I am new to FreePBX and am trying to use it to handle a routing issue between a hosting provider and a SIP enabled Mass Notification System. In order to get myself going, I have installed FreePBX 16.0.10.14 on a VM and one my PC I have two soft phones registered as ext 3000 and ext 3002. The PC and FreePBX are on the same subnet. The Windows Firewall and FreePBX firewall are disabled.
I can place a call from either extension to the other successfully, but I get no audio in either direction.
I can call #43 from either extension and send and receive audio.
I have run wireshark on my PC and when using #43 see the call setup and subsequent RTP packets in both directions (of course). When I try to call from one extension to the other; however, I see the call setup, but no RTP stream.
I have tried enabling call recording to force audio to pass through FreePBX but still no RTP traffic.
I have limited the CODEC to ulaw, but have also tried with other CODECs enabled.
In Asterisk SIP Settings, confirm that External Address and Local Networks are correctly set. If you change these, after Submit and Apply Config, you must restart Asterisk.
Possibly, there is a quirk in the OS or the virtualization system related to two softphones competing for the same resource. Shut one of them down and set it up on a mobile device (connected by Wi-FI on the same LAN subnet) or another PC.
If you still have trouble, at the Asterisk command prompt, type pjsip set logger on
make a failing test call, paste the Asterisk log for the call at pastebin.freepbx.org and post the link here. If you are too new to post links, post the last 8 hex characters of the URL.
The above assumes pjsip extensions. If you used chan_sip, please explain, and replace the above command with sip set debug on