Either create a translation pattern in CM in a partition that the gateway/trunk inbound CSS (Calling Search Space) sees that translate 1239876543 to 5432 or create a translation rule/profile in your voice gateway to do the translation either on the ingress from your service provider or egress before the call is sent to CM. If you do a search for this in the community you'll find dozens of post related to this.
Or you could simply use the DID number, preferably in +E.164 format as your extension. I’ve never understood the concept of having to alter the called or for that sake the calling number for outbound calls instead of actually using the number as assigned by the service provider.
voice translation-rule 10
rule 1 /^1239876543$/ /5432/
!
voice translation-profile PSTN-IN
translate called 10
!
dial-peer voice 1 pots
description Inbound dial peer from PSTN
translation-profile incoming PSTN-IN
For more details please look at this document Determine Voice Translation Rules
Worth knowing is that this only works on SIP or H.323 gateways. If you have a MGCP gateway you’ll need to use the translation options in CM. Also please note that this is a suggestion that you’d need to adopt to fit into your specific configuration.
Make use of the translation example which @Roger Kallberg provided. Make sure to add rules to cover your all DID.
You confused me with what numbers to use in your original post as you mentioned something that was unrelated to what you want to translate. I’ve updated both of my responses to translate to the number you wanted.
On your other part off the question, how do you think that anyone that does not have access to your configuration would be able to give you an answer on if it would have any affect on your current configuration? If you want help with that you’d need to share the details that pertains to this. As a start your gateway running configuration would be helpful.
Because I’ve tied the translation profile to an inbound dial peer that is used for calls coming from a traditional PSTN connection, such as a ISDN circuit. If you have a SIP trunk for your PSTN connection you’d use a voip dial peer for the same. As you did not give away any specifics on this I assumed that you’d be using a traditional PSTN circuit, aka pots.
With that type of connection my suggestion would not work straight off, you’ll have to adopt it to fit your specific configuration. Please share your configuration so that we don’t need to continue this guess work. Without the appropriate information it is virtually impossible to give you any quality assistance with this.
It would also be helpful if you where to share the output from these debugs, debug ccsip message and debug voip ccapi inout. Have them running at the same time and do a terminal monitor to get the output, to turn it off do a terminal no monitor.
