Globalised dial plan (called/calling party transformations) question

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May 8th, 2010
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We are trying to setup a Globalised dial plan using called/calling party transformations.

We are running a CUBE with an incoming sip trunk from our ITSP and a sip trunk going into the CUCM cluster.

The goal is to have all incoming calls from the PSTN globalised with +. And outgoing calls to the PSTN should be possible with both + and normal 0.

Can anybody present me with an example on how to configure this?



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Jonathan Schulenberg Sat, 05/08/2010 - 12:58
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The closest thing to an example that exists is the E.164 dial plan is documented in the UCM SRND Dial Plan chapter. There is also a session on this at Cisco Live if you are attending in person or virtually. It takes a lot of work to understand how all of the pieces interact as well as the impact to other UC applications. At an extremely high level:

  • Incoming calls use the Incoming Called Party Transformation CSS (UCM 7.1+ only) of the inbound trunk to globalize the number. Prior to 7.1 you did not have a CSS option here which is difficult with SIP Trunks since they do not have the ISDN TON feature.
  • (Optionally) this can be localized again on the user's phone using the Incoming Calling Party Transformation CSS on the device. I prefer not to do this if the customer is receptive to seeing ANI displayed globally.
  • Outbound calls are globalized through translation patterns in the device-level CSS of the device. UCM performs all routing in global form only after it is translated.
  • Outbound calls are localized using the Outbound Called and Calling Party Transformation CSSes on the trunk.

Be sure you know the difference between transformation (modify existing call) and translation (new call). They cannot be used interchangably.

The main hurdle is that Cisco never provides a finished product example. The SRND will give you isolated examples; however, they don't fit together into any working final design. I would strongly suggest working with a partner who has gone through this and actually understands all of the details involved. On your own it will take several attempts and at least 100 hours before you have this all figured out and understand all of the details [gotchas].

jeroenhermans Sat, 05/08/2010 - 13:09
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Hi Jonathan,

Thanks for your reply.

From what I've read globalizing the inbound cll can indeed be done on the gateway configuration using the TON. But as you say I'm using a sip trunk to the cube and this does not have the functionality.

I tried going h323 from the cucm tothe gateway and then sip to the provider but I'm having all sorts of issues with that setup (no ring back, inbound call cannot be answered, no moh, etc). With the sip setup all of this is working fine.

Can I use the Incoming Calling Party Settings on the sip trunk to then use Calling Party Transformation Patterns?

Thanks for your help,


Jonathan Schulenberg Sat, 05/08/2010 - 13:16
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Adding H.323 to a gateway that ultimately does SIP wouldn't get you anywhere. The gateway still would not know the TON value as there was no ISDN connectivity to the provider.

Yes, the the Called/Calling Party Transformations are what should be in the partition within the CSS assigned to the SIP trunk for incoming calls.

jeroenhermans Sat, 05/08/2010 - 13:21
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Hi Jonathan,

I figured I could the use translation patterns on the h323 dial-peer pointing to the CUCM and then set the TON based on the calling number.

Anyway, I'm not getting this to work so I'll try with the sip trunk incoming css

I'll let you know how it goes.



jeroenhermans Sat, 05/08/2010 - 13:47
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Managed to get the inbound globalization to work.

Created 2 partitions, 1 national and 1 international.

created 1 css with these 2 partitions inside.

applied the css to the sip trunk Incoming Calling Party Settings

Created 2 Calling Party Transformation Patterns. 1 for every partition. these are removing the 0's and adding the +.

This is working great. Now all I need is the outbound part.




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