[Asterisk-Users] Calls from PSTN -> SIP

[Anton Yurchenko]

Hello,

I have a Sip Express Router (www.iptel.org) setup as a registrar and a 
SIP proxy for all the calls between the local IP-phones(Snom100). And I 
can forward the call from those phones to the * fine. But how do I set 
up so that calls coming from PSTN Zap channels go to spesific SIP 
phones, without putting all of them in the sip.conf and extensions.conf?
How to forward them to the SER router? Or for example to group of 
phones? Is there a way to use and ${EXTEN} variable with SIP channnels?
I want the authentication features so that users would have to 
register/authorize on the SER proxy, becouse it`ll allow much richer 
usages ot the whole SIP features and allow for more scalability.
But that cancels out registering/authorization on asterisk.

-- 

Anton Yurchenko<phila at dg.net.ua>
Digital Generation