Hi Ahmed, Ahmed Munir wrote: > Hi, > > Current I'm working on OpenSIPS + FreeRadius, where FreeRadius is for > AAA. Accounting and Authentication are working well i.e. SIP phones > get authenticated and can make calls between them. > I want to know how can I distinguish calls, the flow is listed down below; > > User A's number is 1234 and User B's number is 1235. > Both users' phone registered on UAS (OpenSIPS+FreeRadius), can make > SIP-SIP (on-net) calls. > If User A do not registered his number, he can make call to User B > where User B is registered on UAS like PSTN-SIP call. > If User A is registered on UAS and make a call to User B who is not > registered on UAS but located on PSTN, SIP-PSTN.
Are A and B SIP users ? how comes that if B is not registered, he gets on PSTN ? Regards, Bogdan > > > In summary I need to know how can I configure SIP-SIP, SIP-PSTN and > PSTN-SIP peers and how can I distribute their routes? Further added, > which modules, modparam and function requires for it? > > -- > Regards, > > Ahmed Munir > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Bogdan-Andrei Iancu www.voice-system.ro _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
