Asterisk dropping connected Call

Hi Guys,
I’ve been looking round at other posts but can’t find anything quite like it.

I’m using Asterisk connected via SIP to Plusnet.
My Asterisk box has an external IP address

On my laptop I’m using an IAX client connected to the asterisk box with an internal network IP.

When placing a call from my laptop the calls comes up and rings fine. the other end answers then the call cuts after about a second.
I’ve been able to test this with work DDI’s that go straight to announcement the announcement starts playing as soon as the call is answered and I hear the very begining of it then I’m cut off.

my Sip.conf entry to Plusnet:-
externip={removed}
localnet=192.168.200.0/255.255.255.0

[plusnet-out]
callerid={removed}
type=peer
username={removed}
secret={removed}
host=sip.plus.net
fromuser={removed}
fromdomain=sip.plus.net
nat=never
authuser={removed}
disallow=all
allow=alaw
allow=ulaw
callgroup=2
canreinvite=no

my Iax entry:-
[me]
type=friend
host=dynamic
regexten=1234
secret=moofoo
context=default
permit=0.0.0.0/0.0.0.0

I’ve been running iptables on the asterisk box, but turned it off for testing.

I thought it could be something to do with Sip trying to directly connect my laptop to plusnet once the call is established. my laptop’s behind another external IP address. but as it’s an IAX client I think Asterisk will have to sit in the middle of SIP and IAX so I kinda ruled that out.

Any thoughts would be greatly appreciated. it’s taken me long enough to work out how to connect to plusnet with user, authuser, etc.

Cheers,

make sure the externip and localnet are in the general section. put a canreinvite=no in there for good measure too.
make sure your rtp port range (rtp.conf) is not blocked by your firewall

and finally post a console output and/or sip debug of what happens when the call drops. you can restrict a sip debug to plusnet with something like

sip debug peer plusnet
(or plusnet-out, etc etc)