[OpenSIPS-Users] Integration with Asterisk/Trixbox

[Iñaki Baz Castillo]

2009/5/20 James Lamanna <jlamanna at gmail.com>:
> 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?

Set "canreinvite=no" for opensips peer in sip.conf.


> The ones I've found on the web are useful in some ways, but not in others.

This question is more related to Asterisk.


-- 
Iñaki Baz Castillo
<ibc at aliax.net>