[Asterisk-Users] What route do diverted SIP calls travel?

[Sam Bashton]

On Mon, 13 Dec 2004 11:40:46 +0000, Andy Burns
<digiumasterisk at adslpipe.co.uk> wrote:
> If I have inbound SIP calls arriving from a provider's gateway to an
> asterisk server on my LAN, which then routes the call back out via the
> provider's gateway to a PSTN number, once the call is answered do all
> the voice packets pass through my asterisk PBX, or is SIP intelligent
> enough to patch the two PSTN ends of the call direct to each other going
> only via two ports on the provider's gateway?

The data-heavy portion of the traffic is RTP, and that should be a
direct connection using your providers gateway.  Make sure you have
'canreinvite=yes' set in the appropriate section of your sip.conf.

-- 
Sam Bashton