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How to override calling party number on a SIP trunk if none is present

jecker
Level 1
Level 1

Hello,                  

I have a 2821 router that is connected to a SIP trunk.  This router also has a PRI interface that is connected to a Mitel PBX.  The Mitel uses the router has a primary connection to the PSTN.

The Mitel will send calling party number most of the time, but some times it does not.  All calls that are sent to the SIP provider without a calling party number are dropped.  I know if can override the calling party number by using the clid commad on the dail peer, however is it possible to only override the calling party number if i don't recieve one from the Mitel?  It I do recieve a calling party number like a DID I want to forward it along in the SIP call.

Thanks. 

1 Accepted Solution

Accepted Solutions

You should be able to do that with a voice translation rule that matches an empty ANI.
voice translation-rule 30rule 1 /^\+631689\(....\)$/ /01689\1/rule 2 /^$/ /555666777/
voice translation-profile PSTN-OUTtranslate calling 30


First rule is to match any real number from the PBX, adapt that to fit your needs. The second rule matches a null value and changes that to 555666777, adapt that to what you'd like to send to your Telco.

Please remember to rate helpful responses and identify helpful or correct answers.



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View solution in original post

3 Replies 3

You should be able to do that with a voice translation rule that matches an empty ANI.
voice translation-rule 30rule 1 /^\+631689\(....\)$/ /01689\1/rule 2 /^$/ /555666777/
voice translation-profile PSTN-OUTtranslate calling 30


First rule is to match any real number from the PBX, adapt that to fit your needs. The second rule matches a null value and changes that to 555666777, adapt that to what you'd like to send to your Telco.

Please remember to rate helpful responses and identify helpful or correct answers.



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Perfect, thanks. 

Your welcome, thanks for the nice rating.

Sent from Cisco Technical Support iPhone App



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