Er, let me elaborate a little bit :-) I understand that canreinvite is supposed to do this, all "peers" are set to canreinvite=yes.. All test boxes are on the same subnet, as well.
-----Original Message----- From: Matt Schulte Sent: Wednesday, May 11, 2005 2:43 PM To: '[email protected]' Subject: Forcing Asterisk to not bridge/transcode RTP traffic Does anyone know how to do this? Just curious, ie SIP callflow A -- Asterisk -- B, RTP goes directly from A to B .. Matt _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
