asterisk

Commit Graph

Author	SHA1	Message	Date
Alexandr Anikin	dd5df75bef	Merged revisions 308098 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r308098 \| may \| 2011-02-16 10:57:22 +0300 (Wed, 16 Feb 2011) \| 2 lines ifdef __linux__ keepalive variables also ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308099 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	551dac2eda	Merged revisions 308010 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r308010 \| qwell \| 2011-02-15 17:34:03 -0600 (Tue, 15 Feb 2011) \| 24 lines Merged revisions 308007 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r308007 \| qwell \| 2011-02-15 17:33:24 -0600 (Tue, 15 Feb 2011) \| 17 lines Merged revisions 308002 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r308002 \| qwell \| 2011-02-15 17:32:20 -0600 (Tue, 15 Feb 2011) \| 10 lines Fix regression that changed behavior of queues when ringing a queue member. This reverts r298596, which was to fix a highly bizarre and contrived issue with a queue member that called into his own queue being transferred back into his own queue. I couldn't reproduce that issue in any way. I think one of the other recent transfer fixes actually fixed this. (closes issue #18747) Reported by: vrban ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308013 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alexandr Anikin	7b44cd8ec7	include tcp keepalive socket calls only on linux, freebsd and others don't have these options on sockets. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307969 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	4a48600231	Add CLI "pri show channels" command. List the current mapping of DAHDI B channels to Asterisk channel names and which calls are on hold or call-waiting. Calls on hold or call-waiting are not associated with any B channel. JIRA LIBPRI-27 JIRA SWP-2547 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307964 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	b1db966684	Merged revisions 307962 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307962 \| rmudgett \| 2011-02-15 13:52:45 -0600 (Tue, 15 Feb 2011) \| 1 line Don't crash when forcing caller id. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307963 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	64ed1ba3e9	Fixes compile error in chan_phone for big endian git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307927 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	b2ef13cb60	Merged revisions 307879 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307879 \| rmudgett \| 2011-02-15 10:13:55 -0600 (Tue, 15 Feb 2011) \| 37 lines No response sent for SIP CC subscribe/resubscribe request. Asterisk does not send a response if we try to subscribe for call completion after we have received a 180 Ringing. You can only subscribe for call completion when the call has been cleared. When we receive the 180 Ringing, for this call, its call-completion state is 'CC_AVAILABLE'. If we then send a subscribe message to Asterisk, it trys to change the call-completion state to 'CC_CALLER_REQUESTED'. Because this is an invalid state change, it just ignores the message. The only state Asterisk will accept our subscribe message is in the 'CC_CALLER_OFFERED' state. Asterisk will go into the 'CC_CALLER_OFFERED' when the SIP client clears the call by sending a CANCEL. Asterisk should always send a response. Even if its a negative one. The fix is to allow for the CCSS core to notify a CC agent that a failure has occurred when CC is requested. The "ack" callback is replaced with a "respond" callback. The "respond" callback has a parameter indicating either a successful response or a specific type of failure that may need to be communicated to the requester. (closes issue #18336) Reported by: GeorgeKonopacki Tested by: mmichelson, rmudgett JIRA SWP-2633 (closes issue #18337) Reported by: GeorgeKonopacki Tested by: mmichelson JIRA SWP-2634 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307883 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	e38fa2d3cd	Merged revisions 307837 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r307837 \| tilghman \| 2011-02-15 01:02:45 -0600 (Tue, 15 Feb 2011) \| 15 lines Merged revisions 307836 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r307836 \| tilghman \| 2011-02-15 01:01:37 -0600 (Tue, 15 Feb 2011) \| 8 lines Need to retrieve the rows affected before using the associated variable. (closes issue #18795) Reported by: irroot Patches: 20110211__issue18795.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307838 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	5c49abfcf3	Merged revisions 307793 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r307793 \| tilghman \| 2011-02-14 14:16:55 -0600 (Mon, 14 Feb 2011) \| 15 lines Merged revisions 307792 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r307792 \| tilghman \| 2011-02-14 14:10:28 -0600 (Mon, 14 Feb 2011) \| 8 lines Increment usage count at first reference, to avoid a race condition with many threads creating connections all at once. (issue #18156) Reported by: asgaroth Patches: 20110214__issue18156.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307795 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	c0e33b03c3	Making trunk compile again. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307752 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	7800a1c330	Merged revisions 307750 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307750 \| tilghman \| 2011-02-14 00:50:23 -0600 (Mon, 14 Feb 2011) \| 23 lines Calling a gosub routine defined in AEL from Dial/Queue ceased to work. A bug in AEL did not distinguish between the "s" extension generated by AEL and an "s" extension that was required to exist by the chan_dahdi (or another channel) that was not supplied with a starting extension. Therefore, AEL made incorrect assumptions about what commands were permissable in the context. This was fixed by making AEL generate a different extension name. However, Dial and Queue make additional assumptions about the name of the default gosub extension. Therefore, they needed to be brought into line with a "macro" rendered by AEL (as a gosub), without breaking traditional dialplans written without the aid of AEL. Related to (issue #18480) Reported by: nivek (closes issue #18729) Reported by: kkm Patches: 20110209__issue18729.diff.txt uploaded by tilghman (license 14) 018729-dial-queue-gosub-try3.patch uploaded by kkm (license 888) Tested by: kkm ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307751 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alexandr Anikin	bd32cb4ba5	lc not found - it's warning, not error, change malloc to ast_calloc again git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307713 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alexandr Anikin	eaf73d6588	change malloc to ast_calloc calls to prevent crash of asterisk git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307677 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	96cbd4ffcd	Merged revisions 307536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r307536 \| qwell \| 2011-02-10 16:39:30 -0600 (Thu, 10 Feb 2011) \| 22 lines Merged revisions 307535 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r307535 \| qwell \| 2011-02-10 16:35:49 -0600 (Thu, 10 Feb 2011) \| 15 lines Merged revisions 307534 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r307534 \| qwell \| 2011-02-10 16:33:09 -0600 (Thu, 10 Feb 2011) \| 8 lines Remove color when executing commands via a remote console. Essentially this makes '-x' imply '-n' on rasterisk. This was done in a different and incomplete way previously, which I'm reverting here. (issue #18776) Reported by: alecdavis ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307537 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	4cba13eb60	Merged revisions 307467 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307467 \| mmichelson \| 2011-02-10 11:44:42 -0600 (Thu, 10 Feb 2011) \| 5 lines Fix a gaffe in the CCSS sample configuration. Discovered by Philippe Lindheimer and pointed out on #asterisk-dev ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307468 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	08460fc094	Fixes bug in chan_sip where nativeformats are not set correctly. The nativeformats field was being overwritten when it should have been appended too. This caused some format capabilities to be lost briefly and some log warnings to be output. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307433 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alexandr Anikin	707cf78c5a	Corrections for properly work with H.323v2 (older) endpoints and other small fixes. Interpret remote side H.225 version. Corrections for H.323v2 endpoints: don't start TCS and MSD before connect, don't start TCS and MSD by accepting H.245 connection, start TCS and MSD by StartH245 facility message. Other fixes: fix non zeroended remoteDisplayName issue, small fixes in call clearing by closing H.245 connection, tcp keepalive introduced on TCP connections (now is hardcoded, will be configurable in the future), don't force H.245tunneling if FastStart is active, don't send Alerting singal more than once per call. (closes issue #18542) Reported by: vmikhelson Patches: issue18542-final-3.patch uploaded by may213 (license 454) Tested by: vmikhelson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307396 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	8f7982f280	Add new manager action MeetmeListRooms. From the submitter: I've added a new manager action to list only the active conferences on an Asterisk system. It shows the same data displayed when you run a 'meetme list' on the Asterisk CLI. (closes issue #17905) Reported by: rcasas Patches: app_meetme.c.patch uploaded by rcasas (license 641) Review: https://reviewboard.asterisk.org/r/874/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307359 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Andrew Latham	0703a9a321	Disable color during running test (closes issue #18776) Reported by: alecdavis Patches: ast_deb_init.diff uploaded by lathama (license 1028) Tested by: andrel, lathama git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307315 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	10362292ef	Merged revisions 307273 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307273 \| jpeeler \| 2011-02-09 15:06:33 -0600 (Wed, 09 Feb 2011) \| 8 lines Add missing debug info for ao2_link for use with REF_DEBUG in ao2 callback. (closes issue #18758) Reported by: rgagnon Patches: branch-1.8-r306540-astobj-fix.diff uploaded by rgagnon (license 1202) trunk-r306540-astobj-fix.diff uploaded by rgagnon (license 1202) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307274 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	e2df246636	Allow parkedmusicclass to be settable for non-default parking lots. (closes issue #17946) Reported by: bluecrow76 Patches: asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307231 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	6b0fa46103	Merged revisions 307228 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r307228 \| jpeeler \| 2011-02-09 13:52:51 -0600 (Wed, 09 Feb 2011) \| 17 lines Merged revisions 307227 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r307227 \| jpeeler \| 2011-02-09 13:52:12 -0600 (Wed, 09 Feb 2011) \| 11 lines Make sure to set parking dial context for non-default parking lots. Since parking_con_dial isn't settable, set all parking lots to "park-dial". (closes issue #17946) Reported by: bluecrow76 Patches: asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff uploaded by bluecrow76 (license 270) modified by me ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307229 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tzafrir Cohen	1540401a4a	clarify warning when no loadable module support Clarify warning message when LOADABLE_MODULES is disabled but we still try to load a module. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307192 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	fc4df44bd8	Merged revisions 307142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307142 \| tilghman \| 2011-02-08 23:39:39 -0600 (Tue, 08 Feb 2011) \| 3 lines Initialize tracking variable in structure properly. Fixes a memory leak. (Reported by The_Boy_Wonder on IRC, fixed by me.) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307143 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	f01e9568d2	Merged revisions 307092 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307092 \| qwell \| 2011-02-08 15:24:01 -0600 (Tue, 08 Feb 2011) \| 9 lines Fix issue with verbose messages not showing on remote console. This code was reworked recently, and since the logchannel list hadn't been created yet at this point, and it was a verbose message, it was being dropped on the floor. Now it'll continue on to where it should be handled. (closes issue #18580) Reported by: pabelanger ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307097 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	0074165356	Merged revisions 307065 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307065 \| mmichelson \| 2011-02-08 15:13:08 -0600 (Tue, 08 Feb 2011) \| 6 lines Add a couple of useful channel variables for the CC recall macro. CC_EXTEN and CC_CONTEXT will allow you to determine the channel and context that will be called when the recall occurs. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307071 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	4f57a3bb7c	Merged revisions 306979 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306979 \| twilson \| 2011-02-08 12:18:08 -0800 (Tue, 08 Feb 2011) \| 16 lines Merged revisions 306973 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306973 \| twilson \| 2011-02-08 12:14:09 -0800 (Tue, 08 Feb 2011) \| 9 lines Merged revisions 306972 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306972 \| twilson \| 2011-02-08 12:05:13 -0800 (Tue, 08 Feb 2011) \| 2 lines Fix comparison for REFER Replaces tags with pedantic=yes ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307061 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Andrew Latham	a350924700	Documentation Updates Note default polling setting in voicemail.conf Add missing config to asterisk.conf Update manpage (issue #16505) Reported by: tzafrir Patches: asterisk_sgml_fixes_demo.diff uploaded by tzafrir (license 46) Tested by: lathama, tzafrir git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307041 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	a46bfe67bd	Merged revisions 306967 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306967 \| jpeeler \| 2011-02-08 13:41:42 -0600 (Tue, 08 Feb 2011) \| 16 lines Merged revisions 306966 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306966 \| jpeeler \| 2011-02-08 13:41:21 -0600 (Tue, 08 Feb 2011) \| 9 lines Merged revisions 306965 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306965 \| jpeeler \| 2011-02-08 13:40:58 -0600 (Tue, 08 Feb 2011) \| 1 line fix this line again ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306968 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	e2cdaf47bb	Merged revisions 306962 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306962 \| jpeeler \| 2011-02-08 13:25:38 -0600 (Tue, 08 Feb 2011) \| 22 lines Merged revisions 306961 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306961 \| jpeeler \| 2011-02-08 13:25:10 -0600 (Tue, 08 Feb 2011) \| 15 lines Merged revisions 306960 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306960 \| jpeeler \| 2011-02-08 13:18:50 -0600 (Tue, 08 Feb 2011) \| 9 lines Backup file storing message duration is not used with IMAP_STORAGE, remove code. The message duration is stored in the body of the email when using IMAP_STORAGE, so nothing needs to happen with the backup file. (closes issue #18718) Reported by: kerframil ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306963 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	9264ab00f5	Merged revisions 306866 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306866 \| jpeeler \| 2011-02-08 10:21:45 -0600 (Tue, 08 Feb 2011) \| 16 lines Merged revisions 306865 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306865 \| jpeeler \| 2011-02-08 10:21:25 -0600 (Tue, 08 Feb 2011) \| 9 lines Merged revisions 306864 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306864 \| jpeeler \| 2011-02-08 10:19:17 -0600 (Tue, 08 Feb 2011) \| 1 line make this safer and fully correct, pointed out by Steve Davis ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306867 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Andrew Latham	c4271440ab	Documentation Updates. Start updates to the man pages. (issue #16505) Reported by: tzafrir Tested by: lathama git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306827 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	8b584000a9	Define the MCID acronym in chan_dahdi.conf.sample. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306793 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	209a39f4b0	Use correct conditional for MCID send. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306791 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	49feb747ba	Pass a MCID request to the bridged channel. Pass a MCID request to the bridged channel so the bridged channel can send it to the network. The ability to send the MCID request on an ISDN span is enabled with the new chan_dahdi.conf mcid_send option. JIRA SWP-2845 JIRA ABE-2736 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306755 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	1277a80a5b	Merged revisions 306674 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306674 \| twilson \| 2011-02-07 14:43:22 -0800 (Mon, 07 Feb 2011) \| 24 lines Merged revisions 306673 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306673 \| twilson \| 2011-02-07 14:40:20 -0800 (Mon, 07 Feb 2011) \| 17 lines Merged revisions 306672 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306672 \| twilson \| 2011-02-07 14:35:20 -0800 (Mon, 07 Feb 2011) \| 10 lines Don't try to pickup a call in the middle of a masquerade If A calls B which doesn't answer and C & D both try to do a call pickup, it is possible for ast_pickup_call to answer the call, then fail to masquerade one of the calls because the other one is already in the process of masquerading. This patch checks to see if the channel is in the process of masquerading before call before selecting it for a pickup. Review: https://reviewboard.asterisk.org/r/1094/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306675 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	a974d1a4ce	Merged revisions 306619 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306619 \| twilson \| 2011-02-07 14:15:27 -0800 (Mon, 07 Feb 2011) \| 24 lines Merged revisions 306618 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306618 \| twilson \| 2011-02-07 13:59:54 -0800 (Mon, 07 Feb 2011) \| 17 lines Merged revisions 306617 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306617 \| twilson \| 2011-02-07 13:51:43 -0800 (Mon, 07 Feb 2011) \| 10 lines Don't allow a REFER w/replaces to replace its own dialog Asterisk currently accepts a REFER with a Refer-To with an embedded Replaces header that matches the dialog of the REFER. This would be a situation like A calls B, A calls C, A transfers B to A, which is just silly. This patch makes the transfer fail instead of making Asterisk freak out and forget to hang other channels up. Review: https://reviewboard.asterisk.org/r/1093/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306670 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	f4ea670a6a	Merged revisions 306575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r306575 \| mmichelson \| 2011-02-07 11:36:56 -0600 (Mon, 07 Feb 2011) \| 9 lines Rearrange a bit of code in the generic CC recall operation. By waiting to call the callback macro after the CC_INTERFACES, extension, priority, and context have been set, this information can be accessed more easily within the callback macro. Reported by Philippe Lindheimer. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306576 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	2db3c9e058	Fixes use of ast_format_cap_append where ast_format_cap_copy is necessary. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306541 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Alexandr Anikin	7f86bd2f16	fix trivial issue after dvossel patch, initial zero fill user and peer structure before cap structure allocated. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306499 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	484f9bec0a	Ignore voice frames in chan_dahdi native bridging. Hardware is handling them. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306464 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	3d667d7c0f	Send manager event for blackfilter only if it DOES NOT match. The logic got reversed, oops. Works properly now when multiple blackfilters are present. (closes issue #18283) Reported by: telecos82 Patches: ast_managereventfilter.patch uploaded by telecos82 (license 687) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306432 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	a8aeb04a9f	Add ISDN display ie text handling options to chan_dahdi.conf. The display ie handling can be controlled independently in the send and receive directions with the following options: * Block display text data. * Use display text in SETUP/CONNECT messages for name. * Use display text for COLP name updates (FACILITY/NOTIFY as appropriate). * Pass arbitrary display text during a call. Sent in INFORMATION messages. Received from any message that the display text was not used as a name. If the display options are not set then the options default to legacy behavior. The arbitrary display text is exchanged between bridged channels using the AST_FRAME_TEXT frame type. To send display text from the dialplan use the SendText() application when the arbitrary display text option is enabled. JIRA SWP-2688 JIRA ABE-2693 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306396 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	0beeb00ef3	Merged revisions 306356 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306356 \| qwell \| 2011-02-04 13:24:29 -0600 (Fri, 04 Feb 2011) \| 16 lines Merged revisions 306346 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r306346 \| qwell \| 2011-02-04 13:21:43 -0600 (Fri, 04 Feb 2011) \| 9 lines Don't fallthrough to 'unknown' in the 'ringing' case. This could cause improper exits from the queue. (closes issue #18499) Reported by: zaltar Patches: app_queue.patch uploaded by zaltar (license 1148) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306359 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	9fc0d9be66	Fix compiler warning. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306326 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	4d8feab7fa	Merged revisions 306324 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r306324 \| rmudgett \| 2011-02-04 12:53:06 -0600 (Fri, 04 Feb 2011) \| 9 lines Don't send redirecting updates to the caller if the dialplan forked the call. Each fork in the dial could be redirected and confuse the caller. For ISDN the DivLeg1 and DivLeg3 messages would get confused because ISDN redirects calls in sequence not in parallel. * Also fixed a formatting inconsistency in app_dial.c and make a warning message more useful about what frame type could not be written. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306325 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	ed3a4856aa	Revert changes to extconf.c It seems extconf.c already defines some local ast_debug() functions. Theses should be removed and replaced with logger.h. A patch will be added to reviewboard shortly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306292 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	3556e4c2d4	Replace ast_log(LOG_DEBUG, ...) with ast_debug() (closes issue #18556) Reported by: kkm Review: https://reviewboard.asterisk.org/r/1071/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306258 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	9f65acf33e	Fix compile error in codec ilbc translator. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306257 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	285d953fdf	Merged revisions 306215 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r306215 \| jpeeler \| 2011-02-03 17:49:28 -0600 (Thu, 03 Feb 2011) \| 20 lines Fix SIP deadlock involving state changes. Once again a call to pbx_builtin_getvar_helper (and pbx_builtin_setvar_helper) has caused locking problems. Both of these functions lock the channel when the channel argument is passed in! In this case, the suspected problem (the backtrace makes it impossible to tell) was the private being locked in sip_set_rtp_peer and then: transmit_reinvite_with_sdp try_suggested_sip_codec pbx_builtin_getvar_helper (Traced to verify that the fix was only required in 1.8 and later.) (closes issue #18491) Reported by: cmaj Patches: chan_sip_fix_deadlocks_bug_18491.txt uploaded by cmaj (license 830) Tested by: cmaj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306216 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago

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