[Freeswitch-users] SIP incoming call routing
talk2ram at gmail.com
Wed Oct 29 02:42:28 PDT 2008
On Wed, Oct 29, 2008 at 2:52 PM, Saurabh Aggarwal <
saurabh_aggarwal at hotmail.com> wrote:
> 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?
have you looked at this example
-------------- next part --------------
An HTML attachment was scrubbed...
More information about the FreeSWITCH-users