Help with SIP Diversion Header

Answered Question
Apr 20th, 2010
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Is there a way in UCM to have the diverted party shown transformed to the external phone mask when calls are diverted back to the PSTN?


Scenario: ITSP to CUBE to UCM --- SIP end-to-end. IP phone is forwarded to PSTN number and diversion headers are enabled on the outbound direction of the trunk. But unlike standard calls which are transformed to the external mask at the trunk level, diversion headers show only the private extension. ITSP rejects the diverted call because it does not accept calls from unknown DIDs.


Any ideas?

Correct Answer by kgroves42 about 7 years 1 month ago

On your CUBE you can do something like this



voice translation-rule 9
rule 1 /^9\(.*\)/ /\1/ <-- Strips the 9 from the Called Number


voice translation-rule 11
rule 1 /1...$/ /2125551212/


***  this rule Matches a 4 Digit Diversion number that starts with a 1 from Call Manager and  translates it to a number that is allowed on the SIP  network.  Depending on your Version of CM or Gateway type you might not see the 4 Digits  Diversion header, so you might try this rule 1 /.*/ /2125551212/. ***



voice translation-profile strip9
translate called 9
translate redirect-called 11


dial-peer voice 10 voip
translation-profile outgoing strip9
destination-pattern 9T
voice-class codec 1
voice-class sip early-offer forced
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
fax protocol t38 ls-redundancy 2 hs-redundancy 0 fallback none
no vad



Hope that helps

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Correct Answer
kgroves42 Wed, 04/21/2010 - 14:43
User Badges:

On your CUBE you can do something like this



voice translation-rule 9
rule 1 /^9\(.*\)/ /\1/ <-- Strips the 9 from the Called Number


voice translation-rule 11
rule 1 /1...$/ /2125551212/


***  this rule Matches a 4 Digit Diversion number that starts with a 1 from Call Manager and  translates it to a number that is allowed on the SIP  network.  Depending on your Version of CM or Gateway type you might not see the 4 Digits  Diversion header, so you might try this rule 1 /.*/ /2125551212/. ***



voice translation-profile strip9
translate called 9
translate redirect-called 11


dial-peer voice 10 voip
translation-profile outgoing strip9
destination-pattern 9T
voice-class codec 1
voice-class sip early-offer forced
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
fax protocol t38 ls-redundancy 2 hs-redundancy 0 fallback none
no vad



Hope that helps

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