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2011-02-15Merged revisions 308010 via svnmerge from qwell1-1/+0
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: http://svn.digium.com/svn/asterisk/trunk@308013 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-15include tcp keepalive socket calls only on linux, freebsd and othersmay1-0/+2
don't have these options on sockets. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@307969 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-15Add CLI "pri show channels" command.rmudgett3-6/+108
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: http://svn.digium.com/svn/asterisk/trunk@307964 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-15Merged revisions 307962 via svnmerge from rmudgett1-2/+2
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: http://svn.digium.com/svn/asterisk/trunk@307963 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-15Fixes compile error in chan_phone for big endian dvossel1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@307927 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-15Merged revisions 307879 via svnmerge from rmudgett6-32/+117
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: http://svn.digium.com/svn/asterisk/trunk@307883 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-15Merged revisions 307837 via svnmerge from tilghman1-1/+4
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: http://svn.digium.com/svn/asterisk/trunk@307838 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-14Merged revisions 307793 via svnmerge from tilghman1-3/+8
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: http://svn.digium.com/svn/asterisk/trunk@307795 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-14Making trunk compile again.tilghman2-12/+13
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@307752 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-14Merged revisions 307750 via svnmerge from tilghman2-4/+24
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: http://svn.digium.com/svn/asterisk/trunk@307751 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-13lc not found - it's warning, not error,may2-2/+2
change malloc to ast_calloc again git-svn-id: http://svn.digium.com/svn/asterisk/trunk@307713 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-12change malloc to ast_calloc calls to prevent crash of asteriskmay2-12/+13
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@307677 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-10Merged revisions 307536 via svnmerge from qwell2-2/+2
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: http://svn.digium.com/svn/asterisk/trunk@307537 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-10Merged revisions 307467 via svnmerge from mmichelson1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@307468 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-10Fixes bug in chan_sip where nativeformats are not set correctly.dvossel3-5/+24
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: http://svn.digium.com/svn/asterisk/trunk@307433 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-10Corrections for properly work with H.323v2 (older) endpoints and othermay8-54/+104
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: http://svn.digium.com/svn/asterisk/trunk@307396 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-09Add new manager action MeetmeListRooms.jpeeler2-0/+84
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: http://svn.digium.com/svn/asterisk/trunk@307359 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-09Disable color during running testlathama1-1/+1
(closes issue #18776) Reported by: alecdavis Patches: ast_deb_init.diff uploaded by lathama (license 1028) Tested by: andrel, lathama git-svn-id: http://svn.digium.com/svn/asterisk/trunk@307315 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-09Merged revisions 307273 via svnmerge from jpeeler1-1/+5
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: http://svn.digium.com/svn/asterisk/trunk@307274 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-09Allow parkedmusicclass to be settable for non-default parking lots.jpeeler2-0/+6
(closes issue #17946) Reported by: bluecrow76 Patches: asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff git-svn-id: http://svn.digium.com/svn/asterisk/trunk@307231 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-09Merged revisions 307228 via svnmerge from jpeeler1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@307229 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-09clarify warning when no loadable module supporttzafrir1-1/+1
Clarify warning message when LOADABLE_MODULES is disabled but we still try to load a module. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@307192 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-09Merged revisions 307142 via svnmerge from tilghman1-1/+3
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: http://svn.digium.com/svn/asterisk/trunk@307143 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-08Merged revisions 307092 via svnmerge from qwell1-10/+8
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: http://svn.digium.com/svn/asterisk/trunk@307097 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-08Merged revisions 307065 via svnmerge from mmichelson1-0/+3
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: http://svn.digium.com/svn/asterisk/trunk@307071 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-08Merged revisions 306979 via svnmerge from twilson1-2/+2
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: http://svn.digium.com/svn/asterisk/trunk@307061 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-08Documentation Updateslathama4-235/+963
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: http://svn.digium.com/svn/asterisk/trunk@307041 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-08Merged revisions 306967 via svnmerge from jpeeler1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@306968 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-08Merged revisions 306962 via svnmerge from jpeeler1-1/+3
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: http://svn.digium.com/svn/asterisk/trunk@306963 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-08Merged revisions 306866 via svnmerge from jpeeler1-2/+2
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: http://svn.digium.com/svn/asterisk/trunk@306867 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-08Documentation Updates.lathama1-4/+4
Start updates to the man pages. (issue #16505) Reported by: tzafrir Tested by: lathama git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306827 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-08Define the MCID acronym in chan_dahdi.conf.sample.rmudgett1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306793 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-08Use correct conditional for MCID send.rmudgett1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306791 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-07Pass a MCID request to the bridged channel.rmudgett9-1/+40
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: http://svn.digium.com/svn/asterisk/trunk@306755 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-07Merged revisions 306674 via svnmerge from twilson1-1/+2
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: http://svn.digium.com/svn/asterisk/trunk@306675 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-07Merged revisions 306619 via svnmerge from twilson1-0/+8
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: http://svn.digium.com/svn/asterisk/trunk@306670 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-07Merged revisions 306575 via svnmerge from mmichelson1-10/+11
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: http://svn.digium.com/svn/asterisk/trunk@306576 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-07Fixes use of ast_format_cap_append where ast_format_cap_copy is necessary.dvossel1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306541 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-05fix trivial issue after dvossel patch, initial zero fill user and peermay1-2/+2
structure before cap structure allocated. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306499 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-05Ignore voice frames in chan_dahdi native bridging. Hardware is handling them.rmudgett1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306464 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-04Send manager event for blackfilter only if it DOES NOT match.jpeeler1-2/+3
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: http://svn.digium.com/svn/asterisk/trunk@306432 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-04Add ISDN display ie text handling options to chan_dahdi.conf.rmudgett9-4999/+31672
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: http://svn.digium.com/svn/asterisk/trunk@306396 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-04Merged revisions 306356 via svnmerge from qwell1-0/+1
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: http://svn.digium.com/svn/asterisk/trunk@306359 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-04Fix compiler warning.rmudgett1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306326 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-04Merged revisions 306324 via svnmerge from rmudgett2-16/+18
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: http://svn.digium.com/svn/asterisk/trunk@306325 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-04Revert changes to extconf.cpabelanger1-24/+38
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: http://svn.digium.com/svn/asterisk/trunk@306292 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-04Replace ast_log(LOG_DEBUG, ...) with ast_debug()pabelanger39-777/+739
(closes issue #18556) Reported by: kkm Review: https://reviewboard.asterisk.org/r/1071/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306258 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-04Fix compile error in codec ilbc translator.dvossel2-3/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306257 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-03Merged revisions 306215 via svnmerge from jpeeler1-0/+4
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: http://svn.digium.com/svn/asterisk/trunk@306216 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-03Merged revisions 306127 via svnmerge from twilson1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306127 | twilson | 2011-02-03 13:03:26 -0800 (Thu, 03 Feb 2011) | 23 lines Merged revisions 306126 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r306126 | twilson | 2011-02-03 12:56:00 -0800 (Thu, 03 Feb 2011) | 16 lines Merged revisions 306119 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306119 | twilson | 2011-02-03 12:36:34 -0800 (Thu, 03 Feb 2011) | 9 lines Set hangup cause in local_hangup When a call involves a local channel (like SIP -> Local -> SIP), the hangup cause was not being set. This resulted in SIP channels sometimes getting a 503 error instead of a 486 when the far side sent a busy. In Asterisk 1.8+ this also can cause issues with CCSS that involve a local channel. This patch sets the hangupcause for one side of the local channel to the other in local_hangup for outbound calls. ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306128 f38db490-d61c-443f-a65b-d21fe96a405b