[Asterisk-Users] SIP/iax routing question

[snacktime]

On Fri, 25 Mar 2005 04:07:13 -0500, Nabeel Jafferali
<nabeel at jafferali.net> 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.