Forwarded Calls to SIP PSTN fail from Cisco Gateway

[Shawn Smith]

Hello All,

      I have a situation where calls that are forwarde back out to the SIP PSTN are being rejected because the calling number is not recognized. 

1). Phone A on PSTN calls Phone B which is on an Avaya CS1000M PBX.

2). Cisco Voice Gateway terminates SIP Call and hands off PRI to CS1000M

3). Phone B is forwarded to Phone C on the SIP PSTN

4). Call is rejected because the SIP Provider does not recognize the calling number

Debug (attached) of CCSIP MESSAGES and ISDN q931 show the original call coming in and being redirect back out to the PSTN, but the calling number is clearly the original calling number.  Does anyone know of a way (within the Cisco Vocie Gateway) to change the origional calling number to a number recognized by the Provider?  Thanks for any help!!

Shawn C. Smith

[Leonardo Santana]

You can use voice translation rules:

http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0080325e8e.shtml

Or you can try to modify the sip header:

http://www.cisco.com/en/US/products/sw/voicesw/ps5640/products_configuration_example09186a0080982499.shtml

The simple is the translation rule!

Rate this if helps you

Regards

Leonardo Santana

[Shawn Smith]

Thank you for your reply Leonardo! 

     I have tried voice translation rules and SIP Header modification.  Because the redirected connection is H.323 to SIP there is no Diversion Header to modify, and if I modify the From header all calls will be affected.  As for the Voice translation rule I tried to modify re-direct called and re-direct target and neither worked.  with that being said, now that I have stepped away from this for a few hours and am looking at it freshly I believe that you are correct that I can accomplish what I need with a voice translation rule.  I will try again and see how it goes.  Thank you again!!

Shawn C. Smith

[Shawn Smith]

After trying translation rules I found that the translation rules were allowing the voice gateway to send calls back out to the SIP PSTN if both PRIs to the CS1000 were down, which is not desired by the SIP provider and caused other problems.  Delving further into the issue I found that the PRI's to the CS1000 were QSIG PRI's which apparently does not support the "redirect" information element.  I changed the PRIs to primary-ni and calls forwarded to the PSTN are now working.  Lookinf at the q931 debug I can see that the "redirect" IE  is there and populated correctly, which allows the SIP "diversion Header" to be created.  Thanks all for the assistance.