No audio between internal extensions

FreePBX Endpoints
1

Hi

I have installed FreePBX 15.0.16.72 ( 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:

https://pastebin.freepbx.org/view/f610255b

What am I missing here ?

2

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

3

https://pastebin.freepbx.org/view/8653fbcf

This is with pjsip logger on

4

In Asterisk SIP Settings, set External Address correctly and set Local Networks to
192.168.0.0 /23
then restart Asterisk and retest.

5

My local network is set to 192.168.0.0/16. 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?

6

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

7

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