OK, sounds like it’s time for some SIP debug. Simply coming back and saying “Nope” (which is how I interpret your answer) doesn’t move the ball forward.
The NAT thing is still our number one suspect. In fact, unless you’ve got some kind of Codec mismatch (which would cause the call to fail on pickup), there are almost no things that can cause audio problems that aren’t provider/router/firewall related.
So, I’m going to say it again.
One-way audio (even when it’s both ways) is either going to be a NAT issue, an external addressing problem, or a router/firewall config. Remember that FreePBX has a firewall too, so you need to make sure that all of the addresses in there are correct as well.
I recommend using the pastebin link provided by Sangoma (see Consistent Asterisk/FreePBX Crash Issue) and post the “full” log output from a failed call with SIP DEBUG turned on, perhaps one of us will be able to spot the problem.