[asterisk-users] Internode weirdness

[Da Rock]

I have an updated asterisk 1.8 server running on Freebsd 8.1, and 
connecting through a Freebsd 8.1 pf firewall with a dumb modem adsl 
connection (in other words FreeBSD is doing all the hard work). I am 
trying to connect with Internode nodephone, but they aren't really 
willing to spend the time to work it out (depending on who you get to 
talk to), and they reckon its all working as it should.

I was originally running a 1.4 server trying to get it working, but when 
I didn't have a great deal of success setting it up, and I noticed 
features were missing that I wanted, and 1.8 was finally ported, I 
jumped on the chance and updated.

I was originally able to get outgoing calls working after quite a bit of 
fiddling with settings, but no incoming. I finally found some info to 
tweak the firewall to suit the asterisk and voip services, and now I can 
finally get perfect incoming calls- but now outgoing won't work at all! :(

I've been hammering at this for days now- working my google foo like 
crazy to get some clues as to why. Nada...

So what am I missing? The only facts I have are:

Internode insist their setup gets around NAT issues, so in an ordinary 
ATA setup you don't need nat. The proviso is that it needs to be on a 
dmz- basically they say open all connections from their server and 
direct them to the ATA. (I did have outgoing calls working in this 
scenario, but I couldn't get incoming; and to boot if I had other 
clients outside the NAT- which I am looking at doing as well, just not 
going through internode- it basically won't allow it)

The firewall is setup to NAT port 5060 as 5060 to the internode server 
and redirected on return. RTP 10000-20000 is directed through to the 
server as well.

SIP debug on: On making an outgoing call I get retransmission timeout 
errors and this:

WARNING[988]: chan_sip.c:19069 handle_response_invite: Re-invite to 
non-existing call leg on other UA. SIP dialog 
'481cf0543743e6bb7006991d409ed3bc at 150.101.178.33:5060'. Giving up.

The dialog can change too- if I change fromdomain it changes accordingly.

-- SIP/sip-out-0000001d is circuit-busy
   == Everyone is busy/congested at this time (1:0/1/0)

Tcpdumps and logs show messages going out of asterisk and both 
interfaces on the firewall, but none coming in.

Registry and peers list show the Internode connections are fine- 
qualifying is working.

I have also followed recommendations to separate incoming and outgoing 
peers (despite the added complexity), so I have an sip-in and sip-out 
peers with settings for internode; although even if I comment out one 
and adjust the dialplan it still shows the same error.

I also tried turning off the externip setting- no luck.

I'm at the end of my tether- I'm ready to turn a laptop into a missile! 
And the lack of interest is killing me.... Any help would be much 
appreciated at this point- its doing my head in!

Cheers