How to fix INVITE header in asterisk?

Question

Asterisk

There is a registered trunk on the asterisk with some cloud PBX.
When registering, I see from them:
Reg. Contact : sip : [email protected]ххх.ххх.ххх.ххх :5060;vireg =
ххххххххххххххх_15 IP:5060 SIP/2.0 And representatives of this cloud ast say that INVITE should contain this unknown parameter vireg=хххххххххххххххх_15 That is, it should look like this: Request-Line: INVITEsip:[email protected]:5060;vireg=ххххххххххххх_15 SIP/2.0
how can the header be modified using asterisk? I didn't find anything about vireg in rfc. On the asterisk forums, I did not see data on the Request-Line modification anywhere

Answer 1

And how do these representatives justify the need for this parameter?

Answer 2

Try writing like this:
This is part of the problem. You will also need to save the vireg received from the provider in the database (for example, Asterisk itself) and substitute it in Dial

Answer 3

Everything turned out to be much simpler:
The INVITE header of the packet differs depending on how the call is sent:
If you send (there may be discrepancies in terminology) as a final peer:
DIal(SIP/trunk_name,90,m)
Then in tcpdump I see Request-Line: INVITEsip :[email protected]:5060;vireg=ххххххххххххххххх_15 SIP/2.0
That is exactly as specified in Reg. contact.
And if you send it as a sip trunk, sending a DID number:
DIal(SIP/trunk_name/5555555,90,m)
Then I see Request-Line: INVITE sip:[email protected]:5060 SIP/2.0
The problem is solved, but if someone can explain the reason, I will be grateful.