No audio between internal extensions


I have installed FreePBX ( Current Asterisk Version: 16.11.1 ). The endpoints are Grandstream phones (1615/1760/2160), which are able to register with the PBX server successfully. I am able to dial from one extension to the other, but cannot hear voice from either extensions. I dialled *60 (Speaking clock) and I can hear the time announcement clearly, though while running the *43 (Echo test), I cannot hear back my voice. All the phones and PBX are on in the same network/LAN.

I tried going through the Asterisk log file (full) after setting “sip set debug on” through Asterisk CLI ( both through the FreePBX GUI), but not sure which lines to paste here, so pasted 1000 lines here:

What am I missing here ?

That command is for the legacy chan_sip. Instead, type
pjsip set logger on
make another test call, paste the log.

This is with pjsip logger on

In Asterisk SIP Settings, set External Address correctly and set Local Networks to /23
then restart Asterisk and retest.

My local network is set to Should I use the same or use the one you suggested. The FreePBX server has no direct connection to internet. It sits behind a router/firewall (and so does all the phones). Will it still require setting up the external address?

Thanks for the help Stewart. Changing the values resolved the problem.

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