Using SIP Normalization to strip a leading 1 off of Calling party number

[David Kritzberg]

Customer has a need to send calls to a 3rd party AA via SIP trunk accessed by two route patterns. The issue is that the CID passing through from the SIP carrier is an 11 digit#. (for example CID would read as 18185551212).  The AA uses CID to associate the calling party to a directory so that the call gets personalized. This directory is limited to 10 digits. They don't want to affect all calls into the primary CUCM system, just ones that are sent to the specific SIP trunk. After kicking it around, the only thing I can think of is to apply a Normalization script to the specific outbound SIP trunk to modify the CID.  Any thoughts?  Anyone good with normalization scripts?????

[Maren Mahoney]

I see your comment about not affecting 'all calls into the primary CUCM system'. But I would think you could use a voice translation profile on the outbound dial-peer using that specific trunk. Am I missing something?

The following would take any number that is 10 or more digits (with or without a +) and convert to just the 10 digits.

voice translation-rule 1
 rule 1 /^.*\(..........\)/ /\1/ 
!
voice translation-profile DNIS-to-10
 translate calling 1
!
dial-peer voice 10 voip
 translation-profile outgoing DNIS-to-10

Maren

[David Kritzberg]

In this particular case, the is no CUBE involved, so no dial-peer. The SIP trunk in CUCM points directly to the Vocera (3rd party) server

[Roger Kallberg]

You should be able to use a calling party transformation pattern in CM.

Create a partition that you put the calling party transformation in, create a calling search space that you put the partition in. Attach the calling search space on the SIP trunk to the system under calling party transformation.

[Maren Mahoney]

I agree with Roger. Calling Party Transformation Pattern.

• Create a Partition and CSS specific to this purpose.
• Create a Calling Party Transformation Pattern (very similar to a translation pattern in concept), and configure it with the partition from above.
• Configure the SIP trunk with the Calling Party Transformation CSS, and be sure to uncheck the "Use Device Pool..." checkbox.

Let us know if you have questions!

Maren

[David Kritzberg]

Thanks guys that did it!

[Roger Kallberg]

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@David Kritzberg wrote:

Thanks guys that did it!

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Thanks @David Kritzberg for updating that the problem is solved. Please remember to mark the post as answered and give helpful votes to the answers you think deserves it.

[JoaoPatrx]

Hi guys,

Sorry for not starting a new thread but I believe I have a related problem.

I am integrating Teams Direct Routing via Ribbon SBC with a SIP Trunk to/from CUCM. I am using in Teams the same extension number with the prefix 88 and I have a route pattern pointing to the trunk with 88XXXXX. Everything is working fine but I need external calls made from Teams to go out with the full number lets say extension 0001 to go out with 555-30001 and 0002 with 555-30002. Also keeping internal calls working with the extension number. In the CUCM environment this is achieved with the device CSS attached to the phone and the calling party transformation(CSS_3XXXX, CSS_4XXXX...). Since the device now is a trunk how can I change the calling number? Keep in mind I have several different locations each with it's own DDI.   

[JoaoPatrx]

Just solved it in Service Parameters - Inbound Calling Search Space for Remote Destination -> RDP + Line CSS