mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
5774 Commits (a3d2d156eec8a99cbb9c9fe407dd7f44dac8fcbd)
| Author | SHA1 | Message | Date |
|---|---|---|---|
David Vossel
|
a3d2d156ee |
Merged revisions 201344 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r201344 | dvossel | 2009-06-17 10:20:26 -0500 (Wed, 17 Jun 2009) | 16 lines SIP registry ref count error During a sip reload, the list of sip_registry objects are supposed to be traversed, unlinked, and destroyed, but destruction never takes place due to a ref counting error. This causes a memory leak when registry items are removed from sip.conf and reloaded. While the registries are removed from the global list, they are not removed from the scheduler. Because of this, SIP register attempts continue to be sent out for the item even though it may no longer be in the .conf. (closes issue #15295) Reported by: amorsen Review: https://reviewboard.asterisk.org/r/282/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@201365 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
David Vossel
|
f5fca5c8e1 |
Merged revisions 201223 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r201223 | dvossel | 2009-06-16 17:29:30 -0500 (Tue, 16 Jun 2009) | 2 lines fix issue with build_contact introduced by the "SIP trasnport type issues" commit ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@201225 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
David Vossel
|
8e5e00bd07 |
Merged revisions 200946 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r200946 | dvossel | 2009-06-16 11:03:30 -0500 (Tue, 16 Jun 2009) | 32 lines SIP transport type issues What this patch addresses: 1. ast_sip_ouraddrfor() by default binds to the UDP address/port reguardless if the sip->pvt is of type UDP or not. Now when no remapping is required, ast_sip_ouraddrfor() checks the sip_pvt's transport type, attempting to set the address and port to the correct TCP/TLS bindings if necessary. 2. It is not necessary to send the port number in the Contact header unless the port is non-standard for the transport type. This patch fixes this and removes the todo note. 3. In sip_alloc(), the default dialog built always uses transport type UDP. Now sip_alloc() looks at the sip_request (if present) and determines what transport type to use by default. 4. When changing the transport type of a sip_socket, the file descriptor must be set to -1 and in some cases the tcptls_session's ref count must be decremented and set to NULL. I've encountered several issues associated with this process and have created a function, set_socket_transport(), to handle the setting of the socket type. (closes issue #13865) Reported by: st Patches: dont_add_port_if_tls.patch uploaded by Kristijan (license 753) 13865.patch uploaded by mmichelson (license 60) tls_port_v5.patch uploaded by vrban (license 756) transport_issues.diff uploaded by dvossel (license 671) Tested by: mmichelson, Kristijan, vrban, jmacz, dvossel Review: https://reviewboard.asterisk.org/r/278/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@200987 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Kevin P. Fleming
|
7375533824 |
Merged revisions 165180,200689 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r165180 | mnicholson | 2008-12-17 12:49:12 -0600 (Wed, 17 Dec 2008) | 14 lines This patch adds a new 'ignoresdpversion' option to sip.conf. When this is enabled (either globally or for a specific peer), chan_sip will treat any SDP data it receives as new data and update the media stream accordingly. By default, Asterisk will only modify the media stream if the SDP session version received is different from the current SDP session version. This option is required to interoperate with devices that have non-standard SDP session version implementations (observed by toc on the bug tracker with Microsoft OCS which always uses 0 as the session version). http://reviewboard.digium.com/r/94/ (closes issue #13958) Reported by: toc Tested by: toc ........ r200689 | kpfleming | 2009-06-15 15:42:38 -0500 (Mon, 15 Jun 2009) | 12 lines Accept T.38 re-INVITE responses with invalid SDP versions. This commit changes the 'incoming SDP version' check logic a bit more; when 'ignoresdpversion' is *not* set for a peer, if we initiate a re-INVITE to switch to T.38, we'll always accept the peer's SDP response, even if they don't properly increment the SDP version number as they should. If this situation occurs, a warning message will be generated suggesting that the peer's configuration be changed to include the 'ignoresdpversion' configuration option (although ideally they'd fix their SIP implementation to be RFC compliant). AST-221 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@200707 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Mark Michelson
|
369810c36c |
Merged revisions 200514 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r200514 | mmichelson | 2009-06-15 10:22:11 -0500 (Mon, 15 Jun 2009) | 11 lines Merged revisions 200513 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200513 | mmichelson | 2009-06-15 10:21:46 -0500 (Mon, 15 Jun 2009) | 5 lines Add INFO to our allowed methods so that endpoints know they may send it to us. AST-223 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@200516 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
|
cb76dba60a |
Merged revisions 200146 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r200146 | mmichelson | 2009-06-11 16:17:14 -0500 (Thu, 11 Jun 2009) | 5 lines Fix a crash due to a potentially NULL p->options. Thanks to mnicholson for pointing it out. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@200152 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Mark Michelson
|
b95f51e4fc |
Merged revisions 199958 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r199958 | mmichelson | 2009-06-10 15:15:48 -0500 (Wed, 10 Jun 2009) | 6 lines Only try to use the invite_branch on outgoing INVITEs with auth credentials. I have added a comment to the code to help ease understanding of the logic here as well. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@199963 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
David Vossel
|
64af4b8465 |
Merged revisions 199818 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r199818 | dvossel | 2009-06-09 15:47:57 -0500 (Tue, 09 Jun 2009) | 11 lines CLI NOTIFY sending wrong transport type. SIP's cli NOTIFY command only used UDP rather than copying the transport type from the peer. (closes issue #15283) Reported by: jthurman Patches: sip-notify-tcp-svn199728.patch uploaded by jthurman (license 614) Tested by: jthurman, dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@199820 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Mark Michelson
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87eda713ad |
Recorded merge of revisions 199588 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r199588 | mmichelson | 2009-06-08 12:32:04 -0500 (Mon, 08 Jun 2009) | 9 lines Fix a deadlock that could occur when setting rtp stats on SIP calls. (closes issue #15143) Reported by: cristiandimache Patches: 15143.patch uploaded by mmichelson (license 60) Tested by: cristiandimache ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@199590 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Mark Michelson
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64097edf92 |
Merged revisions 199227 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r199227 | mmichelson | 2009-06-05 08:51:08 -0500 (Fri, 05 Jun 2009) | 14 lines Correct "dahdi show channels" output when specifying a group. Since a DAHDI channel may belong to multiple groups, we need to use a bitwise and instead of equivalence to determine whether to display the channel information. (closes issue #15248) Reported by: gentian Patches: 15248.patch uploaded by mmichelson (license 60) Tested by: gentian ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@199229 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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David Vossel
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0a9c235bc1 |
Merged revisions 199139 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r199139 | dvossel | 2009-06-04 14:10:16 -0500 (Thu, 04 Jun 2009) | 9 lines Merged revisions 199138 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199138 | dvossel | 2009-06-04 14:00:15 -0500 (Thu, 04 Jun 2009) | 3 lines Additional updates to AST-2009-001 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@199141 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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David Vossel
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a313821999 |
Merged revisions 198824 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r198824 | dvossel | 2009-06-02 12:55:35 -0500 (Tue, 02 Jun 2009) | 8 lines fixes issue with channels not going down after transfer Iax2 currently does not support native bridging if the timeoutms value is set. We check for that in iax2_bridge, but then set timeoutms to 0 by default. If the timeoutms is not provided it is set to -1. By setting timeoutms to 0 it is processed causing a bridging retry loop. (closes issue #15216) Reported by: oxymoron Tested by: dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@198826 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Joshua Colp
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fcdc8c20f4 |
Merged revisions 198791 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r198791 | file | 2009-06-02 10:48:06 -0300 (Tue, 02 Jun 2009) | 5 lines Correct documentation for the register line, specifically where the domain should be specified. (closes issue #14367) Reported by: Nick_Lewis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@198793 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
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90dfe15ab7 |
Merged revisions 198248 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r198248 | file | 2009-05-29 23:31:48 -0300 (Fri, 29 May 2009) | 2 lines When removing all packets from a dialog we also need to free the data if present. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@198249 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
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8706b4ad69 |
Merged revisions 197697 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r197697 | file | 2009-05-28 15:45:11 -0300 (Thu, 28 May 2009) | 2 lines Fix a bug where the trunkmtu setting was not set to the default value of 1240 on load but was on reload. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@197700 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Eliel C. Sardanons
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36915a8789 |
Merged revisions 197621 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r197621 | eliel | 2009-05-28 12:01:48 -0400 (Thu, 28 May 2009) | 19 lines Merged revisions 197562 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197562 | eliel | 2009-05-28 11:21:32 -0400 (Thu, 28 May 2009) | 13 lines Use the address we already know when reloading a peer with nat=yes. If we already have an address for a peer, and we are reloading the sip configuration, try to use that address to contact the peer, instead of getting it from the Contact. (closes issue #15194) Reported by: ibc Patches: sip.patch uploaded by eliel (license 64) Tested by: manwe ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@197696 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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David Vossel
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ddba5b90b0 |
'iax show peer blah' now outputs whether or not peer 'blah' is in trunk mode or not.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@197623 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Mark Michelson
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faaeca2980 |
Merged revisions 197606 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r197606 | mmichelson | 2009-05-28 10:32:19 -0500 (Thu, 28 May 2009) | 22 lines Recorded merge of revisions 197588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197588 | mmichelson | 2009-05-28 10:27:49 -0500 (Thu, 28 May 2009) | 16 lines Allow for media to arrive from an alternate source when responding to a reinvite with 491. When we receive a SIP reinvite, it is possible that we may not be able to process the reinvite immediately since we have also sent a reinvite out ourselves. The problem is that whoever sent us the reinvite may have also sent a reinvite out to another party, and that reinvite may have succeeded. As a result, even though we are not going to accept the reinvite we just received, it is important for us to not have problems if we suddenly start receiving RTP from a new source. The fix for this is to grab the media source information from the SDP of the reinvite that we receive. This information is passed to the RTP layer so that it will know about the alternate source for media. Review: https://reviewboard.asterisk.org/r/252 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@197618 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
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815067bf3e |
Merged revisions 197467 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r197467 | file | 2009-05-28 10:47:45 -0300 (Thu, 28 May 2009) | 15 lines Merged revisions 197466 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197466 | file | 2009-05-28 10:44:58 -0300 (Thu, 28 May 2009) | 8 lines Fix a bug where the flag indicating the presence of rport would get overwritten by the nat setting. The presence of rport is now stored as a separate flag. Once the dialog is setup and authenticated (or it passes through unauthenticated) the proper nat flag is set. (closes issue #13823) Reported by: dimas ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@197470 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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David Vossel
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cb1b99ac9c |
Fixes merge issue for r196453.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@197087 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Sean Bright
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70b31d202a |
Merged revisions 196988 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r196988 | seanbright | 2009-05-27 09:02:54 -0400 (Wed, 27 May 2009) | 9 lines Display an error message when chan_alsa fails to load due to a missing or inaccessible configuration file. Before this change, when chan_alsa failed to load due to a missing or inaccessible configuration file, no message would be displayed. With this change, when chan_alsa fails to load due to a missing or inaccessible configuration file, a message will be displayed. (closes issue #14760) Reported by: Nick_Lewis Patches: chan_alsa.c-confload.patch uploaded by Nick (license 657) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@196990 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
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4a63041eaf |
Merged revisions 196721 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r196721 | file | 2009-05-26 10:43:13 -0300 (Tue, 26 May 2009) | 7 lines Fix a bug where the sip unregister CLI command did not completely unregister the peer. (closes issue #15118) Reported by: alecdavis Patches: chan_sip_unregister.diff2.txt uploaded by alecdavis (license 585) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@196723 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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David Vossel
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28a71581e0 |
Merged revisions 196416 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r196416 | dvossel | 2009-05-22 16:09:45 -0500 (Fri, 22 May 2009) | 19 lines SIP set outbound transport type from Registration In sip.conf the transport option allows for the configuration of what transport types (udp, tcp, and tls) a peer will accept, but only the first type listed was used for outbound connections. This patch changes this. Now the default transport type is only used until the peer registers. When registration takes place the transport type is parsed out of the Contact header. If the Contact header's transport type is equal to one that the peer supports, the peer's default transport type for outbound connections is set to match the Contact header's type. If the Contact header's transport type is not present, then the peer's default transport type is set to match the one the peer registered with. When a peer unregisters or the registration expires, the default transport type for that peer is reset. (closes issue #12282) Reported by: rjain Patches: reg_patch_1.diff uploaded by dvossel (license 671) Tested by: dvossel (closes issue #14727) Reported by: pj Patches: reg_patch_3.diff uploaded by dvossel (license 671) Tested by: pj, dvossel Review: https://reviewboard.asterisk.org/r/249/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@196453 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
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aee4cf5902 |
Merged revisions 196117 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r196117 | file | 2009-05-22 10:56:47 -0300 (Fri, 22 May 2009) | 12 lines Merged revisions 196116 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r196116 | file | 2009-05-22 10:54:17 -0300 (Fri, 22 May 2009) | 5 lines Fix a bug where using immediate with mISDN caused a cause code of 16 to get sent back instead of 1 if the 's' extension did not exist. (closes issue #12286) Reported by: lmamane ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@196119 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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David Vossel
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456242c645 |
Merged revisions 195995 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r195995 | dvossel | 2009-05-21 14:11:49 -0500 (Thu, 21 May 2009) | 20 lines Merged revisions 195991 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195991 | dvossel | 2009-05-21 14:04:56 -0500 (Thu, 21 May 2009) | 14 lines Sign problem calculating timestamp for iax frame leads to no audio on the receiving peer. There are rare cases in which a frame's delivery timestamp is slightly less than the iax2_pvt's offset. This causes the pvt's timestamp to be a small negative number, but since the timestamp value is unsigned it looks like a huge positive number. This patch checks for this negative case and sets the ms to zero. A similar check is already done right below this one in the 'else' statement. (closes issue #15032) Reported by: guillecabeza Patches: chan_iax2.c.patch_timestamp uploaded by guillecabeza (license 380) Tested by: guillecabeza (closes issue #14216) Reported by: Andrey Sofronov ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@195998 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
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26087fc760 |
Merged revisions 195449 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r195449 | file | 2009-05-19 11:43:54 -0300 (Tue, 19 May 2009) | 14 lines Merged revisions 195448 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195448 | file | 2009-05-19 11:41:45 -0300 (Tue, 19 May 2009) | 7 lines Fix a bug where direct RTP setup would partially occur even when disabled if the calling channel was answered. (issue #13545) Reported by: davidw (issue #14244) Reported by: mbnwa ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@195451 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
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7d2da8cec8 |
Merged revisions 195089 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r195089 | file | 2009-05-18 10:36:17 -0300 (Mon, 18 May 2009) | 5 lines Fix a bug where specifying an empty outboundproxy would cause packets to get sent to ourself. (closes issue #15106) Reported by: timeshell ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@195091 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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David Vossel
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fa29e4c3fc |
Merged revisions 194874 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r194874 | dvossel | 2009-05-15 17:44:44 -0500 (Fri, 15 May 2009) | 23 lines Merged revisions 194873 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194873 | dvossel | 2009-05-15 17:43:13 -0500 (Fri, 15 May 2009) | 17 lines IAX2 REGAUTH loop IAX was not sending REGREJ to terminate invalid registrations. Instead it sent another REGAUTH if the authentication challenge failed. This caused a loop of REGREQ and REGAUTH frames. (Related to Security fix AST-2009-001) (closes issue #14867) Reported by: aragon Tested by: dvossel (closes issue #14717) Reported by: mobeck Patches: regauth_loop_update_patch.diff uploaded by dvossel (license 671) Tested by: dvossel ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@194876 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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David Vossel
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f8538620ab |
Merged revisions 194833 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r194833 | dvossel | 2009-05-15 15:52:12 -0500 (Fri, 15 May 2009) | 24 lines Merged revisions 194557,194685 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194557 | dvossel | 2009-05-14 17:59:43 -0500 (Thu, 14 May 2009) | 10 lines IAX2 "Ghost" Channels There is a bug tracker issue where people are reporting "Ghost" channels in their 'iax2 show channels' output. The confusion is caused by channels being listed as "(NONE)" with format "unknown". These are not channels of coarse. They are usually just pending registration or poke requests, but it is confusing output. To help make sense of this I have added two columns to 'iax2 show channels'. One shows the first message which started the transaction, and the second shows the last message sent by either side of the call. This helps diagnose why the entry exists and why it may not go away. (closes issue #14207) Reported by: clive18 Review: https://reviewboard.asterisk.org/r/246/ ........ r194685 | dvossel | 2009-05-15 10:40:37 -0500 (Fri, 15 May 2009) | 6 lines Update to previous IAX2 "Ghost" Channels patch. Fixed some comments made on reviewboard for the previous patch. (issue #14207) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@194835 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Mark Michelson
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0fb8658cbe |
Merged revisions 194496 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r194496 | mmichelson | 2009-05-14 17:20:51 -0500 (Thu, 14 May 2009) | 30 lines Merged revisions 194484 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194484 | mmichelson | 2009-05-14 17:17:55 -0500 (Thu, 14 May 2009) | 24 lines Fix a race condition where a reinvite could trigger a 482 response. The loop detection/spiral detection code in chan_sip used the owner channel's state as a criterion for determining if the incoming INVITE is a looped request. The problem with this is that the INVITE-handling code happens in a different thread than the thread that marks the owner channel as being up. As a result, if a reinvite were to come in very quickly, say from another Asterisk on the same LAN, it was possible for the reinvite to arrive before the owner channel had been set to the up state. This patch corrects the problem by using the invitestate of the sip_pvt instead, since that can be guaranteed to be set correctly by the time the reinvite arrives. Since there is a switch statement further in the INVITE-handling code, the AST_STATE_RINGING state also checks the invitestate of the sip_pvt in case we should actually be treating the channel as if it were up already. (closes issue #12215) Reported by: jpyle Patches: 12215_confirmed.patch uploaded by mmichelson (license 60) Tested by: lmadsen ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@194507 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Mark Michelson
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5107dfdcbd |
Merged revisions 193954 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r193954 | mmichelson | 2009-05-12 15:28:13 -0500 (Tue, 12 May 2009) | 18 lines Update spiral support in trunk and 1.6.X to match what is in 1.4. In 1.4, a SIP spiral is treated the same way as a call forward. This works much better than what is currently in trunk and 1.6.X. The code in trunk and 1.6.X did not create a new call to the recipient of the spiral, instead trying to continue the same call. In addition to just being plain wrong, this also had the side effect of only being able to spiral calls to other SIP channels. With this in place, as long as call forwards are honored, SIP spirals will work properly. This means that it will work for outbound calls made by the Queue, Dial, and Page applications. For originated calls and spool calls, however, the spiral will not work properly until a generic call forward mechanism is introduced into Asterisk. (relates to issue #13630) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@193961 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Richard Mudgett
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77974e657f |
Merged revisions 193614 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r193614 | rmudgett | 2009-05-11 14:11:29 -0500 (Mon, 11 May 2009) | 19 lines Merged revisions 193613 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193613 | rmudgett | 2009-05-11 14:09:00 -0500 (Mon, 11 May 2009) | 12 lines Sent wrong message to clear a call we started if the other end has not responed yet. In the state MISDN_CALLING (i.e. SETUP was sent but no answer has arrived yet), it is not allowed to clear the call with RELEASE_COMPLETE. It must be cleared with DISCONNECT. A RELEASE_COMPLETE is only allowed as an answer to a SETUP. (See Q.931 ch. 5.3.2, 5.3.2.a, 5.3.2.b) Patches: chan-misdn-ccstate7.patch uploaded by customer. JIRA ABE-1862 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@193616 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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David Vossel
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2a1045148c |
Merged revisions 193387 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r193387 | dvossel | 2009-05-08 15:32:51 -0500 (Fri, 08 May 2009) | 7 lines TCP not matching valid peer. find_peer() does not find a valid peer when using pvt->recv as the sockaddr_in argument. Because of the way TCP works, the port number in pvt->recv is not what we're looking for at all. There is currently only one place that find_peer searches for a peer using the sockaddr_in argument. If the peer is not found after using pvt->recv (works for UDP since the port number will be correct), a temp sockaddr_in struct is made using the Contact header in the sip_request. This has the correct port number in it. Review: http://reviewboard.digium.com/r/236/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@193389 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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David Vossel
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40841203d4 |
Merged revisions 193263 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r193263 | dvossel | 2009-05-08 09:52:19 -0500 (Fri, 08 May 2009) | 15 lines Merged revisions 193262 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193262 | dvossel | 2009-05-08 09:51:09 -0500 (Fri, 08 May 2009) | 9 lines "misdn show config" segfaults asterisk, if no MSN lists (closes issue #14976) Reported by: alecdavis Patches: misdn_config.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis, FabienToune ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@193265 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Richard Mudgett
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2826d3dea3 |
Merged revisions 193077 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r193077 | rmudgett | 2009-05-07 17:24:04 -0500 (Thu, 07 May 2009) | 12 lines Merged revisions 193050 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193050 | rmudgett | 2009-05-07 17:17:06 -0500 (Thu, 07 May 2009) | 5 lines Give a more helpful message when an incoming call's dialed extension does not match. Added the dialed extension and context to the chan_misdn messages warning that the dialed number cannot be matched in the dialplan. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@193079 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Tilghman Lesher
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1d63fab6f8 |
Merged revisions 192938 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r192938 | tilghman | 2009-05-07 12:13:36 -0500 (Thu, 07 May 2009) | 6 lines Send DTMF frame before playing back audio. (closes issue #14858) Reported by: barryf Patches: 20090507__bug14858.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@192941 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Tilghman Lesher
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fc6b76aa20 |
Merged revisions 192933 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r192933 | tilghman | 2009-05-07 11:43:56 -0500 (Thu, 07 May 2009) | 17 lines Merged revisions 192932 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192932 | tilghman | 2009-05-07 11:29:08 -0500 (Thu, 07 May 2009) | 10 lines Eliminate repetition of fullcontact during reconstruction. If the fullcontact field appears in both the sippeers and the sipregs table, then during reconstruction of the field, it will otherwise be doubled. (closes issue #14754) Reported by: Alexei Gradinari Patches: 20090506__bug14754.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@192935 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Matthew Fredrickson
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2532a5d7a6 |
Merged revisions 190946 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r190946 | mattf | 2009-04-28 17:05:05 -0500 (Tue, 28 Apr 2009) | 1 line Make sure that we do not clear the down flag on the BRI during PTMP link transients. Also refix SS7 audio that the early media patch broke. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@192812 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
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0b8f27066a |
Merged revisions 192808 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r192808 | file | 2009-05-06 14:38:51 -0300 (Wed, 06 May 2009) | 10 lines Fix a bug where a timer would be created but not acknowledged. This scenario crept up if chan_iax2 was loaded with no configuration file present. It would create a timer and tell it to go at an interval but the thread that normally acknowledges it would not be created because no configuration file was present. The timer will now be closed if no configuration file is present. (closes issue #15014) Reported by: madkins ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@192809 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
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3201a8d6a0 |
Merged revisions 192634 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r192634 | file | 2009-05-06 10:34:35 -0300 (Wed, 06 May 2009) | 14 lines Merged revisions 192633 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192633 | file | 2009-05-06 10:30:51 -0300 (Wed, 06 May 2009) | 7 lines Update some old logic to stop both begin and end DTMF frames from reaching the core if rfc2833 is not enabled. (closes issue #15036) Reported by: dimas Patches: v1-15036.patch uploaded by dimas (license 88) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@192636 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
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883b290df3 |
Merged revisions 192387 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r192387 | file | 2009-05-05 11:22:47 -0300 (Tue, 05 May 2009) | 10 lines Fix a bug with setting t38pt_udptl at the user or peer level. If an incoming call authenticated as a user or peer and t38pt_udptl was not set to yes in general then no UDPTL session would be present and any T38 related things would fail. This commit changes it so that if after authenticating T38 is enabled but no UDPTL session is present one will be created. (issue AST-215) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@192401 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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David Vossel
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ed00c79ed0 |
Merged revisions 192214 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r192214 | dvossel | 2009-05-04 17:44:51 -0500 (Mon, 04 May 2009) | 17 lines Merged revisions 192213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192213 | dvossel | 2009-05-04 17:37:31 -0500 (Mon, 04 May 2009) | 11 lines global mohinterpret setting is ignored mohinterpret and mohsuggest global variables were not copied over during build_users and build_peers. (closes issue #14728) Reported by: dimas Patches: v1-14728.patch uploaded by dimas (license 88) Tested by: dimas, dvossel ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@192216 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Tilghman Lesher
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226719ab81 |
Merged revisions 191560 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r191560 | tilghman | 2009-05-01 15:01:21 -0500 (Fri, 01 May 2009) | 13 lines Merged revisions 191559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r191559 | tilghman | 2009-05-01 15:00:23 -0500 (Fri, 01 May 2009) | 6 lines SIP Response 410 maps to cause code 22 (or 23), not 1. (closes issue #14993) Reported by: BigJimmy Patches: causepatch uploaded by BigJimmy (license 371) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@191562 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Tilghman Lesher
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bd33e2b9f3 |
Merged revisions 191494 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r191494 | tilghman | 2009-05-01 13:18:00 -0500 (Fri, 01 May 2009) | 4 lines Set debug message back to DEBUG level. (closes issue #15007) Reported by: hulber ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@191553 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Tilghman Lesher
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88eae5d322 |
Merged revisions 191219 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r191219 | tilghman | 2009-04-29 18:06:56 -0500 (Wed, 29 Apr 2009) | 2 lines Make H.323 compile with FDLEAK detection code enabled ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@191223 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Russell Bryant
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f205cc4041 |
Merged revisions 190357 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r190357 | russell | 2009-04-23 16:13:07 -0500 (Thu, 23 Apr 2009) | 10 lines Merged revisions 190356 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r190356 | russell | 2009-04-23 16:07:07 -0500 (Thu, 23 Apr 2009) | 2 lines Remove a bogus ast_channel_unlock(). ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@190371 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
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988bf263dd |
Merged revisions 190287 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r190287 | file | 2009-04-23 16:15:30 -0300 (Thu, 23 Apr 2009) | 13 lines Merged revisions 190286 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r190286 | file | 2009-04-23 16:13:18 -0300 (Thu, 23 Apr 2009) | 6 lines Fix a bug in chan_local glare hangup detection. If both sides of a Local channel were hung up at around the same time it was possible for one thread to destroy the local private structure and have the other thread immediately try to remove the already freed structure from the local channel list. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@190291 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Jeff Peeler
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0fc1e98188 |
Merged revisions 189993 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r189993 | jpeeler | 2009-04-22 14:23:49 -0500 (Wed, 22 Apr 2009) | 18 lines Make chan_h323 respect packetization settings Previously, packetization settings were ignored and now they are not. A new config option 'autoframing' has been added to mirror the way chan_sip handles it. Turning on the autoframing option (available both as a global option or per peer) overrides the local settings with the remote packetization settings. Testing was performed with varying packetization levels with the following codecs: ulaw, alaw, gsm, and g729. (closes issue #12415) Reported by: pj Patches: 2009012200_h323packetization.diff.txt uploaded by mvanbaak (license 7), modified by me ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@189996 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Tilghman Lesher
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80ac94cb45 |
Merged revisions 189911 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r189911 | tilghman | 2009-04-22 11:01:30 -0500 (Wed, 22 Apr 2009) | 7 lines Do not continue to receive DTMF, when the channel is hungup and about to be destroyed. (closes issue #14858) Reported by: barryf Patches: 20090421__bug14858.diff.txt uploaded by tilghman (license 14) Tested by: barryf ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@189913 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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David Vossel
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8c665aa1af |
Merged revisions 189771 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r189771 | dvossel | 2009-04-21 15:28:37 -0500 (Tue, 21 Apr 2009) | 11 lines Fixes segfault when switching UDP to TCP in sip.conf after reload. If transport in sip.conf is switched from UDP to TCP, Asterisk segfaults right after issuing a sip reload. The problem is the socket type is changed to TCP but the fd may still be present for UDP. Later, when the TCP session should be created or set using an existing one, it isn't because the old file descriptor is still present. Now every time transport is changed during a sip.conf reload, the file descriptor is set to -1, signifying it must be created or found. (closes issue #14727) Reported by: pj Tested by: dvossel Review: http://reviewboard.digium.com/r/229/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@189774 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |