[cisco-voip] Setting up 2610 as a SIP gateway to PBX

Parent, André Andre.Parent at banq.qc.ca
Fri Jan 30 14:13:12 EST 2009


On the SIP side you have to configure a sip user agent in the gateway in order to redirect what's coming from your sip trunk to the T1. Here's an example :

SIP use agent :
sip-ua
 sip-server ipv4:IPADRESSOFASTERISK

If you want to direct calls the other way, let's say all extensions in the range of 6XXX coming from your T1, you create a VOIP dialpeer 

The dial-peer :
!
dial-peer voice 8503 voip
 preference 1
 destination-pattern 6...
 monitor probe icmp-ping
 session protocol sipv2
 session target ipv4:IPADRESSOFASTERISK
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad

and create a POTS dialpeer for what will come from the T1.
 
The above examples work on a 2811 and Asterisk.



André Parent
Bibliothèque et Archives nationale du Québec
Téléphone : (514) 873-1101 poste 3363
andre.parent at banq.qc.ca


-----Message d'origine-----
De : cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] De la part de x25 at pobox.com
Envoyé : 29 janvier 2009 14:52
À : cisco-voip at puck.nether.net
Objet : [cisco-voip] Setting up 2610 as a SIP gateway to PBX

I've been given a task which I am not familiar with, so I need some help.

Basically, the idea is to connect 2610 router with a PBX, over T1 PRI, and then allow other devices 
(Asterisk, for example - or even an ordinary SIP softphone) to make calls through PBX via Cisco gateway.

Something like this:


Asterisk ----- SIP -----> 2610 -----> T1 PRI -----> Mitel 3300 PBX


I have configured 2610 and brought T1 link up. Hoever, I am bit lost as to how I need to configure router in 
order to allow SIP clients talk to PBX. SIP is IP based protocol, so I presume 2610 will do some voodoo 
conversion before sending it to PBX. I do have username/password for PBX, just need to find a way to register 
a SIP client over this link (can't talk SIP directly to 3300).

So, I have controller and serial interface setup:

isdn switch-type primary-dms100

controller T1 1/0
 framing esf
 clock source line primary
 linecode b8zs
 cablelength short 133
 pri-group timeslots 1-24
 description Link to Mitel 3300

interface Serial1/0:23
 no ip address
 encapsulation hdlc
 isdn switch-type primary-dms100
 isdn incoming-voice voice
 no cdp enable

And at this point, I am bit lost. After trying to read lots of documentation, I've reached the "information 
overload" phase, where I am more confused than before, so I need help :)

I assume that I also need dial-peer defined, like this (this was just for testing)

dial-peer voice 1023 pots
 destination-pattern .T
 port 1/0:23
 forward-digits all

Could anyone just explain basic steps (and which features are needed) in order to make this work?

I tried adding random things to configuration (sip-ua enabled, then enabling voice service voip, etc), but 
the main problem is that I don't understand what I need exactly, so any help is welcome :)

Thanks.
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