Does Asterisk support CTI meets REFER?

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.

Asterisk is a back to back user agent, with some optimisation when both sides use RTP, the codecs are compatible, and there is no need to monitor the contents of the RTP stream.

If it receives a REFER, it will transfer the leg on which it received the REFER, but will not signal the REFER to the other side (it will re-invite, if appropriate).