12-18-2020 03:45 PM
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?
Solved! Go to Solution.
12-18-2020 05:13 PM
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
12-18-2020 05:03 PM
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.
12-18-2020 05:15 PM
use first redirecting/last redirect on gateway to use to use the desk phone extension when forwarding.
12-19-2020 02:59 PM
How would i accomplish that? Are there commands?
12-19-2020 09:54 PM - edited 12-19-2020 09:55 PM
Hope this is a CUCM setup, u can do it from the gateway page on CUCM.
12-20-2020 09:01 AM
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.
12-20-2020 06:05 PM
There could be some sort of option to translate. can you share the gateway model.
12-22-2020 10:02 AM
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@"
12-18-2020 05:13 PM
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
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