Here is an idea of how I test too.
Upload some music that doesnt skip any beats and doesnt have a heavy beat, such as Mozart, to a conference bridge in FreePBX.
(Music with heavy beat or wide dynamic range will cause clipping distortion and gating effect perceived as sound level dropout or false choppiness, or graininess so, no RAP/HIP Hop
If your phones can do G722 (and extension is configured), then your test music should sound dynamite.
Then if you can, at a remote site with a fast laptop or desktop, install a torrent program like uTorrent and download 3-4 different linux distros at the same time, which have DVD sized ISO’s. This allows you to nuke and un-nuke the pipe at will while you listen to Mozart (Marriage of Figaro works really good because of the wide dynamic range and no overdriven distortion like heavy metal) on the speakerphone or headset/handset in the conference bridge to listen for quality issues. You can pause and continue all the torrents with one mouse click for testing.
If your firewall isnt up to it, ie too small a cpu, etc, this can freeze/reboot your phone, stop or pause the music playback from the conference bridge, freeze/reboot/zombie your firewall causing you to need to reboot it, etc.
I had 2000+ connections downloading the last Linux Mint when it just came out. That hammers the internet, firewall, switches and even phone if your test machine is plugged into a phone.
Then, you can adjust shaping, queuing, whatever and quickly find problems.
Also, another handy tool is the built in FreePBX echo at *43. With your torrents running, your voice should have minimal delay coming back to you and should be clear as a bell without static and choppiness.
If you map *43 as a misc destination, then you can assign a DID to it and use loopback/hairpin calls to add the FXO, PRI, or other TDM trunk into the loop and hear the entire path while testing.
That can eliminate low sound levels on your dial tone trunks as a cause too. (This does happen more often than you think. Asking your carrier to bump up or down the level on your trunks makes a difference)
Although in your case, intercom calls between extensions dont traverse trunks
Another thing is *60 and listen to Allison tell you the time in a loop and GSM codec.
If toggling the torrents on/off ruins the quality of the music, echo test, or allison telling you the time, then something in your infrastructure is not working right.
If you download free voice prompt replacements in native Asterisk format, then the voice prompts sound MUCH better and therefore are much more indicative of quality issues if they exist.
This website has free super high quality voice prompts for Asterisk/Freepbx. Pick “Asterisk Native” for the best quality.
http://www.voicevector.com/Downloads.php
Good luck