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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nick Kartsioukas Sent: Friday, July 15, 2005 12:22 AM To: [email protected] 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: [EMAIL PROTECTED] The closest I can see to do this with the Dial() command is: Dial(SIP/[EMAIL PROTECTED]) but I'm not sure if that will even parse correctly... So: exten => _X,1,Dial(SIP/[EMAIL PROTECTED]) 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 [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
