Integrating Nortel CS1000 & Rauland Responder 5 via Brek

1. Brekeke Product Name and version: SIP Server - version 2.4.4.8/286

2. Java version:

3. OS type and the version: XP

4. UA (phone), gateway or other hardware/software involved: Nortel CS1000

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html : 1

6. Your problem:
Looking for help on connecting a Nortel CS1000 to the Brekeke SIP Server along with a Rauland Responder 5 system.
I'm able to get a phone on the Nortel PBX to talk to an X-Lite SIP softphone connected to the Brekeke - by configuring a domain on the Brekeke that is the same as the Nortel CS1000 - and putting the soft phone in that domain.
But the Rauland doesn't appear to use a domain (or haven't seen it can) so the endpoints register to the Brekeke without a domain. How do I get these 2 to talk?
The endpoints associated with the Rauland seem to be able to call the Nortel endpoints - cause they just follow the dial plan rules. But the reverse (Nortel calling Rauland devices) seem to come into the Brekeke with the domain information associated with the Nortel - which of course the Brekeke doesn't know what to do with, so it responds with a 404 Not Found.
Ideas or suggestions?

if not use domain on brekeke, can you make calls between rauland and nortel on both direction?
what dial plans created on brekeke?

Hi Jeff,

Are you a Rauland-Borg's reseller?
If so, you can get direct support from Brekeke's support team.

Please contact support@brekeke.com .

which edition of SIP Server are you using?

paste your current Dialplan rules here.

I am using version: 2.4.4.8/286

Here are my dialing plan rules:

Rule 1: From CS1K
Matching Pattern
$request=^invite
to=sip:(.+);phone-context
Deploy Pattern
To=sip:%1@
$auth=false
&net.sip.replacesdp.multipart=true

Rule 2: To CS1K
(note this way works fine)
Matching:
$request=^INVITE
To=sip:(.+)@
To=sip:7(.+)@
Deploy pattern:
to=sip:%1@47.185.51.110

I have a sniffer trace showing the request come in with the invite of:
<sip:2004;phone-context=cdp.udp@nash.lab;user=phone>.
This call fails - but if I change the domain name on the CS1K side to the IP address of the Brekeke SIP server,
<sip:2004;phone-context=cdp.udp@10.1.1.116;user=phone>
the call is successful.

I recently saw a new post to the wiki showing the dialing plan, but it does not seem to work.
How do I ignore the part of the invite that has @nash.lab or replace it with @10.1.1.116?

Thanks for the help!

Ok..

The $request definition in "Rule 1: From CS1K"..
$request=^invite should be $request=^INVITE .


If the issue still persists, add the domainname "nash.lab" in the hosts file.
The hosts file is "%SystemRoot%\system32\drivers\etc\hosts".

Thank you, Thank you, THANK YOU!!!
The case sensitive piece did the trick.

I'm getting a delay though in the ringing from the CS1K to a SIP endpoint - I see a TCP session trying to get established before it sends the SIP invite. But the CS1K sends a RST 3 or 4 times before it is established - then once it is, the Invite is sent and everything works. I've got a sniffer trace of it - just don't know why it's doing that.
If you have any thoughts or would be nice enough to look at the sniffer trace, let me know.

Does CS1K send INVITE over TCP??

If a SIP endpoint doesn't support TCP, add $transport=UDP in rule-1's Deploy pattern.
It converts the transport protocol to UDP.

I got it resolved - it had something to do with the L2 switch I was using (Linksys). Moved everything to a different switch and voila - it's working fine.

So I believe I'm good to go.
One last question (at least for now) - what does the rule "&net.sip.replacesdp.multipart=true" do?

Just trying to figure out the logic of what is actually happening.

With "&net.sip.replacesdp.multipart=true", the SIP Server converts CS1K's multipart-content to a single SDP-content.

Many of SIP clients including Rauland Responder can not accept a multipart-content.

Hi Jef,

can you please post or send by email the config part on the CS1000 ? and the related part on the SIP server ?

Thanks

FYI:
http://wiki.brekeke.com/wiki/Nortel-CS1000

Would anybody happen to have information on the programming on the CS1000 side to setup the SIP trunk? I see the information here on the Brekeke side and have a fair understanding of how the dial plans need to be deployed, however the Nortel programming is a complete mystery.

I know this is an old topic, but it fits our ultimate goal of making the Responder 5 talk to Nortel via Brekeke.

Thanks,

-Brian.

Brian,
Which version of CS1000/SIP trunk are you using?
Are you using Avaya SES too?

Nortel CS1000 was working well without issue but it seems Avaya did some modifications..

Add Brekeke SIP Server as a Gateway Endpoint in NRS (Network Routing Service) Manager.
Set the Brekeke SIP Server's IP address in the [Static endpoint address] field there.