SIP Trunk Provier requires PAID Header

[digi]

Hello,

I have been successfully running a Multi-Tenant hosted instance of VitalPBX 3 for some time, with over a 100 users, with Different DIDs allocated to each tenant. A couple of the tenants have recently reported that when they make a call out to numbers, for some mobile operators the telephone number on the caller ID is displayed correctly, on others it says “Unknow Number” or “Private caller”. I am being told by my SIP Trunk Provider that is likely to be happening because I may not be sending the correct “PAID” header, this is their terminology all I can find is some results for P-Asserted-Identity, are the the same thing?.

I have done some reading up on this and I have been pointed to the Trunk Settings

And the device profile settings

Is anyone able to provide any advice on ways I can alter the header and PAID header, and is there anywhere I can find a complete explanation of the options for “Get DID From” and “Get CID From”?

Thanks
Paul

[digi]

Hello since, posting this I have been advised the format for the header that they are expecting should like something like the example below:

Example = P-Asserted-Identity: sip:+441234567899@123.456.789.123

Example = P-Asserted-Identity: <sip:+44[outbound CLI]@[Public IP of PBX]>

I am unsure how to achieve this format of the header, is anyone able to point me in the right direction, please?

Also, in addition, they have also provided a very large doc for me to read, but in short, if I understand correctly the INVITE should contain the following:

“Headers contained within Invite towards SIP Trunking Endpoint
From: <sip:+441625827748@83.245.6.117>;tag=3541226335-339769
Contact: sip:+441625827748@83.245.6.117
P-Asserted-Identity: sip:+441625827748@83.245.6.117”

Thanks

Paul

[miguel]

That’s something that Asterisk does by default. You can see those headers or what’s sending the PBX by using SNGREP.

If you are using VitalPBX 4, the command comes installed by default, but on version 3, you must install it using the command “yum install sngrep -y”

Once is installed, you can execute it as shown below.

sngrep -c -r

Make the call, select the invite using the arrow keys, and then enter to see all the events on call.

[digi]

Hi Miguel, Thank you that is useful, I have since discovered the reason for the issue. My Trunk had the Profile setting set to None, where it should in fact be set to use a Device Profile, either the Default PJSIP Profile or any profile that has the the required configuration for the SIP Trunk Provider.

I think you replied to my Support ticket,

! That man is a HERO !