[Asterisk-Users] DID on SIP channel

[Denis Galvão - iSolve]

Hi Joshua. Thanks for your reply. I will try to be more clear.

Imagine this environment:

Extensions(2222,2223,2224,...) ---- Asterisk ---- Nortel MCS ---- PSTN

I have a sip channel configured in Asterisk with Nortel. I received 
just one user/passswd to register Asterisk on Nortel. This user is a 
real phone number(30302221) that can be reached from PSTN. With just 
one number/user I have no problem to route a PSTN call to the 
correspondent Asterisk extension.

The problem occur when Nortel is configured to have some alias(other 
phone numbers) with the same user/password, that I told before.

30302222 -+
30302223 -|
30302224 -|---user:30302221 passwd:secret
30302225 -|
30302226 -+

Just 30302221 is registered by Asterisk on Nortel. If I call one of 
this numbers I will reach Asterisk, so I want to configure a DID for 
each one of them to ring an especific extension:

Phone #      Extension #
30302222 ->  2222
30302223 ->  2223
30302224 ->  2224
30302225 ->  2225
30302226 ->  2226

P.S.: I received on Asterisk an INVITE with the phone number called.

Could someone help me!?

Can Asterisk handle this!?

Thanks.

Denis Galvão.


On 07 de jun de 2005, at 23:19, 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!?
>
> Denis.
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