Outgoing Sip retransmission nat problems

I am installing freePBX in the company i work for.
It’s rather large place, and there’s a lot of NAT’ing and the topology is somewhat unclear for me, and i’m struggling with the admins about what’s right and what’s not.

Here is what I know:

FreePBX box has internal ip and external IP (see the log for more info)
I can make SIPcalls from my phone using 3G network to the PBX with sound.
I can see that it’s ringing when i try to call the phone, but i cannot take it.

I can see that i get this warning in the log:
[2013-12-06 10:07:40] WARNING[2201] chan_sip.c: Retransmission timeout reached on transmission [email protected]:5060 for seqno 102 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

I’ve turned on SIP Debug, and made a testcall from the trunk to Asterisk, where i try to pick up the phone.

I hope that you can give me any advice on what the problem is, as I really want this beautifull piece of software to work.
Link for log: