I read the tutorial but it seems not to be approrpiate to my environnement. In the enterprise where i do my training, they wish to put in palce two Opensips servers and two Freeswitch servers. The Opensips serves must register to their sip provider via a sip trunk (the registration is already done). The opensips servers act as sip proxy and route sip signaling to FreeSwitch servers, Freeswitch servers handle media. There is no sip phones that will be connected to the servers, when a call comes through their sip trunk, the call is handled according to an IVR menu, there is no person to answer the incoming call.
So i need clarification on the way i have to proced , step by step. - For example, How and where the number (telephone line) corresponding to the sip trunk is configured? In opensips or in freeswitch server? Thanks a lot -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/dialplan-module-tp7602459p7602507.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
