Do you have two Local Networks fields in Asterisk SIP Settings? There seems to be a FreePBX bug for pjsip trunks in that regard. See this thread: One way audio .
If that’s your problem, you may be able to use the same fix.
If you still have trouble, capture the SIP on an incoming call. At the Asterisk command prompt, type
pjsip set logger on
and make a failing incoming call. The SIP will appear in the Asterisk log, interspersed with the normal entries. Post the (suitably redacted) section of the log including the incoming INVITE and the 200 OK response.