CS1K SIP Trunks to sipX / SCS500 SIP PBX

[DBrewsky]

Has anyone successfully setup SIP trunks between a CS1K (Signaling Server) and true SIP PBX (ie: Asterisk, sipXecs or SCS500)? I am able to dial from the SIP phones through the SIP PBX to the CS1K, but can't dial from a phone off the CS1K back to the SIP phone....

Ex:
CS1K ext 5302 dials SIP PBX ext 5751. Have 57 in the CS1K going to virtual trunks, and the signaling server setup with an endpoint of the SIP PBX. Although the signaling server won't ever recognize the SIP PBX as ever registering as an endoint. But... Am thinking it all has to do with the URI.. In a ethereal/wireshark capture, I see lab.tucson@xdomain.net (xdomain was made up), but the call never reaches the SIP PBX which is SIPX.XDOMAIN.NET.. The SIP information from the SIP PBX shows: 5751@SIPX.XDOMAIN.NET. Which is great.. Thats what it's supposed to be...

Any help would be appreciative....


DB....

 

[yuvami]

Hi, I went to a seminar for the scs500, and someone aske if the scs500 could be connected to the CSe 1000 e, the instructor said "no supported"

 

[DBrewsky]

Well, there is a difference between not supported and won't work... Example, CISCO 79xx series phones in SIP mode can work on the SCS500, also the Nortel 11xx series phone is not supported, yet I was able to get it to work.. Although it's not a sound investment due to cost, as the 11xx series set only supports one line..

But thanks for the reply...



DB..

 

[gabtek]

Any news on this?

 

[teekow]

3 things........

don't add the SIP PBX as a dynamic SIP endpoint, instead add it as a Static SIP endpoint since it is probably registering with itself much like the CS1K registers to its own NRS.

next, check to see if the IP you are sending the setup messages to is in fact the SIP server or is it the Proxy and/or Redirect server for the SIP PBX. Try the other if they have different IP addresses

lastly, make sure that lab.tuscon@xdomain.net can be resolved by your SIP pbx. If it can't resolve that FQDN, it may timeout without setting up the call......

post back when you find anything

Rob

 

[gabtek]

Guys, I am need a bit more details on the Nortel side. I can now call from Asterisk VOIP to CS1k phones, that works perfectly. Now I need to setup the CS1K... Where should I add the asterisk as end-point? is it in element Manager? or I have to do it with CLI? Can you help me? If I know where to add it.. I am sure it will work. I have setup route and trunk in the element manager, I can access the trunk with the trunk code. it's the SIP GW is already configured to another Cs1k site, so I really need to add astersik as End point. would be great if you could help on this..that's the last step!

Thanks

 

[donavasq]

Hi Guys,

I had the same problem at the beginning with a BCM50 3.0 but I was able to make it work with by doing some workaround with the URI settings.

Try using a hub connected to the TLAN interface and capturing the messages with Wireshark. That's very useful.

 

[yuvami]

You can try at the NRS

 

[tnphoneman]

End points are added in the NRS not element manager.

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[nnnet7]

Hi have set the AudioCodes gate to CS1K (see another my post
with question to some T.38 problems). SIP Trunk is Gate to Gate, Dynamic, where Registart IP is NRS and SIP Proxy is CS1000K Call Server.

 

[sfcstroud]

We are just undertaking the same project. I have and SCS500 r2 and CS1000 R5.0. I had set the ip address of the SCS500 as a static sip endpoint, and was trying to use my exiting sip trunks but with no luck. But if you are getting calls into the CS1000 but not receiving, it could be the sip invite string is not RFC standard. Could you post your Sip trunk configuration? Most appreciated.

 

[ineo06]

Hi,

Does someone have already make a sip trunk between SipXecs and CS1K R5 ?

Thanks