This is a general question - not one tied to a particular release.
Let’s say Phone A calls Phone B using CTI originate() and the media for the call is successfully bridged natively on connection of both parties to Asterisk.
However, the SIP signalling is effectively 2 calls - one from Asterisk to each phone, with 2 separate call ids etc.
Some time during the call, Phone B sends a REFER towards Phone A…
…which means it goes back to Asterisk.
Does Asterisk pass it onto Phone A as a REFER or simply reject it with a 404 Not Found?
In other words, the media bridging between the two legs is fine - is this true for the SIP signalling as well?
Many thanks for any pointers.