[Asterisk-Users] Asterisk-to-Asterisk call setup problem (one way works fine)

[Tor Houghton]

Hi,

I'm having a bit of a problem. I have two Asterisk servers, one serving SIP
clients on the outside of a NAT, the other on the inside. The internal one
also serves PSTN and IAX clients.

When I call someone (who is on SIP) from any phone registered with the
internal Asterisk, I get through to them no problem. The issue is when they
try to call me; for some reason they do not get routed (I am assuming the
extensions are wrong).

The outside Asterisk logs this:

Mar 13 00:18:38 NOTICE[6052352]: app_dial.c:545 dial_exec: Unable to create channel of type 'IAX'
  == Everyone is busy at this time

The outside * extensions.conf contains 

PHONES1=IAX/thought at internal

[sip]
exten => 2201,1,Ringing
exten => 2201,2,Dial(${PHONES1},20,Ttm)
exten => 2201,4,Hangup

and the outside's iax.conf is so:

[inside]
context=iax
type=friend
secret=PASSWORD
host=dynamic
tos=nodelay
qualify=yes
trunk=yes

Is there something I've missed here? (I'm suspecting I am, but I can't seem
to find any hints on how to fix it.)

Hope someone can help. Been banging my head against this for a few days
without much luck.

Thanks,

Tor