If I understand it correctly, SIP just handles the signalling between endpoints. When I call someone via a sip proxy, once the connection is made all the audio is going directly from me to the person I am calling correct? What happens if a SIP call is routed through more than one * server?
Also, when setting up an inter asterisk exchange, is all the data routed through the * servers? Chris _______________________________________________ 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
