I have one PSTN line and two SIP trunks. Calls coming in via PSTN can, using follow me, go to my cell phone using a SIP trunk. Calls coming in via SIP, can go out via PSTN. Meanwhile, calls coming in via SIP, while they connect successfully using outbound SIP, no audio is heard at either end.
This is usually caused by incoming RTP packets being blocked by the firewall. In the working cases, Asterisk has audio to send out; the incoming audio appears as ‘replies’ and is passed to Asterisk.
For an easy test, configure Follow Me to play hold music instead of ringing. If that allows audio to pass correctly when answered, the ‘proper’ fix is for your router/firewall to forward the RTP port range (default is UDP ports 10000-20000) to the LAN IP address of the PBX.
1 Like
It works fine with Follow Me set to play hold music. This behavior is acceptable, without changing rules on the firewall.
This likely explains some issues I had with SIP clients on cell phones. I resolved those issues by using a VPN connection for them.
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.