[Freeswitch-users] SIP incoming call routing

[Saurabh Aggarwal]

We are using freeswitch as a SIP proxy, where we are letting people register with freeswitch, and in-turn we do the SIP registration for them to "arbitrary" sip servers (as requested by users) - each user gets his own sip gateway in the freeswitch configuration. Then they can make outgoing calls and calls are routed through their specific SIP gateway.
 
Now the problem is that when a call is received from one of these SIP registrations, it hits the public.xml where I can't seem to figure out how to get the SIP gateway information from which it came in. The SIP gateway name actually contains the information where it should be routed to. Any ideas on how to approach this problem?
 
Question - is it possible to do it in the dialplan (dynamic) or do we have to write an application to do this mapping?
 
-Saurabh
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