CUCM 9 changing from 4 digit to 7 digit extension

[Peter Valdes]

Hi All,

We have a couple of sites on CUCM 9.0 with 4 digit extension, we want to change to extension numbering plan to 7 digit to avoid duplicate extension on a HUB and SPOKE cluster. I am not sure what is the best way is to approuch the task.

GW is H323 at each spoke.

We use 0 to dial outside PSTN.

We use *801 XXXX to dial between intercluster within a region (Australia / New Zeland). To dial between intercluster, we place a (*) before the 8 = *801 1234

We want to use the intercluster caller-ID as the 7 digit extention.

For example when completely migrated to 7 digit extension:

Site A - extension 801 1234 (DID number is 9123 1234)

Site B - extension 802 1234 (DID number is 9234 1234)

Both sites A and B are registered on the same CUCM Pub.

Site C - extension 803 1234 (DID number is 9345 1234)

Site C has its own separate CUCM from site A and B.

We want to be able to call each other via intercluster and PSTN (if WAN is down). E164 is not ideal.

Thanks

[Ayodeji Okanlawon]

I am not sure what you want us to help with. You have already mapped out your dial plan, which looks ok. Routing between clusters will be via ICT and route patterns. Within site users can use abbreviated dial. You need to configure xlation patterns for this. Inter site dialling will require users dialling the full 7 digit extension. This is because you can't use abbreviated dialling for inter site calls because you could run into overlapping dial plan issues

[Peter Valdes]

Hi aokanlawon,

Thanks for the reply.

I have never done the migration to more than 4 digit extension. Guides would be nice. for each task link Cisco document or links.

I know this will involce transformation mask base on some of the threads I read here in Cisco forum but not enough information on what to setup on CUCM and VG.

Thanks

[Ayodeji Okanlawon]

There are a few things that you need to do

1. Update the extension on the phones/UDP using BAT. There is no specific document i can send to you, you will need to work out a way to do this. If you are using static phones, then you might need to export all your phones, then update the extension on then and "re-BAT" them into cucm

2. You will need to ocnfigure xlation patterns, you can search on cisco cco or google for how to do this.

3. Gateway changes: Depening on what type of gateway you have, you may need to update your dial-peers if you are using sip or h323 to include the new 7 digit extensions on the phones. If its MGCP then you need to look at the inbound call routing section significant digits to allow for the new 7 digits

I dont know or see what you need xformation mask for...

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[Selvarathnam Nadar]

Hi,

Try doing the below:

In Site A

Create Route Group:

Route Group Name: Site_B

Distribution Algorithm: Top Down

Selected Devices:

     Site B Intercluster Trunk

     Gateway for PSTN (S0/SU0/DS1-0@xyz (All Ports)

Create Route List:

Name it and add Route Group (Site_B)

Create Route Pattern:

Route Pattern: *.802!

Gateway/Route List:

Prefix Digits (Outgoing Calls): 801

Dicard Digits: PreDot

In Site A Voice Gateway:

!

voice translation-rule 101

rule 1 /^8021.../ /92341.../

!

voice translation-profile DNIS_Out

translate called 101

!

dial-peer voice xxx pots

translation-profile outgoing DNIS_Out

!

Assume that you have only 4 digit number (i.e. 1234)

When you call Site B 1234 (i.e. *.8021234) the router pattern will try to going from IP WAN (via intercluster trunk to Site B), and in the Site_B phone display it will appear as 8011234, if the link is down or not reachable the router pattern will use the next path from the group i.e. PSTN line, and for that the call will come to H.323 gateway, assume the called number is *.8021234 and that will be replaced with 92341234 and the call will route to Site_B user 8021234 via PSTN.

You can replace your ANI (calling party number also)

here I think you no need to change the 4 digit extension to 7 digits. 

[Peter Valdes]

Thanks aokanlawon, I will check out that step and let you know.

Selvarathnam, I will also use your steps and let you know. Question, we are planning to use this for all our branch locations and I see limitations on the setup such as voice translation rules which can only support 15 rules. I understand your message and I know it will work. I just need to know a step guide involve using translation patters for calling and called number. Also, my manager has decided not to change the IP phone directory number, retain the 4 digit on the phone and do translation on the back end of CUCM and voice gateway.

I hope this makes sense.

We are using VG h323 and gatekeeper controlled intercluster calls.

Thanks

[Selvarathnam Nadar]

Then you need to add the voice transaltion-rule in H323 gateway as per.

[Peter Valdes]

Thanks for the confirmation. We have over 160+ location with each having a unique 3 digit prefix (801, 802, 803, 501, 502, 503 and do on...). The first digit on the 3 digit prefix ranged from [5-9]XX XXXX.

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[Selvarathnam Nadar]

Hi Valdesp,

If you say all the calls to the remote site (160+ location) or PSTN are routed through H323 voice gateway, then try to doing the below:

Now, you have 4 digit extension configured in all the location, am I right?

And, call to all location working fine by dialing *, am I right?

Route Pattern: *.[5-9]!

H323 Gateway:

!

voice translation-rule 101

rule 1 /^1/ /8011/

!

voice translatio-profile INTERSITE_ANI

translate calling 101

!

voice transaltion-rule 102

rule 1 /^8021/ /92341/

!

voice translation-profile PSTN_DNIS

translate called 102

!

dial-peer voice 123 voip

description **Intersite outgoing**

destination-pattern [5-9]XXXXXX

translation-profile outgoing INTERSITE_ANI

!

dial-peer voice 456 pots

description **PSTN Outgoing**

destination-pattern 9.......

prefix 9

translation-profile outgoing PSTN_DNIS

!

Explain:

A single route pattern is created in Site_A for all the intersite calling, the caller dials *8021234 which is located in Site_B. The call setup request is forwarded to H323 voice gateway, as 8021234 called number and 1234 as calling number (site_A caller extension). This calling number 1234 is replaced as 8011234 and displayed in Site_B called user phone. When there is PSTN call the called 8021234 is replaced to 92341234

[Peter Valdes]

Hi Selvarathnam,

Answer to your questions:

"Now, you have 4 digit extension configured in all the location, am I right?" Yes, you are correct

"And, call to all location working fine by dialing *, am I right?" Yes, you are correct

Your solution makes sense. Question: Does this apply to both site A and site B registered to the same CUCM at the HUB? And when calling site C which is on a separate CUCM cluster?

The reason why we want to retain 4 digit extension on the phone directory is to not confuse the user as they got use to seeing 4 digit numbers on their handsets. Also the 3 digit prefix is a unique ID per spoke location when calling between interclusters.

There's also the Unity Connection which I will need a solution to avoid overlapping extensions using 4 digit or 7 digit which ever works best, but at the same time, retain the 4 digit extension on the IP Phones.

Router Pattern:

*.[5-9]XXXXXX - via Intercluster Trunk (Gatekeeper)

0.!# - via Local VG Route List

Thanks again for your reply.

Regards

[Selvarathnam Nadar]

Yes, it can be work with Site_C also, just apply the rule in site C dial peer.

[Peter Valdes]

Thanks.
Question, we have more than 15 locations in our Australia / New Zealand region. How do I cover all branch spoke sites on "voice transaltion-rule 102"?
I'll test this very soon and let you know. Much appreciate your help guys.


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