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Deutschland Telekom CUCM SIP CUBE Integration For Call Forwarding

We are running CUCM Version with an ASR1002-X as the CUBE Running version 16.6.2.


CUCM ------ CUBE ------ ITSP (Telekom)


We are able to make outbound calls with no problems. The issue comes on Call Forward All from CUCM to the  ITSP. I am attaching the integration documentation for integrating from CUCM/CUBE to Telekom.


I have modified the P-ASSERTED-ID to show the E164 number for the Directory Number. If I follow this document it changes all out bound calls to the main number, which we do not want.

We have contacted Telekom and their response was they support 1TR118 and 302 Move Temporarily. But I am unable make either work. Very little information from Cisco on this that I can find. The 1TR118 talks about turning the screening on and of, as I understand it in the From header, but once again, I can't find any way of doing this on the CUBE. When I configure the CUCM Trunk to Use the Last Redirect Number, the call forwards but has the original called number.

This is my first time working with Deutsch Telekom and I am looking for assistance from someone who may have already done this. TAC hasn't been able to resolve the issue.


I can provide the config and call traces on request, looking for some guidance.


I am also trying to get CFA working with the same provider. We are having this issue

So if you have a CUCM in the same location, a software MTP could be your answer. If you don't, like me, then Deutsche Telekom are very unhelpful.


Hey guys,


I know, it's an old post, but maybe it still helps.


Please see my config example for Deutsch Telekom in the following post:


If you have any special questions about the SIP trunk to Deutsch Telekom, just reply here or write me a PM.


One thing I can say is, that they want to have an internal number set in PAI.

So, if your number range you got from Deutsch Telekom is e.g. +49 1111111 XXX, then the PAI needs to have one of those numbers.


The first 3 rules in the SIP-profile 2000 in the config file (I have provided in the other post), does everything for you (assuming, that you have configured the sip trunk in CUCM correctly):


voice class sip-profiles 2000

 rule 1 request INVITE peer-header sip P-Asserted-Identity copy "sip:(.*)@" u01

 rule 2 request INVITE peer-header sip Diversion copy "sip:(.*)@" u01

 rule 3 request INVITE sip-header P-Asserted-Identity modify "sip:.*@(.*)" "sip:\u01@\1"


BR Björn

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