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
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Bogdan-Andrei Iancu
www.voice-system.ro
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