Andrew - Read my post again. In fact, let me quote, italicize and place in bold the part you missed:
No need to be a jerk. I'm not pointing blame at FreePBX. I'm trying to figure it out, along with the other 6 people in this thread that have experienced the same problem. Because you can't duplicate it, you are the one insisting it is an Asterisk issue. In fact, I had dropped the issue due to your confrontational nature in participating in this thread. But, as additional users continue to experience the problem, it pops back up on my radar and I look into it a little further.
Now that we have addressed that, you still didn't answer my question in my last post. Let me remind you:
I'm not sure how you tested, but if you watch Rob's video here, you will see that he tested using method #2 from my post:
Myself and others have confirmed that it works fine if forcing the g722 codec from the extensions page. However, since you haven't answered the question, you have not proved that you duplicated the scenario I proposed as Method #1.
Please forgive me for using the Open Source FreePBX Distro and using it's community forum for seeking help with a problem I experienced using... the FreePBX Distro. In fact, the whole nature the community exploded right after we purchased some Commercial Modules. The whole FreePBX/Sangoma vs Ward Mundy debate started, and the Sangoma Community Forum became much more confrontational and condescending. Let's just say that it didn't take long for me to start questioning the wisdom of where we spent our money.
Since I am not a developer, I obviously do not approach such matters the same as you would. I simply don't have the experience or focused expertise you possess as a developer. But, my troubleshooting skills have shown me that Asterisk can transcode properly. (See Rob's video.) But, by configuring Asterisk, using FreePBX, with two different approaches to using the g722 codec, you get different results. In my simple mind, these results don't isolate FreePBX or Asterisk as the source of the problem. I apologize that I have used the Sangoma/FreePBX "Community Forums" for discussion, rather than strict issue reporting.
Would it be possible to get a straight answer about your approach to testing this issue? Did you force the g722 codec on the extension page? If so, you, myself and others (including Rob in his video) confirm it works, just as you insist. However, there are about 6 people (excluding any Sangoma/FreePBX developers or support staff) that have experienced this problem while configuring FreePBX by going to the 'Settings -----> Asterisk SIP Settings' in FreePBX, enabling the g722 codec, then making it the top priority. Using this method of configuration, we (at least 6 FreePBX users) experience one way audio.
I hope you are able to understand my frustration with the confrontational reply I received and are able to simply answer my question, rather than accuse me of any ill intent.