[OpenSIPS-Users] dialplan module

[Francjos]

I read the tutorial but it seems not to be approrpiate to my environnement.
In the enterprise where i do my training, they wish to put in palce two
Opensips servers and two Freeswitch servers. The Opensips serves must
register to their sip provider via a sip trunk (the registration is already
done). The opensips servers act as sip proxy and route sip signaling to
FreeSwitch servers, Freeswitch servers handle media.
There is no sip phones that  will be connected to the servers, when a call
comes through their sip trunk, the call is handled according to an IVR menu,
there is no person to answer the incoming call.

So i need clarification on the way i have to proced , step by step.
- For example, How and where the number (telephone line) corresponding to
the sip trunk is configured? In opensips or in freeswitch server?

Thanks a lot




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