Is anyone using voicenetwork.ca on Sail? I recently attemted to use them and added them as a carrier and the wierd thing is when I create a trunk it connects and shows as registered their end but when I call the DiD number it says I'm off line

I spoke to their support but they don't have any knowledge of Sail and they said that the call is going out from them and then getting bounced back to them

Is there some command that I can use to see what is happening my end? They told me to set it up as per their Trxbox setup and to make sure that insecure was set to very and to allow there IP access to my server

I have tried as SIP and IAX2, I can't get it to register as an IAX2

below is the example SIP setup for Trixbox
PEER DETAILS:
host=sipgw.voicenetwork.ca
context=from-trunk
type=friend
username=1390976565
fromuser=1390976565
sendrpid=yes
qualify=yes
canreinvite=no
insecure=port,invite
secret=password
USER DETAILS:
type=peer
host=sipgw.voicenetwork.ca
qualify=yes
canreinvite=no
context=from-trunk
and the Asterisk example:
This register line must appear after the [global] section
; and BEFORE any other peer definitions
register => 1390976565:password@sipgw.voicenetwork.ca/1390976565
;
; Make sure that you have checked the Send DID option under the EDIT DID option
;
; Edit your SIP.CONF file and place the following lines in it
[voicenetwork-in]
type=peer
host=sipgw.voicenetwork.ca
fromuser=1390976565
dtmfmode=rfc2833
qualify=yes
canreinvite=yes
context=voicenetwork-incoming ; incoming DID calls will arrive in the voicenetwork-incoming context
[voicenetwork-out]
host= sipgw.voicenetwork.ca
type=friend
username=1390976565
fromuser=1390976565
fromdomain=voicenetwork.ca
sendrpid=yes
qualify=yes
secret=password
dtmfmode=rfc2833
disallow=all
allow=ulaw
context=voicenetwork-incoming ; incoming DID calls will arrive in the voicenetwork-incoming context
insecure=port,invite
;
; Edit your EXTENSIONS.CONF and place the following information in it
;
[voicenetwork-incoming]
exten => _X.,1,Answer
exten => _X.,2,Wait(1)
exten => _X.,3,SayDigits(${EXTEN})
exten => _X.,4,Wait(1)
exten => _X.,5,Echo
exten => _X.,6,Hangup
; To use VoiceNetwork.ca to termination your calls
; add the following line to your extensions.conf file
;
exten => _X.,1,Dial(SIP/voicenetwork_out/${EXTEN})
Thanks in advance,
Del