In a SIP/TLS (Pjsip) call, call will be established and we have a normal call for 30 seconds after that Asterisk terminates the call.
As I checked the logs, in the OK that Asterisk sends to the user, it puts the default Pjsip TLS port in the contact, and the user replies Ack to that port, not port in the header, so this port is not open on the firewall, as we have just a port forwarding on the firewall (XYZW->5061). After that Asterisk tries for re-transmitting the 200(OK), and then sends a BYE.
Is it a bug? or is there any specific config for this behavior in Pjsip settings?
FreePBX is 15 and the Asterisk version is 16.17.0.
I checked two different softphones (GSwave and Microsip)
External IP and local net are defined.