You can make RTP pass through Asterisk, or not. Look in voip-info.org about "Native Bridge" and "sip.conf" "canreinvite" option. And may be this page will be usefull too:
http://www.voip-info.org/wiki/index.php?page=Asterisk+SIP+not-proxy Regards On 10/31/06, Mike Williams <[EMAIL PROTECTED]> wrote:
Hey, This is probably a rather stilly question... If I pick up my SIP phone that's registered to my asterisk server and dial a number that asterisk recognises as destined for a SIP trunk (could be a static route, or an ENUM lookup) or another SIP device registered on said asterisk server (internal extension to extension call), what route does the actual audio take? The control connection (port 5060) obviously goes via the asterisk server as it has to work out where to send the control to, but I could quite easily imagine the audio going directly handset to remote server or handset to asterisk to remote, and handset to handset or handset to asterisk to handset. Thanks -- Mike Williams _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
-- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org"; _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
