>You're missing the point.  If a call comes in from a PSTN (POTS) line 
>into a PSTN<-->SIP gateway, the PSTN line MAY provide Calling Party ID, 
>if that functionality is enabled at the exchange, but it WILL NOT 
>provide Called Party ID as there is no standard for providing such 
>information through a standard POTS line. 

I dont know what gateway do you use. For example with audiocodes you can
manipulate Called party ID.
The same with Patton. If your gateway cant do that then all calls from that
gateway will be routed 
in the same way by SipX.    

In any case the "TO:" field should be populated by the gateway...maybe it is
something like 
"TO: [email protected]". Then your prefix to manage all calls from that
gateway will be "unknown". 


Heros


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