Hi, I want to implement call forking in a scenario where multiple UAs (using Linphone) register with the same SIP URI (say "[email protected]"). Going by the SIP protocol, each of these UAs shall use their own IP address in the Contact header while sending the REGISTER request.
I'm using openSIPS 1.5.2 (no-tls) for the proxy/registrar functions. I'm successfully able to register multiple UAs with same SIP URI ([email protected]). Problem comes when i make a call from some other UA ([email protected], also registered with the same openSIPS proxy) and route the call via proxy. Note that the outgoing INVITE does not have a Route header (i'm using Linphone for all UAs). On receiving the call, openSIPS fails to route the message. It is unable to resolve "abc.com" domain. Is there something that i'm missing in openSIPS configuration. I thought if the registration of the UA is successful, openSIPS would simply dip into its registration DB, get the contact details and route the msg forward. Perhaps, it is trying to do a DNS SRV/NAPTR lookup for "abc.com" and eventually fails. Looks like i need to define "abc.com" as the local domain somewhere in openSIPS cfg. Any help in this regard would be appreciated. Here is a pictorial view of what is to be achieved: Registration ------------------ UA1 ([email protected], 192.168.5.14) -----> openSIPS UA2 ([email protected], 192.168.5.38) -----> openSIPS UA3 ([email protected], 192.168.5.25) -----> openSIPS Call Forking ----------------- | ----------> UA1 UA3 (calls [email protected]) ---> openSIPS (forks to UA1, UA2) --- | |-----------> UA2 /csj
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