]> git.ipfire.org Git - thirdparty/asterisk.git/commit
Merged revisions 165599 via svnmerge from
authorJoshua Colp <jcolp@digium.com>
Thu, 18 Dec 2008 17:14:27 +0000 (17:14 +0000)
committerJoshua Colp <jcolp@digium.com>
Thu, 18 Dec 2008 17:14:27 +0000 (17:14 +0000)
commit7333579bf97f6d8ea70b19f695e524ffd8dd9ce9
tree36d9bc00f9676d0a13c03fdbbd8ef1a9fc564688
parent4e3184c90838cd6551e3800f22198907639f9fbe
Merged revisions 165599 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r165599 | file | 2008-12-18 13:13:32 -0400 (Thu, 18 Dec 2008) | 11 lines

  Merged revisions 165591 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.4

  ........
    r165591 | file | 2008-12-18 13:11:42 -0400 (Thu, 18 Dec 2008) | 4 lines

    Only care about a compatible codec for early bridging if we are actually bridging to another channel. If we are not we actually want to bring the audio back to us.
    (closes issue #13545)
    Reported by: davidw
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@165603 65c4cc65-6c06-0410-ace0-fbb531ad65f3
main/rtp.c