Sip Trunking For BCM50 1

[dogg1]

I'm trying to install SIP Trunking on a BCM50. I have all Signaling Server prompts set that relate to SIP. I have also setup the SIP options in the BCM. The question is, when I look in the Sig Server under Maintenance->GEN CMD->SIP->SIPSHOW->RUN I notice that the Active Proxy isn't registering? Can anybody tell me why and what prompt I need to change.

Thanks!!!!

 

[atascoman]

I believe you have to have the BCM register via an NRS server to get it to sync up. Is your SS a co-res with the NRS installed?

 

[dogg1]

Yes my SS is a co-res with NRS. I have put in all my programming in the NRS and SS. In the SS the Active Proxy shows that it isn't registering and keeps timing out. So, apparently I have missed something in programming, but not sure where.


Thanks for the feedback.

 

[CyberX666]

Under SIP settings make sure your "Outgoing Transport" is set to UDP. BCM does not support TCP SIP.

 

[CyberX666]

Just to add:

In the NRS:
1) BCM and SRG do not support TCP SIP Transport. TCP is default on the NRS while UDP is default on the BCM/SRG.

Under Configuration, Gateway Endpoint, endpoint x (x = BCM or SRG endpoint in question), change the SIP Transport to UDP. Commit and save the change.

2) BCM and SRG also do not support Dynamic Registration, so Static must be used.

In the NRS under Configuration, Gateway Endpoint, endpoint x (x = BCM or SRG endpoint in question), ensure SIP Support is set to Static. Although CS1000 can use "Dynamic" for SIP between the CS1000, BCM does not support Dynamic but only Static.

hope this helps

 

[dogg1]

Thanks Cyber,

I have the settings in place that you recommended but still not registering. I noticed however, that my domains in the Sig Server and NRS are different. Will this error keep the BCM from Registering?

Thanks in advance!

 

[dogg1]

Well I have finally got half working. I can now recieve calls on the BCM but cannot not make calls. The Alarm comes up with " Call rejected, due to all 0 keyccoded trunks being in use on the H. 323 interface". I have no H 323 trunks in this particular BCM, just SIP. My guess is it can't find the SIP trunks? When I look in the BCM Monitor the lines don't even show up. Any help would be appreciated

Thanks!


dogg1

 

[gberger]

When you did put the keycode for the SIP trunks did you restart the Bcm or restart the "feps" service?

 

[dogg1]

The BCM came with the SIP trunking already installed.

 

[CyberX666]

On the BCM under Telephony-->Lines--> Active VoIP Lines, do you have any lines on window on the right? If so are they in a line pool and that pool assigned to a route and the pool assigned to a set?

Sorry if you have done all this just want to tick all the boxes.

 

[dogg1]

Yes I have that all setup on the BCM.

 

[dogg1]

I was looking in the Applications Resources Tab. It shows 4 licensed SIP Trunks and NO IP Trunks. However, when I go into the Telephony Resources it says I have 4 IP Trunks and no mention of SIP. Do I need to call Nortel? Thanks for eveyones help!

 

[vztech]

No thats correct. So your BCM registered?


sound like your routing might be the issue.

It's only dialtone-VZ

 

[vztech]

could you list your routing

VOIP line 1
Pool:blocA

route 000 use pool: bloc A

Destination code
etc.

It's only dialtone-VZ

 

[dogg1]

OK,

So I have all my Voip Trunks in Pool: BlocA. I'm using route 002 for BlocA (Private), route 001 for BlocA (National), and finally Route 000 has Pool A. I'm absorbing 0 in the dialing plan.

 

[vztech]

did you mean route 1 block b National ?

Try this
Under Remote Access Package -select 01 ,add blocA (line pool access)
then go to Lines-active voip lines- select 001 then under restrictions- use remote package 01.

do this to to the 4 voip trunks.

did you already setup your routing table for the ip trunks under telephony resources ? sip setting, domain etc ?

It's only dialtone-VZ

 

[dogg1]

Hey vztech,

I think the problem i'm having is in the Sip Setup area. I have not put anything in the routing table because I was unsure as to what I needed if anythng. I have a BCM50 3.0, so the tabs may be a bit different. In the SIP Setting tab I have my local domain entered, fallback disabled. Gateway says its running. In the SIP Proxy Tab i have my Domain Name entered, "Route all calls using Proxy" checked, MCDN Protocol is CSE. That is all I have changed in that tab. My SIP Media parameters are good, that much I do know. My SIP URl, I have the same domains in there as I do in the Signaling Server so I'm pretty sure that tab is ok too. The Authentication Tab is not being used at this time.


Thanks for all your help!

 

[vztech]

dogg

add a route in sip
you need to tell the voip trunks where you want to go when you dial a number.


I have a 3.0 I'll fire up and look at.
but from what I remember you need
Name: ( wherever the far end is ) Bermuda
Destination digits : example. 56 ( to call x5676 at the far end)
Domain name:
IP address: at the far end ( I had mine go to other BCM's)
but I think this IP should be your sig server
select CSE,MCDN

let me know how you make out

It's only dialtone-VZ

 

[tnphoneman]

Did you check to see if the payload size matches at both ends. The CS1k defaults to 20 and the BCM defaults to 30. That causes a problem calling one way but not the other.

I bet you can call the BCM but the BCM can not call the CS1k.

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[SupportDude]

This has been a point of contention for a while!
The NRS does not support SIP proxy unless it is a 5.5 running on the Linux OS. Otherwise, its just a redirect server.
You won't see registration in the NRS when using SIP because SIP doesn't use a gatekeeper.
When configuring SIP on the BCM50r3.0, the only thing you should have in the SIP Proxy tab is the domain, nothing else. The SIP parameters, domain, and URI map should match up to the Sig Server.
The BCM50 uses the IP routing table for the SIP calls.
Make sure you don't have the wildcard box checked in the H323 setup ... which also should have nothing configured.
Hope this helps!


-SD-

http://www.mysupportguy.net