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?
Thanks.
PS:
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.
It shouldn’t. It should use the value of Port to Listen On (in Asterisk SIP Settings → chan_pjsip tab), which should be set to XYZW, and your firewall should forward XYZW->XYZW. After changing the port, restart Asterisk.
If you want to use XYZW externally but 5061 internally, you can instead set external_signaling_port=XYZW . See
This is not exposed in FreePBX, so put in /etc/asterisk/pjsip.transports_custom_post.conf [0.0.0.0-tls](+type=transport) external_signaling_port=XYZW
Your next problem is if you want a mobile app using TLS to work both on local Wi-Fi and over mobile data. You would need a DNS setup with appropriate SRV records pointing to 5061 when queried internally, but XYZW externally.