moving to E164 call routing. issues

[MattCooper30]

CUCM 12.5 / single cluster; 1 pub 2 subs

Working on moving to E.164 call routing.  we have 30 PRIs, that are located at remote offices, as well as our main campus.  We are hoping to sign off on a SIP trunk deployment this week. 

In the mean time, we are currently having an outage with our long distance carrier, and i am working on getting the calls directly to the office, since we are having carrier issues.  They both reside on the same server, hence we should be able to dial them directly. 

I have a translation pattern built, but in RTMT, i am getting a (100) Invalid Information Element Contents.

I believe i am missing a dial peer on the router, which is causing the router to drop the call. 

I am looking at implementing this without breaking the existing call flow, which unfortunately is all across the board.  they have 4 digit 5 digit and 6 digit call flow, across the PRIs, some of which look for 10 digit, some look for 7 digit. 

once we get the SIP trunk, i am working on migrating the existing dial plan to e.164, simplifying the routing patterns, route lists, etc. 

Any help would be greatly appreciated.

[Carlo Poggiarelli]

Hi Matt,

How did you deploy the VG, SIP,H323 or MGCP?

Please send the output of a debug isdn q931 and how you configured the translation pattern.

To test the E.164 call routing, you could create a dedicated partition and CSS to assign to a test phone and TP. Than on VG you could add a dedicated dialpeer.

Please let me know 

Regards

Carlo

[MattCooper30]

thank you both for responding.  Sorry.  been dealing with a long distance carrier issue for the last two days and have been tied up.  i will do an output for the debug tomorrow, and show how i configured the translation pattern.  i do have dedicated partitions and CSS to assigned test phones.  i havent done a dedicated dial peer on the VG / or the Router that has the PRIs, as i am not a 100% familiar with them.  havent really looked them up.

[Roger Kallberg]

Without you sharing more details around your specific setup it would be virtually impossible for anyone to provide you with any help. For one you’ll need to describe how your current integration with the voice gateway(s) is setup and if it’s dependent upon dial peers it would be needed for you to share the configuration of the gateway where you are trying to pass the call via.