]> git.ipfire.org Git - thirdparty/asterisk.git/commit
Merged revisions 318331 via svnmerge from
authorTerry Wilson <twilson@digium.com>
Mon, 9 May 2011 20:23:15 +0000 (20:23 +0000)
committerTerry Wilson <twilson@digium.com>
Mon, 9 May 2011 20:23:15 +0000 (20:23 +0000)
commitf96cf8821239051b90d47649c1af9f2f914bb887
treed0b2a46fa94c41a9fcda45ed7c3b6e7185026cfc
parent607164ad91e65963f53cb7df71622671e489f98d
Merged revisions 318331 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2

........
  r318331 | twilson | 2011-05-09 15:04:41 -0500 (Mon, 09 May 2011) | 12 lines

  Don't offer video to directmedia callee unless caller offered it as well

  Make sure that when directmedia is enabled, that video is not offered to the
  callee even if it supports it. p->vrtp will not exist since the caller didn't
  offer video.

  (closes issue #19195)
  Reported by: one47
  Patches:
        sip_cant_add_video_rtp uploaded by one47 (license 23)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@318337 65c4cc65-6c06-0410-ace0-fbb531ad65f3
channels/chan_sip.c