You're actually confusing me when you say this due to the fact you're not giving much information, probably why nobody has responded yet. If the SIP server on the Nortel does an INVITE for the phone number, then asterisk will act accordingly and go to the phone number in the context you set for it. Note that if the Nortel is incapable of handling a challenge for credentials, you'll have to use a peer entry with insecure=very to match based on it's host/IP address.
- Joshua Colp. (file in #asterisk on Freenode) -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, June 07, 2005 7:12 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] DID on SIP channel Hi all. I need to implement the DID funcionality in a SIP channel with an ITSP. Is this possible to get it working!? The ITSP that im using has the "alias" feature in its SIP server(Nortel MCS5200), they provide just one register user/password and below this user they put a lot of other phone numbers. Ex.: register => 30302222 alias => 30302223 alias => 30302224 etc... Any clue for it!? Denis. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@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 _______________________________________________ Asterisk-Users mailing list Asterisk-Users@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