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New Member

Translation rule problem with SIP trunk

Hi All,

Network Topology: We have a Fenestrae (faxination) server connected through a SIP trunk to CUCM 7.0.2 which is connected to H.323 Cisco GW.

Here it is: Fenestrae <--> SIP Trunk <--> CUCM <--> H.323 Cisco GW  <--> PSTN.

Problem Description: Calls from Fenestrae to the PSTN go out with no calling number so the PSTN assigns the number 1000 for the outgoing Fax calls. We need to change this number to another for example 1800.

The following setup was done but it is not working:

* SIP trunk has been assigned a CSS having one partition pointing to a translation rule ( ! ) matching all outgoing calls and translating the calling number to 1800 and the translation rule contains the normal CSS which routes the calls.

* I tested this setup by assigning the CSS pointing to the translation rule to my IP phone and it is working fine, the number is translated to 1800, but for some reason it is not working at all with the Fenestrae calls.

Appreciate any help please to solve this problem, thanks.

1 REPLY
New Member

Translation rule problem with SIP trunk

Do you have a specific device pool assigned to the SIP trunk?

My general practice I used to do is setup a Calling Party Transformation, assigned it a unique CSS.  Then assign this CSS to the Incoming Calling Party Settings in the device pool configuration for this particular SIP trunk.

After that do some troubleshooting at the H323 GW, ensure that you can see the translated/transformed number is showing up correctly at the H323 GW.

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