Integrating Nortel CS1000 & Rauland Responder 5 via Brek

jwiseman01 · Post by **jwiseman01** » Thu May 27, 2010 8:37 pm

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?

hope · Post by **hope** » Fri May 28, 2010 9:36 am

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

Mike · Post by **Mike** » Fri May 28, 2010 12:54 pm

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 .

james · Post by **james** » Tue Jun 01, 2010 4:08 pm

which edition of SIP Server are you using?

paste your current Dialplan rules here.

jwiseman01 · Post by **jwiseman01** » Tue Jun 01, 2010 5:29 pm

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!

james · Post by **james** » Fri Jun 04, 2010 10:13 am

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".

jwiseman01 · Post by **jwiseman01** » Fri Jun 04, 2010 3:22 pm

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.

james · Post by **james** » Mon Jun 07, 2010 1:57 pm

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.

jwiseman01 · Post by **jwiseman01** » Mon Jun 07, 2010 3:45 pm

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.

james · Post by **james** » Mon Jun 07, 2010 4:07 pm

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.

al.thewizard · Post by **al.thewizard** » Mon May 09, 2011 7:47 am

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

lakeview · Post by **lakeview** » Mon May 09, 2011 10:29 am

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

bsworkman · Post by **bsworkman** » Thu Jan 17, 2013 7:38 am

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.

hope · Post by **hope** » Thu Jan 17, 2013 9:39 am

http://wiki.brekeke.com/wiki/Rauland-Re ... all-System

davi · Post by **davi** » Thu Jan 17, 2013 1:18 pm

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..

ambrosio · Post by **ambrosio** » Thu Jan 17, 2013 6:42 pm

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.