2009/5/20 James Lamanna <[email protected]>: > 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 <[email protected]> _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
