On Fri, 25 Mar 2005 04:07:13 -0500, Nabeel Jafferali <[EMAIL PROTECTED]> wrote: > > What happens if a SIP call is routed through more > > than one * server? > > If canreinvite=yes for all the peers involved, and t or T is not used in > the Dial command, then the audio would get routed directly between the > endpoints.
> > > Also, when setting up an inter asterisk exchange, is all the > > data routed through the * servers? > > As long as notransfer=no for all the peers involved, then everything but > the endpoints would completely drop out of the call. > > Nabeel > Thanks Nabeel, that's what I needed to know. _______________________________________________ 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
