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

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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.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53110 65c4cc65-6c06-0410-ace0-fbb531ad65f3