> 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 _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
