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ok, i'll try to check it today if i can, otherwise i can only do it tomorrow.

What exact part of sngrep output do i need to check to understand if there were any transcoding?

I assume, that i need to check this section of configuration? What exact part of sngrep output do i need to check to understand if there were any transcoding? v=0 o=- 258567464 258567464 IN IP4 192.168.0.24 s=digphn b=AS:84 t=0 0 a=X-nat:0 m=audio 4014 RTP/AVP 107 9 8 0 111 96 c=IN IP4...

After some time of working on my FusionPBX today i noticed, that every attempt to open any configuration web-page, redirects me to login page, and after login it redirects me to dashboard. So any time i try to open extensions configuration page: https://fusionpbx/app/extensions/extensions.php...

both of them configured to use 722 codec how can i check if there's any transcoding?

None of this option solves poor audio quality problem for me.

I'm using Digium phones (D65) in local network and Bria softphone for remote phone connection. No, i have not tried pass-through media. Where can i set it up?

I currently use my Openvox with Asterisk and all works perfectly, i wanted the same with Fusionpbx. Actually Openvox is 3G and transcoding to G722 in both cases. I'm using VS-GWM420W.

I set up a few local and external sip endpoints and they work fine. Then i tried to configure PSTN gateway (i'm using openvox, it's in the same network) and audio quality for remote side is very bad, voice sounds more metal, but local extension sound is good. endpoints codec - 722,alaw,ulaw...

Yes, looks like i accidentally added lan CIDR to 'domains' ACL instead of 'lan' ACL. Fixing it solved this strange behavior

I don't quite understand, how do i need to configure it in Dialplan -> destinations. To clarify - do i need to set up inbound routes for local calls between internal extensions?

Do i need to manually write dialplan rules for each extension i create? it sounds strange

Actually it's there

first pic - my only destination, for inbound calls from gateways action 11 office-local is ring group with 3 local phones, which works fine. set up 1 inbound route for inbound calls from the gateway, other routes were in default configuration

I'm new to FreeSwitch and FusionPBX accordingly. I set up a few Digium phones and call between them works great. Then i added a softphone, but when i tried to call to Digium phone, call is being hung up. I checked log files for something useful, but i didn't find anything useful about why call...