[OpenSIPS-Users] Call Forking using openSIPS

[Code Warrior]

Hi,

I want to implement call forking in a scenario where multiple UAs (using
Linphone) register
with the same SIP URI (say "user1 at abc.com"). 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 (user1 at abc.com). Problem comes when i make a
call from some
other UA (user2 at abc.com, 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 (user1 at abc.com, 192.168.5.14) -----> openSIPS
UA2 (user1 at abc.com, 192.168.5.38) -----> openSIPS
UA3 (user2 at abc.com, 192.168.5.25) -----> openSIPS

Call Forking
-----------------

| ----------> UA1
UA3 (calls user1 at abc.com)  ---> openSIPS (forks to UA1, UA2) ---  |

                      |-----------> UA2

/csj
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