Modify calling number in SIP invite on CUCM 10.5

[James Hawkins]

Hello,

I am working at a customer with CUCM 10.5 who uses MGCP gateways to access the PSTN via T1 PRI ISDN.

They use four digit DNs internally and need to prefix these with 713657 to make the outbound CLID work ok - i.e. a call to the PSTN from extension 1000 needs to send 7136571000 to the ISDN provider.

I configured this using Calling Party Transformations and this works fine e.g.

A Calling Party Transformation for 1XXX would prefix 713657.

The problem I have is that the customer has a NICE active recording system which communicates with the CUCM cluster using a SIP trunk.

The invites that CUCM sends via the SIP trunk show the full ten digits rather than the four digit extension which will not work according to company deploying the recording system.

If I remove the Calling Party Transformation then the SIP invite shows four digits and the call recording works but the outbound CLID does not work.

Can anyone suggest a way to fix this? The customer does not want to change the gateway protocol from MGCP to H323 which would be my favoured choice. Any change of calling party setting on CUCM (e.g. ticking the use external mask for calling party on route pattern) affects the SIP invite.

Ideally I need a way to modify the number in the SIP invite but I cannot find any example of how to do this.

Any suggestions are welcome.

Thanks

[martyn.rees]

You should be able to achieve this with a SIP normalisation script, which is then applied to the SIP trunk to the recorder

[James Hawkins]

Hi Martyn,

That was my thought too but I cannot find any examples of how to do this and it does not look a simple thing to do.

If you have any examples please can you post them.

Thanks

[Ayodeji Okanlawon]

How about applying a mask on the route-list>route group level. Under calling party transformation you can apply a calling party transform mask of XXXX

Give that  a go..

[Ayodeji Okanlawon]

How about excluding the calling party  transformation CSS from the sip trunk? 

[James Hawkins]

Hi, thanks for your response.

The Calling Party Transformation CSS is not applied to the SIP trunk but is applied to the T1 port of the MGCP gateway.

The Transformations are still applied to the SIP Invite messages via the trunk so I guess this is a quirk of the calling recording profile setup on CUCM.

I did try creating another Calling Party Transformation setup which stripped the unwanted digits and applied it to the SIP trunk but it had no effect.

[hyun]

hi
i have same issue
do you solve this issue??