I have a setup with Asterisk 13.18.3 and 3 PJSIP extensions. Those are used by 2 Android devices with linphone, and 1 Linux desktop with Jitsi. The scenario I’m working with is an attended transfer as follows:
- Android/Linphone-1 calls Linux/Jitsi
- Linux/Jitsi put the call on hold and calls Android/Linphone-2
- after the second call is established, Linux/Jitsi transfers Android/Linphone-1 to Android/Linphone-2
I’ve monitored the SIP traffic on the Asterisk server, and the transfer is successful. My problem is that when Linux/Jitsi sends the REFER request to Asterisk, no new INVITE with Replaces header is generated by Asterisk. It seems Asterisk just merges the calls internally, without the caller and callee know about it.
Is it possible to have Asterisk sends new INVITE with Replaces header?