SIP translation rule Voice Gateways

[BigK]

I have a DID number range from a phone company 562-123-7512 - 7579 

I would like to use 6 digit rule for inbound calls so it will match the last 2 digit but it is not working when tested. 

Rule 1:
Match pattern: ^56212375\(..\)
Replace pattern: \1
Match type: none Replace type: none
Match plan: none Replace plan: none

Thanks!

[Roger Kallberg]

The rule you have should match the range you shared. Would you mind to share a bit more details of your configuration, like where you have assigned the voice translation profile and the whole voice translation section of your configuration.

If you would want to see what is translated you can do a debug voip translation.

[BigK]

@Roger Kallberg 

Thanks for the confirmation. I would have to test again. Here is a bit of what you asked for. Let me know if any more details needed. 

Translation Profile: MS_OFFICE
Rule for Calling number:
Rule for Called number: 22
Rule for Redirect number:
Rule for Redirect-target number:
Rule for Callback number:

Translation-rule tag: 22

Rule 1:
Match pattern: ^56212375\(..\)
Replace pattern: \1
Match type: none Replace type: none
Match plan: none Replace plan: none

[Chris Deren]

Post your "show run" which will show us the actual rule, profile and where it's applied.  Also, are you just looking to strip the first 8 digits or replace them with something else?

[BigK]

@Roger Kallberg @Chris Deren 

Gents

All is working as expected as far as 6 digits extension. when I dial the 10 digits I get a message that the number is not in service. That could be due to the MPLS circuit no activated yet. 

is there away to test the 10 digits to at least know if the call is either coming or leaving the GTWY ? 

[Roger Kallberg]

If it’s a SIP integrated VGW you can do a debug ccsip message.

[BigK]

@Roger Kallberg 

Roger,

what I should I be looking for when I do the debug. what will point me that 10 digits numbers are called in or out. 

Thanks again!

[Roger Kallberg · Accepted solution]

You should look at the SIP dialogue, specific the Invite part would be of interest.