SIP Calls not Routing Externally

[crossmark76]

HI Guys,

I have recently inherited an Avaya system and am trying to configure a new SIP station but am getting an issue dialling externally.

The basic set up is -
SIP trunks from provider in SBC/Session Manager
An Avaya CM 6.3 hanging off this with all VOIP handsets (SIP trunked to Session Manager).

Recently I added a new SIP conference phone which looks to be registered correctly with session manager and I can place calls to the VOIP phones without issue. The problem I am having is that if I dial an external number from the SIP phone I get nothing and looking at TraceSM the packets are trying to route into CM instead of going directly out of the SIP trunk and I'm getting a DENIAL 1751 error.

I assume I have an incorrect setting on Session Manager but don't know much about Session Manager as I am from a CM background. There are no other SIP endpoints apart from AAM so as far as I am aware this would never have worked.

Can anyone suggest where I can start looking of the problem?
Should I expect SIP traffic to route directly out of SM and not route through CM?
Do I need to add the conference phone as an entity?

Let me know if you want further info...

Thanks
Mark

 

[Wanebo]

Start by looking at the user in System Manager. Make sure you have a CM assigned as a feature server. From there go to the CM and make sure the extension has the proper COR assigned to be able to place outbound calls. Those two items should give you a good start.

You can also do a LIST TRACE STATION [extension#] on ASA under the Advanced tab using Start Emulation. That will show you if the call is hitting the CM and what the CM is doing with it. The other option for tracing the call is done in the Session Manager CLI using the command traces and then setting some of the options for the trace there. Andrew Prokop has some good simple guides for the traces utility if you want to google them up.

 

[montyzummer]

So this will be dial pattern and routing policy in Session Manager , essentially you will need a dial pattern associated to a routing policy out to your SBC ,You will be good to add a location for you SIP endpoint then you can specify that in the dial pattern as the originating location.You will also be better off creating application sequences for the endpoint too , below links should get you where you need to be.

https://downloads.avaya.com/css/P8/documents/101044158

ACSS (UC/SBCE/SM/SME)

Not that they mean a thing anymore , get a brain dump pass the test crash the system.

 

[crossmark76]

all sorted it was the cos - thanks guys