External Phone Number Mask

[pea123]

Greetings

I have a Sip Trunk to Telco on a CUCM 9.1. I need the CUCM to send a particular number to the Sip Provider instead of individual phone DNs. If individual phone DNs are used the call fails.
I have opted to use the External Phone Number Mask for the DNs and checked the Use Calling Party's External Phone Number Mask under Route Patterns. The individual phone DNs keep showing for the outgoing calls. I am not using Route Groups and Route Lists.
How can I make a particular number to show for all the phone when calling to the Telco.

 

[whykap]

Uncheck the external phone mask on the route pattern and below fill in the number you want to send. Reset and test.
How are your sip trunks connecting to cucm? via cube or directly to call manager?

 

[pea123]

How are your sip trunks connecting to cucm? via cube or directly to call manager?
This is also my next question. I do have a 2911 router with vic-cards. I would like to be advised were my sip trunk will be connected.I just configure sip on the cucm and sip started working although not perfect. I have internet on my LAN and both the cucm and router are part of the LAN. With regard to sip trunk I never did anything on the router so I guess the sip trunk is on the cucm then.

I have now managed to mask the number according to your advice but I dont get audio when I call out. The call does go through.

 

[whykap]

I don't see how the mask would affect audio. Are you saying you get audio with mask turned off?

 

[pea123]

When I did not mask earlier on I could not connect to the sip provider because I was issuing a wrong caller ID. So by masking I managed to present a correct caller ID to the sip provider and therefore my calls could go through.
Since I can now manage to go through masking is no longer an object of discussion.
I have however noticed that (since I can now get through) the calls do not have audio or ringing alert.
At this point my problem is that of audio.

 

[whykap]

In the sip trunk configuration do you have the media termination point (MTP) checked?

 

[pea123]

My MTP has been checked all the time.

 

[whykap]

I would start looking at debugs at this point. Since you are terminating the sip trunk directly to cucm, you will need to use RTMT for collecting them.