So you want asterisk not to proxy the media.. Simple. Put canreinvite=yes in your peer in sip.conf. It will do it automatically.
On Fri, 2006-02-17 at 01:19 +1100, Zafer Khodr wrote: > I am trying to setup asterisk as shown below. > > > > xxx.xxx.xxx.xxx > /=====(Call Origination)====\ > / | \ > R / |S \ > T / |I \ R > P / |P \ T > / | \ P > S / Asterisk \ > T / /\ \ S > R / / \ \ T > E / / \ \ R > A / / \ \ E > M / / \ \ A > / / \ \ M > / / \ \ > / / \ \ > / __________/ \__________ \ > / | | \ > / | | \ > yyy.yyy.yyy.yyy zzz.zzz.zzz.zzz > Terminating Gateway 1 Terminating Gateway 2 > > > > The main objective to to have asterisk in the path of the call but the RTP > to go directly between the originating and terminating IP's. > I have had a play around with canreinvite but doesn't seem to make a > difference. > > If someone could please help me out by pointing me in the right direction, > that would be great > > Regards, > Zafer > > > > > > > > > > > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-biz mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-biz _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
