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 _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
