In my configuration i have some SIP endpoints and one SIP trunk.
The normal call handling is fine.
Nevertheless, I have a problem if the phone number of the peer has changed during call forwarding or explicit call transfer.
In the SIP trace, I see new phone number in a re-invite message to the SIP-trunk. However, the re-invite message is not pass through to the SIP endpoint.
In the re-invite message the new phone number is in a PAI header. The re-invite message contains also a PRIVACY header with the value “none”.
I think that could not be a problem.
In the pjsip.conf, the option “send_pai” has the value “yes” and the option is set for the SIP trunk and the SIP endpoints.
Do I need a additional configuration?
I use asterisk version 13.9.1.