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${ noResults }
3157 Commits (283f63c594d0c4327c51fdae0587a6a9f1cb15d3)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Richard Mudgett
|
67749ae7bb |
Fix formatting of AMI header for SIP show peer.
ASTERISK-17486 exposed the problem for AMI parsers.
(closes issue ASTERISK-18649)
Reported by: Jacek Konieczny
Patches:
asterisk-sipshowpeer_response_end.patch (license #6298) patch uploaded by Jacek Konieczny
........
Merged revisions 338663 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338664 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
14 years ago |
Leif Madsen
|
dd73844f72 |
Merged revisions 338492 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338492 | lmadsen | 2011-09-29 13:31:33 -0500 (Thu, 29 Sep 2011) | 6 lines Update documentation for SIP_HEADER. The SIP_HEADER function only works on the the initial SIP INVITE. The documentation was updated in trunk, but not in 1.8 or 10, so I'm making them match. (Closes issue ASTERISK-18640) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338493 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Gregory Nietsky
|
540b7a38a9 |
Merged revisions 338416 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338416 | irroot | 2011-09-29 14:13:05 +0200 (Thu, 29 Sep 2011) | 12 lines The rtptimeout setting is ignored on a per peer basis. Not only is the rtptimeout ignored in some cases but rtpkeepalive and rtpholdtimeout is affected. this commit also removes rtptimeout/rtpholdtimeout on text rtp. (closes issue ASTERISK-18559) Review: https://reviewboard.asterisk.org/r/1452 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@338417 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Jonathan Rose
|
857e4fdb14 |
Generate Security events in chan_sip using new Security Events Framework
Security Events Framework was added in 1.8 and support was added for AMI to generate
events at that time. This patch adds support for chan_sip to generate security events.
(closes issue ASTERISK-18264)
Reported by: Michael L. Young
Patches:
security_events_chan_sip_v4.patch (license #5026) by Michael L. Young
Review: https://reviewboard.asterisk.org/r/1362/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337595 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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14 years ago |
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Gregory Nietsky
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a31b5ce87e |
Allow Setting Auth Tag Bit length Based on invite or config option
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14 years ago |
Terry Wilson
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17124a2510 |
Merged revisions 336791 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336791 | twilson | 2011-09-19 17:07:58 -0500 (Mon, 19 Sep 2011) | 2 lines Don't interfere with T.38 reinvites This is an update to the fix for ASTERISK-18340 and ASTERISK-17725 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336792 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Olle Johansson
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dd12759127 |
Merged revisions 336501 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336501 | oej | 2011-09-19 15:33:50 +0200 (Mån, 19 Sep 2011) | 5 lines Add diversion header to a 302 redirect response if we have diversion data (closes issue ASTERISK-18143) patch by oej ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336502 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Olle Johansson
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0cfc3c6e9e |
Merged revisions 336378 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336378 | oej | 2011-09-19 11:40:44 +0200 (Mån, 19 Sep 2011) | 9 lines Add missing unlock at MWI message sending time (closes issue ASTERISK-18573) Patches: sip_mwi_lock.patch (license #5041) by Gregory Hinton Nietsky Thanks to irrot for the reminder, to Gregory for the patch! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336381 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Jonathan Rose
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1701266596 |
Merged revisions 336294 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336294 | jrose | 2011-09-16 14:53:40 -0500 (Fri, 16 Sep 2011) | 13 lines Fix bad RTP media bridges in directmedia calls on peers separated by multiple Asterisk nodes. In a situation involving devices on separate Asterisk trunks, the remote RTP bridge would break when starting a call with directmedia. This patch queues a new type of control frame so that our RTP bridge loop can properly detect when these situations occur and check to see if peers need to be updated in order to send their media to the proper location. (Closes issue ASTERISK-18340) Reported by: Thomas Arimont (Closes issue ASTERISK-17725) Reported by: kwk Tested by: twilson, jrose ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336307 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Olle Johansson
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7c7e697c3a |
Merged revisions 335319 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335319 | oej | 2011-09-12 15:25:30 +0200 (Mån, 12 Sep 2011) | 12 lines Lock the peer->mvipvt to avoid crashes with SIP history enabled After the launch of 1.6 event-based MWI we have two threads handling the peer->mwipvt, which cause issues with SIP history additions in combination with the max limit for number of history entries. Review: https://reviewboard.asterisk.org/r/1373/ (closes issue ASTERISK-18288) Thanks to irrot for peer review. Work with this bug funded by IPvision AS ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@335323 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Stefan Schmidt
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0f50837b20 |
Merged revisions 335259 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335259 | schmidts | 2011-09-12 11:09:19 +0000 (Mon, 12 Sep 2011) | 6 lines build_peer doesnt unlink a peer object from peers_by_ip container which leads to a wrong refcounter value. adding an ao2_unlink from the peers_by_ip container fix it. Review: https://reviewboard.asterisk.org/r/1428/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@335260 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Matthew Jordan
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4e57652651 |
Merged revisions 335064 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) | 23 lines Updated SIP 484 handling; added Incomplete control frame When a SIP phone uses the dial application and receives a 484 Address Incomplete response, if overlapped dialing is enabled for SIP, then the 484 Address Incomplete is forwarded back to the SIP phone and the HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete application dialplan logic was automatically triggered; now, explicit dialplan usage of the application is required. Additionally, this patch adds a new AST_CONTOL_FRAME type called AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame, it is an indication that the dialplan expects more digits back from the device. If the device supports overlap dialing it should attempt to notify the device that the dialplan is waiting for more digits; otherwise, it can handle the frame in a manner appropriate to the channel driver. (closes issue ASTERISK-17288) Reported by: Mikael Carlsson Tested by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/1416/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@335078 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Matthew Nicholson
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8c5c537997 |
Merged revisions 334156 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334156 | mnicholson | 2011-08-31 13:50:33 -0500 (Wed, 31 Aug 2011) | 4 lines Disable T.38 when we get a invite with image media port set to 0 ASTERISK-17678 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334157 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Kinsey Moore
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1180e8d803 |
Merged revisions 334006 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334006 | kmoore | 2011-08-31 10:18:37 -0500 (Wed, 31 Aug 2011) | 7 lines Correct an AMI protocol violation with SIPshowpeer The response of SIPshowpeer ends with "\r\n\r\n". Since other commands are ended by using \r\n this confuses any interfacing script. (closes issue ASTERISK-17486) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@334007 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Terry Wilson
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02d2a80332 |
Merged revisions 333836 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333836 | twilson | 2011-08-29 16:38:31 -0500 (Mon, 29 Aug 2011) | 15 lines Refresh peer address if DNS unavailable at peer creation If Asterisk starts and no DNS is available, outbound registrations will fail indefinitely. This patch copies the address from the sip_registry struct, which will be updated, to the peer->addr when necessary. If dnsmgr is enabled, the registration fails without the patch because even though the address on the registry is updated via dnsmgr, the address is just copied on the first try. Since we use ast_sockaddr_copy, dnsmgr can't update the address that is copied to the sip_pvt or peers. Closes issue ASTERISK-18000 Review: https://reviewboard.asterisk.org/r/1335/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@333837 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Jonathan Rose
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6ae104a4b2 |
Merged revisions 332118 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332118 | jrose | 2011-08-16 12:38:19 -0500 (Tue, 16 Aug 2011) | 16 lines ASTERISK-18067 ASTERISK-15479 - White Space affects mailbox value, multiple MWI subs Before, having multiple subscriptions to mailboxes on a sip peer set via the mailbox setting in sip.conf would only result in updates being sent on whichever mailbox triggered the mwi event. Now all of them get counted regardless. Also fixes a bug involving parsing of the mailbox option in sip.conf so that trailing and leading spaces before/after commas are trimmed. (closes issue ASTERISK-18067) Reported by: aragon (closes issue ASTERISK-15479) Reported by: Ben Winslow Patches: chan_sip.c-mwi_multi_mailbox_fix-1.6.2.13.diff (License #5288) patch uploaded by Ben Winslow ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@332119 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Matthew Nicholson
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0059096c5e |
Merged revisions 332026 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332026 | mnicholson | 2011-08-16 10:06:31 -0500 (Tue, 16 Aug 2011) | 2 lines use DEFAULT_STORE_SIP_CAUSE to set the default value for the 'storesipcause' option AST-580 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@332027 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Matthew Nicholson
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b83a26eefa |
In 10 and trunk this option is disabled by default.
Merged revisions 332021 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332021 | mnicholson | 2011-08-16 09:20:43 -0500 (Tue, 16 Aug 2011) | 7 lines Added the 'storesipcause' option to sip.conf to allow the user to disable the setting of HASH(SIP_CAUSE,<chan name>) on the channel. Having chan_sip set HASH(SIP_CAUSE,<chan name>) on the channel carries a significant performance penalty because of the usage of the MASTER_CHANNEL() dialplan function. AST-580 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@332022 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
David Vossel
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132a7e75a0 |
Merged revisions 331867 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331867 | dvossel | 2011-08-15 10:12:16 -0500 (Mon, 15 Aug 2011) | 6 lines Fixes locking inversion issues present in the handling of the sip REFER method. (closes issue ASTERISK-18082) Reported by: James Van Vleet ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@331868 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Kinsey Moore
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678ece77d5 |
Merged revisions 331517 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331517 | kmoore | 2011-08-10 17:23:08 -0500 (Wed, 10 Aug 2011) | 10 lines SIP Notify via AMI or CLI leaks SIP PVTs Any SIP notify sent via AMI or CLI leaks a SIP PVT with ref count +2. Removing the additional ref just before the invite and adding an unref following it corrects the issue as seen via REF_DEBUG. The unref existed in a distant revision and it appears as though the wrong ref operation was removed. (closes issue ASTERISK-18091) Review: https://reviewboard.asterisk.org/r/1332/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@331518 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Richard Mudgett
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9bd1af5e42 |
Merged revisions 331248 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331248 | rmudgett | 2011-08-09 17:12:59 -0500 (Tue, 09 Aug 2011) | 15 lines Misc minor items found in code. * Add some reentrancy protection in pbx.c when creating the contexts_table hash table. * Fix inverted test in chan_sip.c conditional code. * Fix uninitialized variable and use of the wrong variable in chan_iax2.c. * Fix test of return value in app_parkandannounce.c. Explicitly testing for -1 is bad if the function does not actually return that value when it fails. * Fixup some comments and add some curly braces in features.c. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@331265 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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David Vossel
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252a251ac2 |
Merged revisions 330578 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330578 | dvossel | 2011-08-02 11:07:02 -0500 (Tue, 02 Aug 2011) | 2 lines Optimization to buffer initialization fix. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@330579 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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David Vossel
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03fd1aca68 |
Merged revisions 330575 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330575 | dvossel | 2011-08-02 10:53:21 -0500 (Tue, 02 Aug 2011) | 5 lines Fixes uninitialized string buffer in log message. (closes issue ASTERISK-17200) Reported by: lmadsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@330576 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Jason Parker
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8cf16c1c15 |
Merged revisions 329994 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329994 | qwell | 2011-07-28 10:45:24 -0500 (Thu, 28 Jul 2011) | 6 lines Fix a SIP transfer deadlock. The locking in this function is very scary. There are like 6 structs involved. (closes issue AST-470) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@329995 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Sean Bright
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2fb05fb506 |
Merged revisions 329895 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329895 | seanbright | 2011-07-28 07:34:33 -0400 (Thu, 28 Jul 2011) | 2 lines Make the output of Externhost in 'sip show settings' more consistent. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@329896 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Gregory Nietsky
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aacc128812 |
Remove lastmsgssent from sip it has not been working since 1.6
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14 years ago |
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Kinsey Moore
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af58e75ce1 |
Merged revisions 328935 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328935 | kmoore | 2011-07-20 14:00:23 -0500 (Wed, 20 Jul 2011) | 8 lines Inband DTMF regression The functionality of inband DTMF in chan_sip relied upon ast_rtp_instance_dtmf_mode_get/set not working properly to avoid calling ast_rtp_instance_dtmf_begin/end on RTP streams with inband DTMF. According to documentation, ast_rtp_instance_dtmf_begin/end is meant only for RFC2833 DTMF, never inband. This fixes the regression introduced in revision 328823. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/2.0@328936 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Kinsey Moore
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98c4fee4cb |
Merged revisions 328823 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328823 | kmoore | 2011-07-19 12:57:18 -0500 (Tue, 19 Jul 2011) | 11 lines RTP bridge away with inband DTMF and feature detection When deciding whether Asterisk was allowed to bridge the call away from the core, chan_sip did not take into account the usage of features on dialed channels that require monitoring of DTMF on channels utilizing inband DTMF. This would cause Asterisk to allow the call to be locally or remotely bridged, preventing access to the data required to detect activations of such features. (closes 17237) Review: https://reviewboard.asterisk.org/r/1302/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.10@328824 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Mark Murawki
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c4b8e21079 |
Merged revisions 328608 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328608 | markm | 2011-07-18 08:35:57 -0400 (Mon, 18 Jul 2011) | 9 lines If the sip private structure is null, sip_setoption() will defref the null pointer and crash. Ideally, sip_setoption shouldn't be called if there is a lack of a sip private structure. But this will fix a crash. (closes issue ASTERISK-17909) Reported by: Mark Murawski Tested by: Mark Murawski ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.10@328611 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Richard Mudgett
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ee2096fe55 |
Make hint watcher callback take const strings for context and exten parameters.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.10@328329 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Richard Mudgett
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cc8daa1385 |
Merged revisions 328302 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328302 | rmudgett | 2011-07-14 18:12:06 -0500 (Thu, 14 Jul 2011) | 6 lines Missing SIP pvt and channel unlock in sip_set_rtp_peer(). Regression introduced by -r326144. Add missing SIP pvt and channel unlock in sip_set_rtp_peer(). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.10@328317 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Leif Madsen
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7caa2349af |
Merged revisions 328209 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines Introduce <support_level> tags in MODULEINFO. This change introduces MODULEINFO into many modules in Asterisk in order to show the community support level for those modules. This is used by changes committed to menuselect by Russell Bryant recently (r917 in menuselect). More information about the support level types and what they mean is available on the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.10@328247 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Richard Mudgett
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0e613fd544 |
Merged revisions 327211 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327211 | rmudgett | 2011-07-08 16:41:58 -0500 (Fri, 08 Jul 2011) | 9 lines INVITE 403 Forbidden response always retransmits the maximum times. Asterisk sends a 403 Forbidden response if authentication fails for an INVITE as required. However, it ignores the ACK and keeps retransmitting the response. * Made not delete the to-tag in the dialog so the expected ACK can be matched with the dialog and stop the retransmissions. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327212 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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David Vossel
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513c680b8c |
Adds pass-through support for codec CELT.
This patch adds pass-through support for CELT. CELT formats are defined in codecs.conf and can be configured to any sample rate a CELT endpoint supports. This patch also addresses a crash in channel.c resulting from a frame list being freed incorrectly. This crash was discovered while testing a CELT translator which had to split encoded audio into multiple frames. The codec translator is not a part of this patch, but may be contributed in the future. Review: https://reviewboard.asterisk.org/r/1294/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326855 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Matthew Nicholson
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ba1cc98f1a |
Merged revisions 326683 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326683 | mnicholson | 2011-07-07 10:28:25 -0500 (Thu, 07 Jul 2011) | 3 lines use sips: or sip: depending on the transport in use when building reply digest URIs ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326684 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Matthew Nicholson
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14553512ee |
Merged revisions 326681 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326681 | mnicholson | 2011-07-07 10:25:49 -0500 (Thu, 07 Jul 2011) | 3 lines make the uri parameter used in reply digests more standards compliant in certain cases by prepending "sip:" or "sips:" to it ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326682 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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David Vossel
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a7c6f0445e |
Fixes newlines from being stripped from out of dialog sip MESSAGES.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326544 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Tilghman Lesher
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7d179abfd4 |
Merged revisions 326411 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326411 | tilghman | 2011-07-05 17:08:29 -0500 (Tue, 05 Jul 2011) | 14 lines Add the attribute "type" to each "<use>" for menuselect. This matters only when autoconf fails to detect that weak linking is supported. External optional dependencies will become optional in both cases, as they are removed at compile time when not detected. However, runtime-optional modules are made mandatory when weak linking is not found. This change affects only the external optional dependencies; previously, they were incorrectly required when weak linking support was not detected. Patches: 20110702__issue18062__asterisk_trunk.diff.txt by tilghman (License #5003) Tested by: iasgoscouk ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326412 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Richard Mudgett
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14d510c5b7 |
Merged revisions 326291 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326291 | rmudgett | 2011-07-05 12:22:59 -0500 (Tue, 05 Jul 2011) | 23 lines Used auth= parameter freed during "sip reload" causes crash. If you use the auth= parameter and do a "sip reload" while there is an ongoing call. The peer->auth data points to free'd memory. The patch does several things: 1) Puts the authentication list into an ao2 object for reference counting to fix the reported crash during a SIP reload. 2) Converts the authentication list from open coding to AST list macros. 3) Adds display of the global authentication list in "sip show settings". (closes issue ASTERISK-17939) Reported by: wdoekes Patches: jira_asterisk_17939_v1.8.patch (license #5621) patch uploaded by rmudgett Review: https://reviewboard.asterisk.org/r/1303/ JIRA SWP-3526 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326321 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Richard Mudgett
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76e4e2e777 |
Merged revisions 326144 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326144 | rmudgett | 2011-07-01 16:07:22 -0500 (Fri, 01 Jul 2011) | 16 lines Better way to get chan and pvt lock for issue ASTERISK-17431. Redoes -r308945 for issue ASTERISK-17431 deadlock fix for sip_set_udptl_peer() and sip_set_rtp_peer(). * Lock the channels in the defined order and avoid the need for a deadlock avoidance loop. * Lock the channel before getting the pointer to the private structure to be sure that the pointer will not change due to a masquerade or channel hangup. * To preserve sanity, check that chan and p->owner are the same. (Pointer rearangements should not happen without the protection of locks because bad things tend to happen otherwise.) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326145 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Richard Mudgett
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39a7152df3 |
Merged revisions 325935 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325935 | rmudgett | 2011-06-30 15:39:45 -0500 (Thu, 30 Jun 2011) | 11 lines Misc minor changes in chan_sip. * Add load failure exit if primary SIP container(s) could not get created in chan_sip.c:load_module(). * Removed a redundant static prototype. * Some typos. * Some whitespace. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325936 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Kinsey Moore
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1d93d217f0 |
Merged revisions 325740 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325740 | kmoore | 2011-06-29 16:49:21 -0500 (Wed, 29 Jun 2011) | 7 lines chan_sip: cleanup from the introduction of ast_str Remove the length field from sip_req and sip_pkt in chan_sip since they are redundant (ast_str holds its own length) and refactor the necessary functions. Review: https://reviewboard.asterisk.org/r/1281/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325741 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
Kevin P. Fleming
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37d6d89d97 |
Merged revisions 325416 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325416 | kpfleming | 2011-06-28 16:50:43 -0500 (Tue, 28 Jun 2011) | 3 lines Fix random misspelling noticed on asterisk-users. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325417 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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David Vossel
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bb4e0c7f7c |
Merged revisions 325339 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325339 | dvossel | 2011-06-28 15:31:00 -0500 (Tue, 28 Jun 2011) | 4 lines Fixes locking inversion caused by holding sip pvt lock during async_goto. (closes ASTERISK-17352) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325345 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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David Vossel
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4812697542 |
Fixes issue with video and text not being reinvited correctly with directmedia
If a SDP does not modify the session, we ignore it. However, we were defaulting no text and video support to true before checking to see if the sdp modified anything or not. This would result in process_sdp ignoring an sdp but removing video and text from the call during direct media reinvites. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325151 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Terry Wilson
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04fc1c6cea |
Don't forget to build the Via when sending MESSAGE
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325046 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Richard Mudgett
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04226479b3 |
Merged revisions 324914 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324914 | rmudgett | 2011-06-27 10:37:19 -0500 (Mon, 27 Jun 2011) | 21 lines When subscribing MWI to an unsolicited mailbox the first notification is incorrect. A remote peer subscribed to MWI with the unsolicited option and a local phone subscribed to the remote mailbox. The notify message-summary events are sent correctly except for the first one when subscribing, which will always be 0. This means the phone MWI indicator will be wrong until the mailbox read/unread count changes and the event is fired. Looks like this is a regression from ASTERISK-16149. * Fix the logic to check the cache and if allowed then fallback to manually counting mailbox messages. (closes issue ASTERISK-17997) Reported by: rsw686 Patches: jira_asterisk_17997_v1.8.patch (license #5621) uploaded by rmudgett Tested by: rsw686 JIRA SWP-3551 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324915 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Kinsey Moore
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3c10d69544 |
Merged revisions 324678 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r324678 | kmoore | 2011-06-23 13:29:17 -0500 (Thu, 23 Jun 2011) | 11 lines Merged revisions 324643 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r324643 | kmoore | 2011-06-23 13:21:12 -0500 (Thu, 23 Jun 2011) | 4 lines Addresses AST-2011-008, memory corruption and remote crash in SIP driver. AST-2011-008 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324708 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Richard Mudgett
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10480072aa |
Merged revisions 324491 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324491 | rmudgett | 2011-06-22 14:16:29 -0500 (Wed, 22 Jun 2011) | 1 line Use correct variable for text SRTP media. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324495 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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Terry Wilson
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385b8c6f8b |
Merged revisions 324484 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324484 | twilson | 2011-06-22 13:52:04 -0500 (Wed, 22 Jun 2011) | 20 lines Stop sending IPv6 link-local scope-ids in SIP messages The idea behind the patch listed below was used, but in a more targeted manner. There are now address stringification functions for addresses that are meant to be sent to a remote party. Link-local scope-ids only make sense on the machine from which they originate and so are stripped in the new functions. There is also a host sanitization function added to chan_sip which is used for when peer and dialog tohost fields or sip_registry hostnames are used to craft a SIP message. Also added are some basic unit tests for netsock2 address parsing. (closes issue ASTERISK-17711) Reported by: ch_djalel Patches: asterisk-1.8.3.2-ipv6_ll_scope.patch uploaded by ch_djalel (license 1251) Review: https://reviewboard.asterisk.org/r/1278/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324487 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |