CS1000e SIP Trunking

[CSSMT]

Hi All,

We have a Nortel CS1000e installation. Now we need to integrate this with three sites running on Welltech SIP PBX 6200 and one site running on cisco call manager 4.1. somehow i have managed to bring all these four into my CS, i.e. the users of these PBX's are able to call my Nortel phones. But I am not able to make calls to any of these PBX.

This is what I have done so far.

1 Created D-Channel
2 Created 32 SIP Virtual Trunks
3 Created a SIP Route (route 1)
4 Created RLB (RLI 1)
5 Created CDP - DSC 60 (for cisco i am using 60)and FLEN 6 (as the dial plan is 4 digit for Cisco.
6 COnfigured NRS
a L0,L1, Gateway End point and Routing enteries (gave DN prefix 60)

please someone help and suggest what i am missing.

Regards
Goutam..

 

[Arriero]

CSSMT,

10.0.32.12 is your SS but at the same time is your NRS because you have both applications (SIP Gateway and NRS) in the same server

A normal call should have the following (summarized) process:

1. SIP GW sends an INVITE message to NRS (If NRS and SIP GW are in the same serverm you will see the SS IP Address here)

2. NRS responds 302 temporarily moved and provides the IP address of the endpoint returned from the NRS routing entry database (in your case 10.1.2.56)

3. SIP GW sends and invite to the IP address received from the NRS.


In here you see a 404 not found message, which means that the NRS doesn't find 2200 in its routing entries.

Have you tried the SIP Routing test tool in the NRS? I am assuming somehow your routing entries are not properly configured in the NRS database.

BUT....to be honest now I am totally confused cause in your previous posts you mention that you were dialing 601008 but the trace shows that the NRS is receiving 2220, in other words you are dialing 2220??? Are you doing some pre-translation?? Can you clarify this please!!