Ok, thank you very much. I will try to find any information in asterisk documentation about RTP.
On Fri, Nov 13, 2009 at 3:03 PM, John A. Sullivan III <[email protected]> wrote: > On Fri, 2009-11-13 at 11:44 +0100, Ignacio wrote: >> I have just established a call between 2 sip phones and I have noticed >> that all RTP traffic goes through Asterisk Server. >> >> I was expecting RTP traffic went to one phone to another phone directly. >> >> I set canreinvite=yes in sip.conf in both sip peers. >> >> I also tested it with 2 mgcp phones and same result, all rtp traffic >> goes through Asterisk. >> >> Is there any way to force traffic to go from one phone to another? > <snip> > I don't recall where it is off-hand but, somewhere in the Asterisk > documentation, there is an explanation of how Asterisk makes a decision > about reinvites. You may want to look at that to see if your > environment satisfies all the requirements and how it can be adapted if > it does not - John > -- > John A. Sullivan III > Open Source Development Corporation > +1 207-985-7880 > [email protected] > > http://www.spiritualoutreach.com > Making Christianity intelligible to secular society > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
