Translation Patter to remove + going to ISDN

[Brandon Rebbe]

We have a local PSTN gateway for Webex call with an ISDN PRI connected.  Incoming calls work great and are routed to Webex Call. But when i try to dial out from a phone i get an error. Looked at the Q.931 and it is showing invalid elements and it looks like WxC is putting a + in front of the called and calling number which my provider does not like. 

I am trying to make a translation patter to remove it. 

so i need to take +12314516657 to 12314516657

I have tested this Rule 1 /^+\/ // / in an application i found to test translations and it works, but the router will not accept it. 

i get 
Webex-Call-1(cfg-translation-rule)#Rule 1 /^+\/ // /
% *+ operand could be empty ^
% Invalid input detected at '^' marker.

Anyone have a better way to do this? 

[Brandon Rebbe]

I was able to figure out how to get the digits stripped and it is now working. 

But when i test out forwarding my phone to a cellphone for instance the ITSP blocking it becuase the calling party number and the redirecting number are the same.  When we used CUCM we had a fix for this but i am not sure who to fix this with dial peers or Translation patterns. 

[Nithin Eluvathingal]

use first redirecting/last redirect  on gateway  to use to use the desk phone extension when forwarding. 

[Brandon Rebbe]

How would i accomplish that? Are there commands?

[Nithin Eluvathingal]

Hope this is a CUCM setup, u can do it from the gateway page on CUCM.

[Brandon Rebbe]

Unfortunately it is not CUCM. This is a local PSTN gateway feeding a Webex Call environment.  I wish it was CUCM, i would have it fixed already.   

[Nithin Eluvathingal]

There could be some sort of option to translate. can you share the gateway model. 

[Nathan Gageby]

The fix might be a sip-profile to change all redirecting numbers to a main line.  Information on this can be found here - https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-border-element/118825-technote-sip-00.html#anc12 - I have found that many carriers want this set a certain way when calls are forwarded. The commands would look something like this. 

voice class sip-profiles 1
 request ANY sip-header Diversion modify "sip:(.*)@" "sip:3045551212@"

[Nithin Eluvathingal · Accepted solution]

use the below. 

voice translation-rule 1

 rule 1 /\+\(.*$\)/ /\1/

 !

 !

 !

 !

jlrvg#test voice translation-rule 2 +12314516657

Matched with rule 1

Original number: +12314516657 Translated number: 12314516657

Original number type: none Translated number type: none

Original number plan: none Translated number plan: none