[Asterisk-Users] DID on SIP channel

[Olle E. Johansson]

Joshua Colp wrote:
> 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!?
> 
I guess you are registering with the Nortel SIP server? All the incoming
calls will go to the incoming extension you are registering with them.
If they add aliases for several incoming lines to one registration, you
need to check the To: header. This is only possible in CVS head with the
SIP get header function in the dial plan.

This is one of the reasons I am planning to implement a type=service
object in sip.conf

/O