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
> 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