Further testing revealed one last (hopefully) problem. With call confirm turned on in the queue (necessary to ensure calls don’t go to the users cell phone voicemail), when the agent transfers the call, the transfer to party gets the call confirm message (press 1 to connect).
This in itself is not a big deal; however, it doesn’t matter how quickly the transfer to party answers and presses 1. The transfer to party will always get, “The call has already been answered elsewhere” message.
If you go into Settings>Advanced Settings>Dialplan & Operational> Asterisk Dial Options It is currently set to: HhTtr If you adjust it to: HhTr
Everything works that I can tell. I don’t fully know the impact of removing the “t”, so be advised. Maybe someone from Sangoma or the community can weigh in on that.
Thanks again to everyone for their ideas and help, especially @PitzKey. Hopefully this is a viable solution, because it seems like it is also the most integrated compared to everything we have tried.
It prevents the transfer feature code working for the called party on normal calls. I presume it is still recognized by Queue (if not acted on, the DTMF is passed upstream) As I understand it, FreePBX normally makes calls from queues via Local channels, which then make normal calls to the final destination. I wouldn’t expect it to have any effect on transfers initiated using SIP mechanism.