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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Denis Galv�o - 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 Galv�o. 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: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > [EMAIL PROTECTED] > Sent: Tuesday, June 07, 2005 7:12 PM > To: [email protected] > 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 > [email protected] > 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 > [email protected] > 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 [email protected] 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 [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
