Let me rephrase my previous answer.. canreinvite is the answer HOWEVER if there is a codec negotiation problem between the endpoints asterisk will take over and transcode the media. My only suggestion to you is to watch the debug of the invites and see what its doing.
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
