@black3dynamite said in FreePBX / Random loss of audio ...:
When I was trying out FreePBX, I was using linphone and it worked a lot better.
https://www.linphone.org/
So I was going to switch to linphone. After installing I had same problem with it that I had yesterday with Zoiper, no audio. Checked codecs and all seems ok so doesn't appear to be the same issue.
Checked a SIP message trace from the Asterisk CLI and noticed the following :
From Linphone message trace, in one of the SIP messages I see:
Peer audio RTP is at port xxx.xxx.xxx.150:7078
This is the WAN IP (x'd out) of my router.
When I compare that to one I had taken when I was using the Zoiper client (that had no audio issues) I see the corresponding :
Peer audio RTP is at port 192.168.2.57:8000
This is the LAN IP of the machine the zoiper client was installed on.
I'm thinking that for the linphone client somehow the RTP stream is being sent to the router instead of back to the machine the client is on thus no audio. Is that a correct interpretation of what I'm seeing here?
I don't see anything obvious that's different about the extension setups in FreePBX for each. The setting that I thought might control this on a per extension basis was NAT Mode but that is set to No for both types of clients.
I don't see anything in the Linphone setup that would make me think I can control that from the application. I had thought that if all my extensions resided on the same LAN subnet that I wouldn't need to worry about any of the RTP traffic going to the router like that but perhaps I'm mistaken.