asterisk users - May 2005 - 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

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: 'asterisk-users@lists.digium.com'
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