Trixbox and SIP Inbound route

[c-in2]

Hi,

I have been using AsteriskNow with
Asterisk Build:
Asterisk 1.4.18.1
for a few months now and can setup incomming and outgoing calls over my sip and iax2 trunk

lastnight I installed trixbox CE current release is 2.8.0

I can get it to phone out 100% but can seem to get th inbound routes set up. I have tried everything i can think of

here is how my trunk looks

[trunk_1]
disallow=all
username=2711461xxxx
type=peer
secret=xxxxxxx
qualify=yes
port=4569
host=196.30.xxx.xxx
dtmfmode=rfc2833
dtmf=auto
canreinvite=no
allow=all
dialformat=${EXTEN:1}
hasexten=no

and the incomming route is set up via the gui called it DIAXI and set the DID number to my username as above

I also set the asterisk console to verbose 9 and then i phone it. The phone i'm phoning from just gives a tone like there is nothing there and then hangs up and on the console nothing appears. has anyone had luck with getting this to work.

Else i will have to revert back to the very old but working version on asteriskNow

Please help

 

[gmza]

try adding the line "insecure=invite" to the trunk config so that incoming calls are authenticated based on the source IP address only and not on the username.

 

[c-in2]

Hi,

That didn't work only made it worse by keeping me from dialing out now.

Gpoing to try register a trunk to point to my working asterisk to see if that makes any difference.

Looks like trixbox doesn't even recognize recieving a call because nothing happens on the console were as on asterisk atleast it shows call from xxxx coming in but no location to route to any other ideas

C