[Asterisk-Users] PSTN to SIP gateway

[Jose Raborg]

Nick:

Do you want to route the calls depending on the caller id? Or Do you
want to assign a DID to a SIP?
JR

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Nick
Kartsioukas
Sent: Friday, July 15, 2005 12:22 AM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] PSTN to SIP gateway


I've been looking through the examples and docs, but haven't yet quite
figured out how to do what I want. We've got a T1 coming in carrying a
block of telephone numbers, terminating on an Asterisk box.  Any call
that comes in needs to get sent to a SIP proxy, with a particular
extension format:
	*ANI*DNIS*@sipproxy.address
The closest I can see to do this with the Dial() command is:
	Dial(SIP/*$CALLERIDNUM*$DNID*@sipproxy.address)
but I'm not sure if that will even parse correctly...

So:
	exten => _X,1,Dial(SIP/*$CALLERIDNUM*$DNID*@sipproxy.address)
is what I think I need in my extensions.conf in order to catch all
incoming numbers and initiate a SIP connection for them.

Please have mercy on me, I've been perusing docs all day, and it's
entirely possible I'm just trying to absorb too much too fast and am
missing something obvious :)  Thanks to any who can help!

-- 
Nick Kartsioukas
Sky Way Networks, LLC _______________________________________________
Asterisk-Users mailing list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users