Follow up from previous thread on getting Asterisk out of the call path when no longer required:
In my environment, Asterisk and Avaya Communication Manager connected via SIP Trunking through Avaya Session Manager (SIP Router, like Kamailio), using the Transfer command in my dialplan worked perfectly.
The call maintained with no channels in Asterisk. This is perfect in my use case, where all calls originate and terminate in Avaya, while looping in and out of Asterisk for features like DB entry and callback assist etc.
To Tom’s point, this will not work where Asterisk is the first leg of the call. You would need to have something in front of the call, like a SIP router.