[Asterisk-Users] DID on SIP channel

[Joshua Colp]

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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
denis at isolve.com.br
Sent: Tuesday, June 07, 2005 7:12 PM
To: asterisk-users at 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.
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