asterisk users - Dec 2004 - What route do diverted SIP calls travel?

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?

On Mon, 13 Dec 2004 11:40:46 +0000, Andy Burns
<digiumasterisk@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