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${ noResults }
2668 Commits (4661a27d339f6e31c84e152fd618a0709550cc8c)
| Author | SHA1 | Message | Date |
|---|---|---|---|
David Vossel
|
4661a27d33 |
Merged revisions 223132 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r223132 | dvossel | 2009-10-09 11:54:02 -0500 (Fri, 09 Oct 2009) | 9 lines 'auth=' did not parse md5 secret correctly (closes issue #15949) Reported by: ebroad Patches: authparsefix.patch uploaded by ebroad (license 878) 15949_trunk.diff uploaded by dvossel (license 671) Tested by: ebroad ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@223133 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
David Vossel
|
bdadc892dd |
Merged revisions 223088 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r223088 | dvossel | 2009-10-09 10:49:30 -0500 (Fri, 09 Oct 2009) | 14 lines p->peerauth is always empty in transmit_register() When using callbackextension or specifing the peer name in a registration string, the peer's specific auth settings set by the "auth=" strings within the peer definition are not used by the registration. Thanks to ebroad for reporting the issue and providing the patch. (closes issue #15955) Reported by: ebroad Patches: regauthfix.patch uploaded by ebroad (license 878) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@223089 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
David Vossel
|
1409dd6481 |
Merged revisions 222543 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r222543 | dvossel | 2009-10-07 12:44:52 -0500 (Wed, 07 Oct 2009) | 14 lines Merged revisions 222542 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222542 | dvossel | 2009-10-07 12:41:21 -0500 (Wed, 07 Oct 2009) | 8 lines crash on transfer handle_invite_replaces() attempts to uplock a pvt's owner channel without first verifing that it exists. (issue #16027) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@222544 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Kevin P. Fleming
|
6410aa2418 |
Merged revisions 222176 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r222176 | kpfleming | 2009-10-05 20:24:24 -0500 (Mon, 05 Oct 2009) | 27 lines Recorded merge of revisions 222152 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct 2009) | 20 lines Fix ao2_iterator API to hold references to containers being iterated. See Mantis issue for details of what prompted this change. Additional notes: This patch changes the ao2_iterator API in two ways: F_AO2I_DONTLOCK has become an enum instead of a macro, with a name that fits our naming policy; also, it is now necessary to call ao2_iterator_destroy() on any iterator that has been created. Currently this only releases the reference to the container being iterated, but in the future this could also release other resources used by the iterator, if the iterator implementation changes to use additional resources. (closes issue #15987) Reported by: kpfleming Review: https://reviewboard.asterisk.org/r/383/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@222187 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Kevin P. Fleming
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ba54bb0506 |
Merged revisions 222110 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r222110 | kpfleming | 2009-10-05 14:45:00 -0500 (Mon, 05 Oct 2009) | 25 lines Allow non-compliant T.38 endpoints to be supportable via configuration option. Many T.38 endpoints incorrectly send the maximum IFP frame size they can accept as the T38FaxMaxDatagram value in their SDP, when in fact this value is supposed to be the maximum UDPTL payload size (datagram size) they can accept. If the value they supply is small enough (a commonly supplied value is '72'), T.38 UDPTL transmissions will likely fail completely because the UDPTL packets will not have enough room for a primary IFP frame and the redundancy used for error correction. If this occurs, the Asterisk UDPTL stack will emit log messages warning that data loss may occur, and that the value may need to be overridden. This patch extends the 't38pt_udptl' configuration option in sip.conf to allow the administrator to override the value supplied by the remote endpoint and supply a value that allows T.38 FAX transmissions to be successful with that endpoint. In addition, in any SIP call where the override takes effect, a debug message will be printed to that effect. This patch also removes the T38FaxMaxDatagram configuration option from udptl.conf.sample, since it has not actually had any effect for a number of releases. In addition, this patch cleans up the T.38 documentation in sip.conf.sample (which incorrectly documented that T.38 support was passthrough only). (issue #15586) Reported by: globalnetinc ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@222113 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Tilghman Lesher
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3f4db95d71 |
Merged revisions 221705 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r221705 | tilghman | 2009-10-01 15:09:46 -0500 (Thu, 01 Oct 2009) | 2 lines Revision 220906 (a merge from 1.4) was not merged correctly, causing a problem with non-dynamic peers. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@221744 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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7e94f287c4 |
Merged revisions 221697 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r221697 | dvossel | 2009-10-01 14:33:33 -0500 (Thu, 01 Oct 2009) | 9 lines outbound tls connections were not defaulting to port 5061 (closes issue #15854) Reported by: dvossel Patches: sip_port_config_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@221698 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Matthew Nicholson
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4773fdfdf5 |
Merged revisions 221554,221589 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r221554 | oej | 2009-10-01 02:00:04 -0500 (Thu, 01 Oct 2009) | 3 lines Simplify code for porturi, use TRUE/FALSE constructs when it's just TRUE or FALSE. ................ r221589 | mnicholson | 2009-10-01 10:26:20 -0500 (Thu, 01 Oct 2009) | 9 lines Merged revisions 221588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221588 | mnicholson | 2009-10-01 10:24:00 -0500 (Thu, 01 Oct 2009) | 2 lines Use unsigned ints for portinuri flags. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@221660 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Matthew Nicholson
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2e8eebee56 |
Merged revisions 221484 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r221484 | mnicholson | 2009-09-30 18:04:03 -0500 (Wed, 30 Sep 2009) | 2 lines Cleaned up merge from r221432 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@221485 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Matthew Nicholson
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5d6d3d7676 |
Merged revisions 221432 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r221432 | mnicholson | 2009-09-30 15:40:20 -0500 (Wed, 30 Sep 2009) | 17 lines Merged revisions 221360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221360 | mnicholson | 2009-09-30 14:36:06 -0500 (Wed, 30 Sep 2009) | 10 lines Fix SRV lookup and Request-URI generation in chan_sip. This patch adds a new field "portinuri" to the sip dialog struct and the sip peer struct. That field is used during RURI generation to determine if the port should be included in the RURI. It is also used in some places to determine if an SRV lookup should occur. (closes issue #14418) Reported by: klaus3000 Tested by: klaus3000, mnicholson Review: https://reviewboard.asterisk.org/r/369/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@221477 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Terry Wilson
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9a61ba3e28 |
Merged revisions 221266 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r221266 | twilson | 2009-09-30 12:52:30 -0500 (Wed, 30 Sep 2009) | 32 lines Merged revisions 221086 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) | 25 lines Change the SSRC by default when our media stream changes Be default, change SSRC when doing an audio stream changes Asterisk doesn't honor marker bit when reinvited to already-bridged RTP streams,resulting in far-end stack discarding packets with "old" timestamps that areactually part of a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is a reinvite, unless the 'constantssrc' is set to true in sip.conf. The original issue reported to Digium support detailed the following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in fromITSP, Asterisk dials the app server which sends a re-invite back toAsterisk--not to negotiate to send media directly to the ITSP, but to indicatethat it's changing the stream it's sending to Asterisk. The app servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker bit on the new stream. Asterisk passes through the teimstamp of the new stream, butdoes not reset the SSRC, sequence numbers, or set the marker bit. When the timestamp on the new stream is older than the timestamp on the originalstream, the ITSP (which doesn't know there has been any change) discards the newframes because it thinks they are too old. This patch addresses this by changing the SSRC on a stream update unless constantssrc=true is set in sip.conf. Review: https://reviewboard.asterisk.org/r/374/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@221304 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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f51a43531a |
Merged revisions 220906 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r220906 | tilghman | 2009-09-29 14:57:37 -0500 (Tue, 29 Sep 2009) | 16 lines Merged revisions 220873 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220873 | tilghman | 2009-09-29 12:59:26 -0500 (Tue, 29 Sep 2009) | 9 lines Reduce CPU usage related to building a peer merely for devicestates. This fixes a 100% CPU problem in the SIP driver, found by profiling the driver while the problem was occurring. (closes issue #14309) Reported by: pkempgen Patches: 20090924__issue14309.diff.txt uploaded by tilghman (license 14) Tested by: pkempgen, vrban ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@221027 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Jeff Peeler
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024ee4a43e |
Merged revisions 220718 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r220718 | jpeeler | 2009-09-28 14:10:10 -0500 (Mon, 28 Sep 2009) | 10 lines Fix building of registration entry in build_peer when using callbackextension Check for remotesecret option was unintentionally always true, which therefore caused the secret option to never be used. Thanks to dvossel for pointing out the exact fix. (closes issue #15943) Reported by: tpsast ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@220722 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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22a376de74 |
Merged revisions 219451 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r219451 | dvossel | 2009-09-18 11:20:41 -0500 (Fri, 18 Sep 2009) | 20 lines Merged revisions 219450 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219450 | dvossel | 2009-09-18 11:19:15 -0500 (Fri, 18 Sep 2009) | 14 lines via-header branches not updated correctly on INVITE INVITE requests must always contain a new unique branch id. When a new branch id is created for an INVITE, the dialog's invite_branch variable must be updated so CANCEL requests use the correct branch id. (closes issue #15262) Reported by: maniax Patches: asterisk-1.6.1.0-sip-branch.patch uploaded by tweety (license 608) invite_new_branch_trunk.diff uploaded by dvossel (license 671) Tested by: maniax, dvossel ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@219452 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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f38752b27c |
Merged revisions 219371 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r219371 | dvossel | 2009-09-17 17:37:28 -0500 (Thu, 17 Sep 2009) | 9 lines fixes deadlock when performing directed pickup w Invite/replaces (closes issue #15340) Reported by: lmsteffan Patches: deadlock.patch uploaded by lmsteffan (license 779) Tested by: lmsteffan ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@219376 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Joshua Colp
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1a5d50e555 |
Merged revisions 219324 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r219324 | mmichelson | 2009-09-17 17:22:01 -0500 (Thu, 17 Sep 2009) | 12 lines Merged revisions 219320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219320 | mmichelson | 2009-09-17 17:20:50 -0500 (Thu, 17 Sep 2009) | 6 lines Send a 100 Trying response when we detect a spiral. This was problematic during spiral tests at SIPit... along with some other things as well. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@219370 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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a2d5296ff7 |
Merged revisions 219304 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r219304 | dvossel | 2009-09-17 16:59:21 -0500 (Thu, 17 Sep 2009) | 27 lines Merged revisions 219303 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219303 | dvossel | 2009-09-17 16:29:37 -0500 (Thu, 17 Sep 2009) | 21 lines INVITE w/Replaces deadlock fix This patch cleans up the locking logic in chan_sip.c's handle_invite_replaces() function as well as making use of ast_do_masquerade() rather than forcing the masquerade on an ast_read(). The code had several redundant unlocks that would result in 'freed more times than we've locked!' errors. I cleaned these up as well as moving all the unlock logic to the end of the function. This patch should also resolve the issue people were having with the replacecall channel never being unlocked with one legged calls. (closes issue #15151) Reported by: irroot Patches: invite_w_replaces_1.4.diff uploaded by dvossel (license 671) Tested by: irroot, dvossel Review: https://reviewboard.asterisk.org/r/371/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@219307 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Joshua Colp
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34b56a06d4 |
Merged revisions 219264 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r219264 | file | 2009-09-17 14:57:39 -0500 (Thu, 17 Sep 2009) | 2 lines Ensure no spaces exist before "refresher=" when doing the comparison. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@219267 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Mark Michelson
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c76a7aa074 |
Merged revisions 218933 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r218933 | mmichelson | 2009-09-16 14:25:36 -0500 (Wed, 16 Sep 2009) | 12 lines Reverse order of args to fread. This way, we don't always write a null byte into byte 1 of the buffer (closes issue #15905) Reported by: ebroad Patches: freadfix.patch uploaded by ebroad (license 878) Tested by: ebroad ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@218937 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Joshua Colp
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41ee5cdf32 |
Merged revisions 218918 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r218918 | file | 2009-09-16 13:31:47 -0500 (Wed, 16 Sep 2009) | 5 lines On TCP and TLS connections do not attempt to stop retransmission of the packet internally. This was preventing responses from being properly processed because the packet was not being found causing handle_response to return prematurely. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@218934 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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ea227c48f9 |
Merged revisions 218687 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r218687 | dvossel | 2009-09-15 14:22:37 -0500 (Tue, 15 Sep 2009) | 2 lines upward bound checking for port string to int conversion ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@218688 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Matthew Nicholson
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8ac3a41308 |
Merged revisions 218586 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r218586 | mnicholson | 2009-09-15 11:15:02 -0500 (Tue, 15 Sep 2009) | 15 lines Merged revisions 218578 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218578 | mnicholson | 2009-09-15 11:03:54 -0500 (Tue, 15 Sep 2009) | 8 lines Send request contact header field with response to registrer queries instead of the address of record. (closes issue #14438) Reported by: ravindrad Patches: regquerypatch uploaded by ravindrad (license 684) Tested by: ravindrad ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@218590 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Mark Michelson
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820404f666 |
Merged revisions 218566 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r218566 | mmichelson | 2009-09-15 10:40:14 -0500 (Tue, 15 Sep 2009) | 4 lines Use a better method of ensuring null-termination of the buffer while reading the SDP when using TCP. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@218575 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Mark Michelson
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e25450ee36 |
Merged revisions 218499,218504 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r218499 | mmichelson | 2009-09-15 09:59:50 -0500 (Tue, 15 Sep 2009) | 3 lines Fix off-by-one error when reading SDP sent over TCP. ........ r218504 | mmichelson | 2009-09-15 10:05:53 -0500 (Tue, 15 Sep 2009) | 3 lines Ensure that SDP read from TCP socket is null-terminated. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@218507 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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fb1dd88b2f |
Merged revisions 217916 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r217916 | tilghman | 2009-09-10 18:12:16 -0500 (Thu, 10 Sep 2009) | 2 lines Make calltoken support work with realtime users and peers. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@217926 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Olle Johansson
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9aceafa424 |
Merged revisions 216805 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r216805 | oej | 2009-09-07 18:08:08 +0200 (MÃ¥n, 07 Sep 2009) | 2 lines Since it's possible to have more than 999 calls, I'm changing the call counter roof to something higher. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@217665 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Olle Johansson
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774015c13e |
Merged revisions 217593 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r217593 | oej | 2009-09-10 14:06:55 +0200 (Tor, 10 Sep 2009) | 8 lines Include ActionID in all events that are responsed to AMI Action SIPShowRegistry (closes issue #15868) Reported by: nic_bellamy Patches: manager_SIPshowregistry_actionid.patch uploaded by nic bellamy (license 299) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@217594 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Olle Johansson
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d8d48f385a |
Merged revisions 217368 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r217368 | oej | 2009-09-09 12:39:43 +0200 (Ons, 09 Sep 2009) | 2 lines Not having any TLS session to write to is a serious XMIT_ERROR. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@217369 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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e03fb9bf0c |
Merged revisions 216993 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r216993 | dvossel | 2009-09-08 09:26:30 -0500 (Tue, 08 Sep 2009) | 14 lines caller id number empty parse_uri was not being given the correct scheme's, as a result, uri parsing did not parse the username correctly. One of the side effects of this is an empty caller id. (closes issue #15839) Reported by: ebroad Patches: blank_cidv2.patch uploaded by ebroad (license 878) parse_uri_fix.diff uploaded by dvossel (license 671) Tested by: ebroad, dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@216994 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Olle Johansson
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0ff4f64221 |
Merged revisions 216842 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r216842 | oej | 2009-09-07 18:35:12 +0200 (MÃ¥n, 07 Sep 2009) | 2 lines Make sure we reset global_exclude_static at channel reload ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@216845 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Olle Johansson
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c558138003 |
Merged revisions 216695 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r216695 | oej | 2009-09-07 15:06:19 +0200 (MÃ¥n, 07 Sep 2009) | 8 lines If there is no session timer in the INVITE, set it to default value (not unset minimum = -1) Patch by oej closes issue #15621 Reported by: fnordian Tested by: atis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@216698 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Olle Johansson
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9e57fd58e2 |
Merged revisions 216438 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r216438 | oej | 2009-09-04 16:02:34 +0200 (Fre, 04 Sep 2009) | 35 lines Merged revisions 216430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 lines Make apps send PROGRESS control frame for early media and fix too early media issue in SIP The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI links *before* any call progress. The SIP channel receives these frames and by default signals 183 Session progress and starts sending media. This will cause phones to play silence and ignore the later 180 ringing message. A bad user experience. The fix is twofold: - We discovered that asterisk apps that support early media ("noanswer") did not send any PROGRESS frame to indicate early media. Fixed. - We introduce a setting in chan_sip so that users can disable any relay of media frames before the outbound channel actually indicates any sort of call progress. In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions of Asterisk, this will be enabled. We don't assume that it will change your Asterisk phone experience - only for the better. We encourage third-party application developers to make sure that if they have applications that wants to send early media, add a PROGRESS control frame transmission to make sure that all channel drivers actually will start sending early media. This has not been the default in Asterisk previous to this patch, so if you got inspiration from our code, you need to update accordingly. Sorry for the trouble and thanks for your support. This code has been running for a few months in a large scale installation (over 250 servers with PRI and/or BRI links to old PBX systems). That's no proof that this is an excellent patch, but, well, it's tested :-) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@216647 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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66b3a9f082 |
Merged revisions 216594 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r216594 | dvossel | 2009-09-04 14:32:07 -0500 (Fri, 04 Sep 2009) | 7 lines sip peer matching by address only with TCP/TLS This patch removes the contact header matching logic and adds logic to match all tcp/tls connections by ip only Review: https://reviewboard.asterisk.org/r/354/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@216598 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Russell Bryant
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2c7c09a97d |
Merged revisions 216368 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r216368 | russell | 2009-09-04 08:14:25 -0500 (Fri, 04 Sep 2009) | 12 lines Do not treat every SIP peer as if they were configured with insecure=port. There was a problem in the function responsible for doing peer matching by IP address and port number such that during the second pass for checking for a peer configured with insecure=port, it would end up treating every peer as if it had been configured that way. These changes fix the logic in the peer IP and port comparison callback to handle insecure=port checking properly. This problem was introduced when SIP peers were converted to astobj2. Many thanks to dvossel for noticing this while working on another peer matching issue. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@216436 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Olle Johansson
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d36fda5e78 |
Merged revisions 215891 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r215891 | oej | 2009-09-03 15:02:41 +0200 (Tor, 03 Sep 2009) | 10 lines Add known internal IP address when autodomain=yes (closes issue #14573) Reported by: pj Patches: sip-internip-autodomain1.diff uploaded by mnicholson (license 96) modified by oej Tested by: pj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@215929 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Olle Johansson
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c73b17515c |
Fix bad reports in "sip show channelstats".
Not directly mergeable in svn trunk, needs more tests, therefore committed directly to 1.6.2. (closes issue #15819) Reported by: klaus3000 Patches: asterisk-1.6.2-beta4-sipshowchannelstats-patch-0.2.txt uploaded by klaus3000 (license 65) Tested by: klaus3000, oej git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@215887 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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6859ea5646 |
Merged revisions 215801 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r215801 | tilghman | 2009-09-02 22:43:51 -0500 (Wed, 02 Sep 2009) | 5 lines Default the callback extension to "s". This is a regression. (closes issue #15764) Reported by: elguero Change-type: bugfix ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@215802 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Terry Wilson
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8613cf7fd9 |
Merged revisions 215758 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r215758 | twilson | 2009-09-02 18:31:04 -0500 (Wed, 02 Sep 2009) | 25 lines Merged revisions 215682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r215682 | twilson | 2009-09-02 16:41:22 -0500 (Wed, 02 Sep 2009) | 18 lines Re-send non-100 provisional responses to prevent cancellation From section 13.3.1.1 of RFC 3261: If the UAS desires an extended period of time to answer the INVITE, it will need to ask for an "extension" in order to prevent proxies from canceling the transaction. A proxy has the option of canceling a transaction when there is a gap of 3 minutes between responses in a transaction. To prevent cancellation, the UAS MUST send a non-100 provisional response at every minute, to handle the possibility of lost provisional responses. (closes issue #11157) Reported by: rjain Tested by: twilson Review: https://reviewboard.asterisk.org/r/315/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@215795 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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b469069e23 |
Merged revisions 215681 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r215681 | dvossel | 2009-09-02 16:39:31 -0500 (Wed, 02 Sep 2009) | 10 lines port string to int conversion using sscanf There are several instances where a port is parsed from a uri or some other source and converted to an int value using atoi(), if for some reason the port string is empty, then a standard port is used. This logic is used over and over, so I created a function to handle it in a safer way using sscanf(). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@215683 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Michiel van Baak
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1ce9e5b41d |
Merged revisions 215665 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r215665 | mvanbaak | 2009-09-02 23:23:17 +0200 (Wed, 02 Sep 2009) | 5 lines add Parkinglot info to sip show peer <foo> and skinny show line <foo> If we had this from the start, debugging the 'parking not using configured parkinglot' bug would have been easier. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@215680 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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2d59bd4874 |
Merged revisions 215222 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r215222 | tilghman | 2009-09-01 16:19:40 -0500 (Tue, 01 Sep 2009) | 3 lines Fix register such that lines with a transport string, but without an authuser, parse correctly. (AST-228) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@215569 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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83cb4b97b7 |
Merged revisions 215522 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r215522 | dvossel | 2009-09-02 12:26:40 -0500 (Wed, 02 Sep 2009) | 11 lines SIP uri parsing cleanup Now, the scheme passed to parse_uri can either be a single scheme, or a list of schemes ',' delimited. This gets rid of the whole problem of having to create two buffers and calling parse_uri twice to check for separate schemes. Review: https://reviewboard.asterisk.org/r/343/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@215523 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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d7c8695c3b |
Merged revisions 215466 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r215466 | dvossel | 2009-09-02 11:08:00 -0500 (Wed, 02 Sep 2009) | 16 lines SIP support for keep-alive event keep-alive events are used by Sipura/Linksys for NAT keepalive. There currently don't appear to be any problems with NAT, but everytime a keep-alive event is received, Asterisk responds with a "489 Bad event". This error may indicate to a user that NAT problems exist just because this even is not supported. Now, rather than respond with an error, the packet is consumed and a "200 ok" is sent just to indicate we received the packet. (issue #15084) Patches: chan_sip.keepalive.v1.diff uploaded by IgorG (license 20) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@215467 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Michiel van Baak
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91694d5a9d |
Merged revisions 215462 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r215462 | mvanbaak | 2009-09-02 17:56:46 +0200 (Wed, 02 Sep 2009) | 12 lines Honor configured parkinglot when parking and retrieving parked calls Thank oej for pointing out the fact that sip_new did not copy parkinglot from the peer into the newly created channel. (closes issue #15538) Reported by: gracedman Patches: 2009090100_sipnewparkinglot-161.diff.txt uploaded by mvanbaak (license 7) With mod by me to also fix callparking as well (this uploaded patch only fixed retrieving a parked call) Tested by: gracedman, mvanbaak ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@215465 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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57f3a15238 |
Merged revisions 214199 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r214199 | tilghman | 2009-08-26 11:53:03 -0500 (Wed, 26 Aug 2009) | 6 lines Typo fix ("SIP/2.0 XXX" is 11 chars, not 10) (closes issue #15362) Reported by: klaus3000 Patches: chan_sip.c_logmessagefix_patch.txt uploaded by klaus3000 (license 65) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@214202 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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6acb97f2d5 |
Merged revisions 213738 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r213738 | tilghman | 2009-08-21 17:36:39 -0500 (Fri, 21 Aug 2009) | 2 lines Clarifying comments in sip_register, and removing a dead section ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@213739 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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836d4bcdb5 |
Merged revisions 213716 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r213716 | dvossel | 2009-08-21 17:22:11 -0500 (Fri, 21 Aug 2009) | 10 lines Register request line contains wrong address when user domain and register host differ (closes issue #15539) Reported by: Nick_Lewis Patches: chan_sip.c-registraraddr.patch uploaded by Nick (license 657) register_domain_fix_1.6.2 uploaded by dvossel (license 671) Tested by: Nick_Lewis, dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@213721 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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a65e6a52a1 |
Merged revisions 213635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r213635 | dvossel | 2009-08-21 16:02:50 -0500 (Fri, 21 Aug 2009) | 5 lines fixes sip register parsing when user@domain is used (issue #15008) (issue #15672) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@213636 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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8afb651407 |
Merged revisions 213098 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r213098 | tilghman | 2009-08-19 16:05:17 -0500 (Wed, 19 Aug 2009) | 9 lines Better parsing for the "register" line Allows characters that are otherwise used as delimiters to be used within certain fields (like the secret). (closes issue #15008, closes issue #15672) Reported by: tilghman Patches: 20090818__issue15008.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen, tilghman ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@213117 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
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3f1b6567bf |
Merged revisions 213093 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r213093 | tilghman | 2009-08-19 15:29:41 -0500 (Wed, 19 Aug 2009) | 7 lines If we have realtime caching enabled, 'sip reload' must purge users/peers, even if the config files haven't changed. (closes issue #12869) Reported by: bcnit Patches: 20090819__issue12869__2.diff.txt uploaded by tilghman (license 14) Tested by: lasko ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@213095 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |