Translation pattern and calling number

[ogoegel]

hello

I have a small issue :

decription

one cisco phone (N° 1111) calls number 15

15 is a TP that forward to cisco phone 2222

Cisco phone 2222 does an imediate transfert to cell phone

on the cell phone the calling number is 1111. I would like to have 2222

I tried to change the calling part in the TP without success

Is there a specific parameters to check?

CUCM : Version 14

Cisco phones are 7841

Thx

Olivier

[Maren Mahoney]

Can you provide a screenshot of the Translation Pattern (with the target cell number redacted) showing the Calling Party section? Also please provide the outgoing SIP INVITE for the forwarded call going out to the PSTN.

Maren

[ogoegel]

[ogoegel]

I did a lot of unsuccessful tests. the configuration I sent is the last. it seems that changes on TP only won't work ...

[Maren Mahoney]

First, I assume the Calling Party Transform Mask is 1111 and not 2222 right?

Viewing the SIP INVITE from this CUCM to the SME would be helpful, but I'm guessing that while the "From" in the invite is 1111 that the body of the INVITE (like the Contact header) still shows the original number.

If you can't change the setting as suggested by the other posters, then you might consider a Normalization Script to locate and change this one type of call, or a SIP Profile on the egress gateway to make the modification. 

Maren

[Vineet Kumar Gautam]

Hi Ogoegel,

Please confirm if the call is going out via gateway or SIP trunk. 

In case of gateway: Go to Device>>  Gateway >> Call Routing Information - Outbound calls >> Calling Party Selection >> Select " Last Redirect Number."

In case of SIP Trunk:  Go to Device >> Trunk>> <select the SIP trunk> >> Outbound calls >> Calling Party Selection >> Select " Last Redirect number "

[Maren Mahoney]

@Vineet Kumar Gautam : What you suggest should solve the OPs issue. However, that will affect all outbound calls which may not be desirable.

Maren

[ogoegel]

hello thx for your answers

We don't have GW

the TP does not redirect to cell phone. the TP redirect to Cisco phone configured with Forward all to the cell phone

I will sent screenshot tomorrow

Cisco phones are SIP phones for info

[Roger Kallberg]

How do you get the call to redirect/forward to the cell phone if you don’t have a gateway? A gateway can utilise traditional PSTN services like ISDN or newer type of services like a SIP trunk.

[ogoegel]

ok I understand ... to call outside, the phones sends to an SME and then SBC connected to a SIP Provider

[Roger Kallberg]

From your SME you would have a SIP trunk to the SBC. So follow the advice given by @Vineet Kumar Gautam in the second section.

[ogoegel]

I can not

If I do that, everyboby will be impacted ...

I have 70 sites in France

Each site call the 15 for emergency and 15 is translated for each site . few sites ask to see the real calling number 1111 and others want to see the number of the phone that forward the call. SME trunk is for everybody

is it possible to have the choice or to configure it for each TP?

each TP is on different partition and the calling transformation should be on TP not after

[Nithin Eluvathingal]

If each site has its own call manager, implement the solution provided by @Vineet Kumar Gautam  on the CUCM trunk towards the SME. Avoid making changes on the SME trunk towards the SBC. This approach allows you to choose whether to display the redirecting number as the calling number for specific sites.

[Vineet Kumar Gautam]

If you do not want to impact all the calls for one location then you can follow below approach:

Create another SIP trunk between call manager and SME. you can use different port number to configure multiple SIP Trunks between two servers. Lets say your current SIP trunk is using port 5060 then you can create another SIP trunk with port 5064 between same call manager and SME

Then configure last redirect number on this new SIP trunk (with port 5064).

Create a route list, route group and route pattern or those specific calls. And route these calls to new SIP trunk. Let your all other calls go via old SIP trunk.

[ogoegel]

Hello All

Thx for all the solutions you proposed

We will try to first create a manipulation on our SBC to track the called number and modify the header

Create a new trunk will be done if this change does not work

Olivier