[OpenSIPS-Users] Integration with Asterisk/Trixbox

Iñaki Baz Castillo ibc at aliax.net
Wed May 20 16:46:36 CEST 2009

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>

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