Hi there, I've just changed my VFX connection over to an "asterisk" connection.

I've tried many different configurations (sip/extentions), but I'm not able to get incoming calls to route correctly through to the asterisk box.  Outgoing calls (asterisk to realworld) work fine.  Internal dialing seems to work fine.

Its calls from realworld to asterisk don't seem to reach the box.  (tcpdump shows no sip packets arriving). The box has a real ip address.  Seems to show as registered ok:

asterisk*CLI> sip show registry
Host                            Username       Refresh State                Reg.Time
pan.wxnz.net:5060        7981xxxx            165 Registered        Sun, 02 Aug 2009 22:22:20

Can anyone see anything wrong with my confs?

---sip.conf:
[General]
port = 5060
dtmfmode = rfc2833
disallow=all
;allow=g729  ; tried this in and out..
allow=ulaw
allow=alaw
registertimeout = 20
regseconds = 180
register => 7981xxxx:xxxxxxxxxxxxxxxxxxxx:xxxxxxxxxxxxxxxxxxxx@pan.wxnz.net/7981xxxx

[VFX]
type=peer
fromuser=7981xxxx
host=pan.wxnz.net
insecure=invite,port
canreinvite=no
nat=yes
secret=xxxxxxxxxxxxxxxxxxxx
username=xxxxxxxxxxxxxxxxxxxx
context=default
---

---extensions.conf:
[globals]
static=yes
writeprotect=no
clearglobalvars=no

[general]

[default]
exten => s,1,Verbose(1|Unrouted call handler)
exten => s,n,Answer()
exten => s,n,Wait(1)
exten => s,n,Playback(tt-weasels)
exten => s,n,Hangup()

exten => 1234,1,Dial(SIP/xlite,30)
exten => 1234,n,Hangup()

exten => i,1,Answer()
exten => i,n,Playback(tt-weasels)
exten => i,n,Hangup()

; added this on in just in case... it wasn't picked up by s above...
exten => 7981xxxx,1,Answer()
exten => 7981xxxx,n,Playback(tt-weasels)
exten => 7981xxxx,n,Hangup()
---