We are running Freepbx 14 with Asterisk 13. Our extensions are pjsip.
When we do an attended transfer, if the transfer is completed while the final destination is still ringing, there is no audio.
User 1 calls User 2; User 2 does an attended transfer to User 3. If User 2 completes the transfer while it is still ringing User 3 then User 1 and User 3 have no audio at all. If User 3 puts the call on hold and picks it back up, audio comes back both directions.
An attended transfer where User 2 waits for User 3 to answer, works fine; both User 1 and User 3 have audio both directions. A blind transfer also works fine.