[Asterisk-Users] SIP/iax routing question

[Nabeel Jafferali]

> 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