On Mon, Dec 08, 2003 at 10:20:43PM +0100, Brancaleoni Matteo wrote: > SIP control messages goes always through the server > (port 5060) , only RTP media streams is p2p . > > you can see RTP passing not p2p but by * server if: > * the phone doesn't supports reinvites > or > * set in sip.conf canreinvite=no in the user definition >
Or of course, if Asterisk thinks that it needs to process the stream : for instance, if you want Asterisk to be able to transfer your call (t/T options for Dial). -- Nicolas Bougues Axialys Interactive _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
