Files
asterisk/channels
Olle Johansson cfe66e6b26 Patch based on this patch with small changes for trunk...
Merged revisions 53109 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r53109 | oej | 2007-02-02 01:24:03 +0100 (Fri, 02 Feb 2007) | 4 lines

Disable the direct p2p RTP call setup in SIP. You can enable it in sip.conf, but it is now
considered experimental until we solve the AST_CONTROL_ANSWER with payload and videocaps
stuff.

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53110 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-02 00:26:25 +00:00
..
2007-01-16 08:38:59 +00:00
2007-01-16 08:38:59 +00:00
2007-01-16 08:38:59 +00:00