[Asterisk-Users] DID on SIP channel

Tue Jun 7 20:13:41 MST 2005

I tried exactly what you wrote, but it didnt work out.

I dont know about the type=peer, maybe i've tested it with type=friend. 
Dont you need the user/password on [nortel] context!?

Thanks.

Denis.

On 07 de jun de 2005, at 23:50, Joshua Colp wrote:

> Okay lemme give you something that should work some magic!
>
> Stuff for sip.conf:
> [nortel]
> type=peer
> host=<IP ADDRESS OF NORTEL>
> disallow=all
> allow=ulaw
> context=inbound_nortel
> insecure=very
>
> Stuff for extensions.conf:
> [inbound_nortel]
> exten => 30302222,1,Dial(SIP/whatever)
> exten => 30302223,1,Dial(SIP/bleh)
> ... SO ON...
>
> Use your head to figure out some of the stuff for what you should put 
> in.
>
> - 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 
> Galvão -
> iSolve
> Sent: Tuesday, June 07, 2005 11:48 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] DID on SIP channel
>
> 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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>
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