GSM VOIP Gateway trunk to sip cant get working

Hi,

I have a hg4000 gsm gateway that I am trying to get working with asterisk and am having problems. Not much good with linux etc, so please bear with me.

I am told there are two ways to get calls on the gw to asterisk, one where it is requireing registration, the other which does not. The settings provoded for each case are as follows

If using registration

allow=all
dtmfmode=rfc2833
context=from-hmgw
username=hmgw
Secret=
host=dynamic
canreinvite=no

Without registration:
[hmgw]
; GSM VOIP Gateway HG-4000
allow=all
dtmfmode=rfc2833
context=from-hmgw
host=
canreinvite=no
insecure=very

Now, I have configured sip trunk not using registration, so pretty much simply put in my details as follows in the outgoing settings

allow=all
dtmfmode=rfc2833
context=from-hmgw
host=192.168.0.3
canreinvite=no
insecure=very

Incomming settings I have left blank for the moment, and registration striing is also balck.

With the above, I shoudl in theory be able to make an outbound call, but I am getting trunks are busy message.

Any help appreciated. I have managed to successfully register a sip trunk to a voip provider and can make incomming and outgoing calls okay, so i seem to have the basics right.

Regards,
Eamon