | mmichelson | 2009-06-15 10:21:46 -0500 (Mon, 15 Jun 2009) | 5 lines Add INFO to our allowed methods so that endpoints know they may send it to us. AST-223 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@200515 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	bd9f6cf82d	Merged revisions 200146 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r200146 | mmichelson | 2009-06-11 16:17:14 -0500 (Thu, 11 Jun 2009) | 5 lines Fix a crash due to a potentially NULL p->options. Thanks to mnicholson for pointing it out. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@200149 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	7a3b46c789	The 1.6.0 branch was missing all invite_branch logic. It has now been added. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@199975 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	6cab2f47e6	Merged revisions 199818 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r199818 | dvossel | 2009-06-09 15:47:57 -0500 (Tue, 09 Jun 2009) | 11 lines CLI NOTIFY sending wrong transport type. SIP's cli NOTIFY command only used UDP rather than copying the transport type from the peer. (closes issue #15283) Reported by: jthurman Patches: sip-notify-tcp-svn199728.patch uploaded by jthurman (license 614) Tested by: jthurman, dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@199821 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	29113e4ad5	Merged revisions 199227 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r199227 | mmichelson | 2009-06-05 08:51:08 -0500 (Fri, 05 Jun 2009) | 14 lines Correct "dahdi show channels" output when specifying a group. Since a DAHDI channel may belong to multiple groups, we need to use a bitwise and instead of equivalence to determine whether to display the channel information. (closes issue #15248) Reported by: gentian Patches: 15248.patch uploaded by mmichelson (license 60) Tested by: gentian ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@199228 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	ea62e16ffe	Merged revisions 199139 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r199139 | dvossel | 2009-06-04 14:10:16 -0500 (Thu, 04 Jun 2009) | 9 lines Merged revisions 199138 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199138 | dvossel | 2009-06-04 14:00:15 -0500 (Thu, 04 Jun 2009) | 3 lines Additional updates to AST-2009-001 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@199142 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	d1b1506bc5	Merged revisions 198824 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r198824 | dvossel | 2009-06-02 12:55:35 -0500 (Tue, 02 Jun 2009) | 8 lines fixes issue with channels not going down after transfer Iax2 currently does not support native bridging if the timeoutms value is set. We check for that in iax2_bridge, but then set timeoutms to 0 by default. If the timeoutms is not provided it is set to -1. By setting timeoutms to 0 it is processed causing a bridging retry loop. (closes issue #15216) Reported by: oxymoron Tested by: dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@198825 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	bc1b330dec	Merged revisions 198791 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r198791 | file | 2009-06-02 10:48:06 -0300 (Tue, 02 Jun 2009) | 5 lines Correct documentation for the register line, specifically where the domain should be specified. (closes issue #14367) Reported by: Nick_Lewis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@198792 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Eliel C. Sardanons	93e30e3e23	Merged revisions 197621 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r197621 | eliel | 2009-05-28 12:01:48 -0400 (Thu, 28 May 2009) | 19 lines Merged revisions 197562 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197562 | eliel | 2009-05-28 11:21:32 -0400 (Thu, 28 May 2009) | 13 lines Use the address we already know when reloading a peer with nat=yes. If we already have an address for a peer, and we are reloading the sip configuration, try to use that address to contact the peer, instead of getting it from the Contact. (closes issue #15194) Reported by: ibc Patches: sip.patch uploaded by eliel (license 64) Tested by: manwe ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@197704 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	b422316b51	Merged revisions 197697 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r197697 | file | 2009-05-28 15:45:11 -0300 (Thu, 28 May 2009) | 2 lines Fix a bug where the trunkmtu setting was not set to the default value of 1240 on load but was on reload. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@197698 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	6c05ae7cc1	'iax show peer blah' now outputs whether or not peer 'blah' is in trunk mode or not. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@197622 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	a66b938920	Merged revisions 197606 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r197606 | mmichelson | 2009-05-28 10:32:19 -0500 (Thu, 28 May 2009) | 22 lines Recorded merge of revisions 197588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197588 | mmichelson | 2009-05-28 10:27:49 -0500 (Thu, 28 May 2009) | 16 lines Allow for media to arrive from an alternate source when responding to a reinvite with 491. When we receive a SIP reinvite, it is possible that we may not be able to process the reinvite immediately since we have also sent a reinvite out ourselves. The problem is that whoever sent us the reinvite may have also sent a reinvite out to another party, and that reinvite may have succeeded. As a result, even though we are not going to accept the reinvite we just received, it is important for us to not have problems if we suddenly start receiving RTP from a new source. The fix for this is to grab the media source information from the SDP of the reinvite that we receive. This information is passed to the RTP layer so that it will know about the alternate source for media. Review: https://reviewboard.asterisk.org/r/252 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@197615 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	cd950bcbf8	Merged revisions 197467 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r197467 | file | 2009-05-28 10:47:45 -0300 (Thu, 28 May 2009) | 15 lines Merged revisions 197466 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197466 | file | 2009-05-28 10:44:58 -0300 (Thu, 28 May 2009) | 8 lines Fix a bug where the flag indicating the presence of rport would get overwritten by the nat setting. The presence of rport is now stored as a separate flag. Once the dialog is setup and authenticated (or it passes through unauthenticated) the proper nat flag is set. (closes issue #13823) Reported by: dimas ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@197469 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	2b5d7b9ce7	Fixes merge issue during r196454. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@197086 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	6aa494b8f2	Merged revisions 196988 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r196988 | seanbright | 2009-05-27 09:02:54 -0400 (Wed, 27 May 2009) | 9 lines Display an error message when chan_alsa fails to load due to a missing or inaccessible configuration file. Before this change, when chan_alsa failed to load due to a missing or inaccessible configuration file, no message would be displayed. With this change, when chan_alsa fails to load due to a missing or inaccessible configuration file, a message will be displayed. (closes issue #14760) Reported by: Nick_Lewis Patches: chan_alsa.c-confload.patch uploaded by Nick (license 657) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@196989 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	6c1703877b	Merged revisions 196721 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r196721 | file | 2009-05-26 10:43:13 -0300 (Tue, 26 May 2009) | 7 lines Fix a bug where the sip unregister CLI command did not completely unregister the peer. (closes issue #15118) Reported by: alecdavis Patches: chan_sip_unregister.diff2.txt uploaded by alecdavis (license 585) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@196722 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	5249104890	Merged revisions 196416 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r196416 | dvossel | 2009-05-22 16:09:45 -0500 (Fri, 22 May 2009) | 19 lines SIP set outbound transport type from Registration In sip.conf the transport option allows for the configuration of what transport types (udp, tcp, and tls) a peer will accept, but only the first type listed was used for outbound connections. This patch changes this. Now the default transport type is only used until the peer registers. When registration takes place the transport type is parsed out of the Contact header. If the Contact header's transport type is equal to one that the peer supports, the peer's default transport type for outbound connections is set to match the Contact header's type. If the Contact header's transport type is not present, then the peer's default transport type is set to match the one the peer registered with. When a peer unregisters or the registration expires, the default transport type for that peer is reset. (closes issue #12282) Reported by: rjain Patches: reg_patch_1.diff uploaded by dvossel (license 671) Tested by: dvossel (closes issue #14727) Reported by: pj Patches: reg_patch_3.diff uploaded by dvossel (license 671) Tested by: pj, dvossel Review: https://reviewboard.asterisk.org/r/249/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@196454 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	9495819c2e	Merged revisions 196117 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r196117 | file | 2009-05-22 10:56:47 -0300 (Fri, 22 May 2009) | 12 lines Merged revisions 196116 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r196116 | file | 2009-05-22 10:54:17 -0300 (Fri, 22 May 2009) | 5 lines Fix a bug where using immediate with mISDN caused a cause code of 16 to get sent back instead of 1 if the 's' extension did not exist. (closes issue #12286) Reported by: lmamane ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@196118 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	be7700852b	Merged revisions 195995 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r195995 | dvossel | 2009-05-21 14:11:49 -0500 (Thu, 21 May 2009) | 20 lines Merged revisions 195991 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195991 | dvossel | 2009-05-21 14:04:56 -0500 (Thu, 21 May 2009) | 14 lines Sign problem calculating timestamp for iax frame leads to no audio on the receiving peer. There are rare cases in which a frame's delivery timestamp is slightly less than the iax2_pvt's offset. This causes the pvt's timestamp to be a small negative number, but since the timestamp value is unsigned it looks like a huge positive number. This patch checks for this negative case and sets the ms to zero. A similar check is already done right below this one in the 'else' statement. (closes issue #15032) Reported by: guillecabeza Patches: chan_iax2.c.patch_timestamp uploaded by guillecabeza (license 380) Tested by: guillecabeza (closes issue #14216) Reported by: Andrey Sofronov ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@195997 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	0b6e79502e	Merged revisions 195449 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r195449 | file | 2009-05-19 11:43:54 -0300 (Tue, 19 May 2009) | 14 lines Merged revisions 195448 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195448 | file | 2009-05-19 11:41:45 -0300 (Tue, 19 May 2009) | 7 lines Fix a bug where direct RTP setup would partially occur even when disabled if the calling channel was answered. (issue #13545) Reported by: davidw (issue #14244) Reported by: mbnwa ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@195450 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	bbcf0052fd	Merged revisions 195089 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r195089 | file | 2009-05-18 10:36:17 -0300 (Mon, 18 May 2009) | 5 lines Fix a bug where specifying an empty outboundproxy would cause packets to get sent to ourself. (closes issue #15106) Reported by: timeshell ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@195090 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	70e4d673f1	Merged revisions 194874 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r194874 | dvossel | 2009-05-15 17:44:44 -0500 (Fri, 15 May 2009) | 23 lines Merged revisions 194873 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194873 | dvossel | 2009-05-15 17:43:13 -0500 (Fri, 15 May 2009) | 17 lines IAX2 REGAUTH loop IAX was not sending REGREJ to terminate invalid registrations. Instead it sent another REGAUTH if the authentication challenge failed. This caused a loop of REGREQ and REGAUTH frames. (Related to Security fix AST-2009-001) (closes issue #14867) Reported by: aragon Tested by: dvossel (closes issue #14717) Reported by: mobeck Patches: regauth_loop_update_patch.diff uploaded by dvossel (license 671) Tested by: dvossel ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@194875 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	134d5899c8	Merged revisions 194833 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r194833 | dvossel | 2009-05-15 15:52:12 -0500 (Fri, 15 May 2009) | 24 lines Merged revisions 194557,194685 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194557 | dvossel | 2009-05-14 17:59:43 -0500 (Thu, 14 May 2009) | 10 lines IAX2 "Ghost" Channels There is a bug tracker issue where people are reporting "Ghost" channels in their 'iax2 show channels' output. The confusion is caused by channels being listed as "(NONE)" with format "unknown". These are not channels of coarse. They are usually just pending registration or poke requests, but it is confusing output. To help make sense of this I have added two columns to 'iax2 show channels'. One shows the first message which started the transaction, and the second shows the last message sent by either side of the call. This helps diagnose why the entry exists and why it may not go away. (closes issue #14207) Reported by: clive18 Review: https://reviewboard.asterisk.org/r/246/ ........ r194685 | dvossel | 2009-05-15 10:40:37 -0500 (Fri, 15 May 2009) | 6 lines Update to previous IAX2 "Ghost" Channels patch. Fixed some comments made on reviewboard for the previous patch. (issue #14207) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@194834 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	bd0383c3be	Merged revisions 194496 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r194496 | mmichelson | 2009-05-14 17:20:51 -0500 (Thu, 14 May 2009) | 30 lines Merged revisions 194484 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194484 | mmichelson | 2009-05-14 17:17:55 -0500 (Thu, 14 May 2009) | 24 lines Fix a race condition where a reinvite could trigger a 482 response. The loop detection/spiral detection code in chan_sip used the owner channel's state as a criterion for determining if the incoming INVITE is a looped request. The problem with this is that the INVITE-handling code happens in a different thread than the thread that marks the owner channel as being up. As a result, if a reinvite were to come in very quickly, say from another Asterisk on the same LAN, it was possible for the reinvite to arrive before the owner channel had been set to the up state. This patch corrects the problem by using the invitestate of the sip_pvt instead, since that can be guaranteed to be set correctly by the time the reinvite arrives. Since there is a switch statement further in the INVITE-handling code, the AST_STATE_RINGING state also checks the invitestate of the sip_pvt in case we should actually be treating the channel as if it were up already. (closes issue #12215) Reported by: jpyle Patches: 12215_confirmed.patch uploaded by mmichelson (license 60) Tested by: lmadsen ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@194504 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	4a9497891c	Merged revisions 193954 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r193954 | mmichelson | 2009-05-12 15:28:13 -0500 (Tue, 12 May 2009) | 18 lines Update spiral support in trunk and 1.6.X to match what is in 1.4. In 1.4, a SIP spiral is treated the same way as a call forward. This works much better than what is currently in trunk and 1.6.X. The code in trunk and 1.6.X did not create a new call to the recipient of the spiral, instead trying to continue the same call. In addition to just being plain wrong, this also had the side effect of only being able to spiral calls to other SIP channels. With this in place, as long as call forwards are honored, SIP spirals will work properly. This means that it will work for outbound calls made by the Queue, Dial, and Page applications. For originated calls and spool calls, however, the spiral will not work properly until a generic call forward mechanism is introduced into Asterisk. (relates to issue #13630) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@193960 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	d111483ab8	Merged revisions 193614 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r193614 | rmudgett | 2009-05-11 14:11:29 -0500 (Mon, 11 May 2009) | 19 lines Merged revisions 193613 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193613 | rmudgett | 2009-05-11 14:09:00 -0500 (Mon, 11 May 2009) | 12 lines Sent wrong message to clear a call we started if the other end has not responed yet. In the state MISDN_CALLING (i.e. SETUP was sent but no answer has arrived yet), it is not allowed to clear the call with RELEASE_COMPLETE. It must be cleared with DISCONNECT. A RELEASE_COMPLETE is only allowed as an answer to a SETUP. (See Q.931 ch. 5.3.2, 5.3.2.a, 5.3.2.b) Patches: chan-misdn-ccstate7.patch uploaded by customer. JIRA ABE-1862 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@193615 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	80a37a02a6	Merged revisions 193387 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r193387 | dvossel | 2009-05-08 15:32:51 -0500 (Fri, 08 May 2009) | 7 lines TCP not matching valid peer. find_peer() does not find a valid peer when using pvt->recv as the sockaddr_in argument. Because of the way TCP works, the port number in pvt->recv is not what we're looking for at all. There is currently only one place that find_peer searches for a peer using the sockaddr_in argument. If the peer is not found after using pvt->recv (works for UDP since the port number will be correct), a temp sockaddr_in struct is made using the Contact header in the sip_request. This has the correct port number in it. Review: http://reviewboard.digium.com/r/236/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@193449 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	3c092fb35b	Merged revisions 193263 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r193263 | dvossel | 2009-05-08 09:52:19 -0500 (Fri, 08 May 2009) | 15 lines Merged revisions 193262 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193262 | dvossel | 2009-05-08 09:51:09 -0500 (Fri, 08 May 2009) | 9 lines "misdn show config" segfaults asterisk, if no MSN lists (closes issue #14976) Reported by: alecdavis Patches: misdn_config.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis, FabienToune ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@193264 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	e9cf423d60	Merged revisions 193077 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r193077 | rmudgett | 2009-05-07 17:24:04 -0500 (Thu, 07 May 2009) | 12 lines Merged revisions 193050 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193050 | rmudgett | 2009-05-07 17:17:06 -0500 (Thu, 07 May 2009) | 5 lines Give a more helpful message when an incoming call's dialed extension does not match. Added the dialed extension and context to the chan_misdn messages warning that the dialed number cannot be matched in the dialplan. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@193078 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	6315384b63	Merged revisions 192938 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r192938 | tilghman | 2009-05-07 12:13:36 -0500 (Thu, 07 May 2009) | 6 lines Send DTMF frame before playing back audio. (closes issue #14858) Reported by: barryf Patches: 20090507__bug14858.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@192939 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	2a0f0439e5	Merged revisions 192933 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r192933 | tilghman | 2009-05-07 11:43:56 -0500 (Thu, 07 May 2009) | 17 lines Merged revisions 192932 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192932 | tilghman | 2009-05-07 11:29:08 -0500 (Thu, 07 May 2009) | 10 lines Eliminate repetition of fullcontact during reconstruction. If the fullcontact field appears in both the sippeers and the sipregs table, then during reconstruction of the field, it will otherwise be doubled. (closes issue #14754) Reported by: Alexei Gradinari Patches: 20090506__bug14754.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@192934 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago