]> git.ipfire.org Git - thirdparty/asterisk.git/commit
Merged revisions 371338 via svnmerge from
authorAutomerge script <automerge@asterisk.org>
Thu, 16 Aug 2012 16:22:37 +0000 (16:22 +0000)
committerAutomerge script <automerge@asterisk.org>
Thu, 16 Aug 2012 16:22:37 +0000 (16:22 +0000)
commitda652a5e68c595259a228b4fd2f680489efc46dc
tree60ca639a4d4d3076ea692b54f60fcfd4f0266b5b
parent008377948044303fd27100fa7d8a4b122d2782a8
Merged revisions 371338 via svnmerge from
file:///srv/subversion/repos/asterisk/branches/10

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  r371338 | jrose | 2012-08-16 11:16:04 -0500 (Thu, 16 Aug 2012) | 14 lines

  chan_sip: Trigger reinvite if the SDP answer is included in the SIP ACK

  Under certain conditions, a SIP transaction involving directmedia wouldn't
  trigger a re-invite because the SDP answer was included in an ACK instead
  of in a message that we would have triggered the invite with. This patch
  just queues a source change control frame if the dialog is using
  directmedia when we find sdp for an ACK.

  (closes issue AST-913)
  Reported by: Thomas Arimont
  ........

  Merged revisions 371337 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@371354 65c4cc65-6c06-0410-ace0-fbb531ad65f3
channels/chan_sip.c