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${ noResults }
20998 Commits (51ce432d0743ed839af28bc620fa2bc94eed8245)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Richard Mudgett
|
6fb282becb |
Invalid read in ast_channel_set_caller_event().
Valgrind reported that ast_channel_set_caller_event() was reading data from a freed buffer when using the pre_set structure. Rearange things to pre-calculate the name and number pointer before updating the caller party structure to see if the name or number was changed. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@308903 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Terry Wilson
|
463a39b5d1 |
Merged revisions 308814 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r308814 | twilson | 2011-02-24 11:54:49 -0600 (Thu, 24 Feb 2011) | 19 lines Merged revisions 308813 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r308813 | twilson | 2011-02-24 11:42:16 -0600 (Thu, 24 Feb 2011) | 12 lines Don't broadcast FullyBooted to every AMI connection The FullyBooted event should not be sent to every AMI connection every time someone connects via AMI. It should only be sent to the user who just connected. (closes issue #18168) Reported by: FeyFre Patches: bug0018168.patch uploaded by FeyFre (license 1142) Tested by: FeyFre, twilson ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@308815 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Matthew Nicholson
|
bb1f856e88 |
Merged revisions 308722 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r308722 | mnicholson | 2011-02-24 08:59:41 -0600 (Thu, 24 Feb 2011) | 9 lines Merged revisions 308721 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r308721 | mnicholson | 2011-02-24 08:54:56 -0600 (Thu, 24 Feb 2011) | 2 lines silence gcc 4.2 compiler warning ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@308723 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Terry Wilson
|
f0992a0b5e |
Merged revisions 308678 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r308678 | twilson | 2011-02-23 21:38:22 -0600 (Wed, 23 Feb 2011) | 8 lines Use remotesecret to authenticate with a remote party The remotesecret option was only being used for outbound registration and not for placing calls. This patch uses remotesecret on outbound calls if it is set, otherwise secret is still used. Review: https://reviewboard.asterisk.org/r/1107/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@308679 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Richard Mudgett
|
7b353a26ae |
sig_pri_new_ast_channel() should return NULL when new_ast_channel() fails.
(closes issue #18874) Reported by: cmaj Patches: patch-sig_pri-crash-possible-null-channel-pointer.diff.txt uploaded by cmaj (license 830) JIRA SWP-3172 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@308622 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Andrew Latham
|
e6dd56de09 |
Use ast_debug for console logging
Guessed the log levels based on info that level 3 is the soft roof. Can we create a page / document to define the levels? git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@308526 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Matthew Nicholson
|
f8db85c4b3 |
Merged revisions 308414 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r308414 | mnicholson | 2011-02-21 09:00:22 -0600 (Mon, 21 Feb 2011) | 12 lines Merged revisions 308413 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r308413 | mnicholson | 2011-02-21 08:57:15 -0600 (Mon, 21 Feb 2011) | 5 lines Properly check the bounds of arrays when decoding UDPTL packets. Also, remove broken support for receiving UDPTL packets larger than 16k. That shouldn't ever happen anyway. AST-2011-002 FAX-281 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@308416 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Andrew Latham
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679a7326d6 |
Add HTTP URI Debug logging and update notice
enable reporting of the request URI / URL in debugging change funny debug note to a serious note. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@308393 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Andrew Latham
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e682054a44 |
Add CSS MIME Type
Modern browsers are checking for the MIME Type of pages and in some cases will not load a file if the type is wrong. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@308330 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Tilghman Lesher
|
b1e0478a60 |
A few more (copies of) files to ignore in this directory.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@308288 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Alexandr Anikin
|
6d86c8f1e6 |
added g729onlyA option for announce only AnnexA g.729 codec in
h.323 capabilities. Option can be global or per user/peer. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@308242 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Paul Belanger
|
735eb73607 |
Fix FreeBSD builds.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@308150 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Alexandr Anikin
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3dcf79a893 |
ifdef __linux__ keepalive variables also
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@308098 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Jason Parker
|
c8ef3e081b |
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/branches/1.8@308010 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Alexandr Anikin
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d3a39d5bfa |
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/branches/1.8@307970 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Richard Mudgett
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227c620866 |
Don't crash when forcing caller id.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@307962 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Richard Mudgett
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a5f6367057 |
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/branches/1.8@307879 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Tilghman Lesher
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bff7dd69e0 |
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/branches/1.8@307837 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Tilghman Lesher
|
4a3cecd3ed |
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/branches/1.8@307793 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Tilghman Lesher
|
ff43beaa2d |
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/branches/1.8@307750 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
Jason Parker
|
0962153fea |
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/branches/1.8@307536 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Alexandr Anikin
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aa41d4f640 |
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.
(issue 0018542)
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/branches/1.8@307509 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
15 years ago |
Mark Michelson
|
e60115cc20 |
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/branches/1.8@307467 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Andrew Latham
|
187d17a281 |
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/branches/1.8@307314 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
Jeff Peeler
|
c28a605bbd |
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/branches/1.8@307273 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Jeff Peeler
|
e07ccee87f |
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/branches/1.8@307228 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Tilghman Lesher
|
6be4e4d83c |
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/branches/1.8@307142 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Jason Parker
|
5da9bab0bc |
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/branches/1.8@307092 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Mark Michelson
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c4b05dfaf0 |
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/branches/1.8@307065 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Andrew Latham
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d7783faf19 |
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/branches/1.8@306999 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Terry Wilson
|
cd3b672f45 |
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/branches/1.8@306979 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Jeff Peeler
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49c4800686 |
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/branches/1.8@306967 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Jeff Peeler
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dad67ad1a4 |
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/branches/1.8@306962 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Jeff Peeler
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59502582b3 |
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/branches/1.8@306866 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Andrew Latham
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5f9a130198 |
Documentation Updates.
More updates to the removed doc folder and start updates to the man page. (issue #16505) Reported by: tzafrir Tested by: lathama git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@306826 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Terry Wilson
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36f5dd0013 |
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/branches/1.8@306674 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Terry Wilson
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4b54ce5ce5 |
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/branches/1.8@306619 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Mark Michelson
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aa1ce8bbb2 |
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/branches/1.8@306575 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Jason Parker
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e8bd6696b5 |
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/branches/1.8@306356 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Richard Mudgett
|
cee1db213b |
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/branches/1.8@306324 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Jeff Peeler
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6bb8cc3a9b |
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/branches/1.8@306215 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Terry Wilson
|
5eca7e5bd5 |
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: https://origsvn.digium.com/svn/asterisk/branches/1.8@306127 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
Jeff Peeler
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3ff8273840 |
Merged revisions 306123 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r306123 | jpeeler | 2011-02-03 14:49:48 -0600 (Thu, 03 Feb 2011) | 10 lines Set exception on channel in parking thread when POLLPRI event detected. This is done just to make the code be equivalent to the old select code. As noted in 303106 the same issue was already fixed in this branch, but the exception was not set on the channel in the case of POLLPRI. The reason that this did not cause a problem here is because in 122923 the check in __ast_read to check the exception flag was removed. (related to #18637) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@306124 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Andrew Latham
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752864bc32 |
res_phoneprov add snom 300, 320, 360, 370, 820, 821, 870 support
(issue #18713) Reported by: lathama Patches: snom_dir.diff uploaded by lathama (license 1028) Tested by: lathama git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@305987 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Richard Mudgett
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a785544090 |
Merged revisions 305889 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r305889 | rmudgett | 2011-02-02 18:15:07 -0600 (Wed, 02 Feb 2011) | 17 lines Merged revisions 305888 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305888 | rmudgett | 2011-02-02 18:02:43 -0600 (Wed, 02 Feb 2011) | 8 lines Minor AST_FRAME_TEXT related issues. * Include the null terminator in the buffer length. When the frame is queued it is copied. If the null terminator is not part of the frame buffer length, the receiver could see garbage appended onto it. * Add channel lock protection with ast_sendtext(). * Fixed AMI SendText action ast_sendtext() return value check. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@305923 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Tilghman Lesher
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70edcdef43 |
Eliminate a file descriptor leak when using the FILE() dialplan function.
(closes issue #18731) Reported by: marioabajo git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@305844 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Andrew Latham
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69e83f1a72 |
Replacing doc/* and asterisk.pdf with wiki links
Adding links to http(s)://wiki.asterisk.org git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@305838 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Andrew Latham
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bb2cd0fc1a |
Replacing doc/* with wiki links
Adding links to http(s)://wiki.asterisk.org git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@305798 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Andrew Latham
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b7c58ed676 |
Replace link to old doc with new wiki page.
Link to https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@305753 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Jason Parker
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974aa9e2ad |
Reverse sense of an error test when reading from astdb.
(closes issue #18545) Reported by: jcovert Patches: chan_iax2.c.patch uploaded by jcovert (license 551) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@305692 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |