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

[Rich Adamson]

> > 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.

That assumes the provider is supporting reinvites. They may not.