On Wednesday 08 December 2004 04:44, Gonzalo Gasca Meza wrote: > Hi all, > I have just setup Asterisk, but the problem is that all RTP stream pass > through Asterisk, I mean all call setup and voice stream pass trough > Asterisk once the call is established. Is there a way that call setup is > established, the RTP stream pass just between the SIP endpoints. > > > Example: > Works like this > SIP IP phones <-----------Asterisk RTP stream--------------> SIP IP phone > > > Asterisk > > SIP IP phones <------------------RTP------------------------> SIP IP phone > yes, unless you have canreinvite=no in your sip.conf, assuming that the phones negotiate the same codecs then they should be able to initiate a re-invite so the the stream goes peer to peer taking * out of the loop.
Jon _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
