Shrug Sorry. You've changed a bunch of stuff on your machine. This has caused stuff to break - eg, the requirement for STUN.
I know for a fact it works, as I use Asterisk 13.5 with FreePBX 13 and G722 as my primary phone system.
From here I suggest you start doing some tcpdumps and check the sip headers coming from and going to the phone.
Another option is to deploy a NEW machine. Don't change anything that you don't have an EXTREMELY good reason for changing (eg, RTP ports.). Probably the only thing you want to change is to go to SIp Settings and move G722 to the top of the list there.
Then, try with a phone to get *43 to work. If, on a fresh built machine, with no changes, it doesn't work, I will happily spend any amount of time figuring out what the problem is.
But I know it will work, because I normally build two or three new machines a day with Ast 13 and G722, and they all work. If you have somehow found a way to make it not work, that's really REALLY bad, and I care deeply.