[OpenSIPS-Users] Integration with Asterisk/Trixbox

[James Lamanna]

Hi,
I want to use OpenSIPs as the registrar (and NAT handler) for an
Asterisk/Trixbox installation.
I've got things partially working, but I've totally made a mess of my
config (I can post it if you would like).

Some things that I need:

I'm having problems with SIP<->SIP calls because I need asterisk to
stay in the media stream, so really the call has to be routed like:

phone1 <--> opensips <--> asterisk <--> opensips <--> phone2.

Does anyone have any configs that come close to this that I could stare at?
The ones I've found on the web are useful in some ways, but not in others.

Thanks.

-- James