Inbound DID routing problem - SIP call

I’m in the process of replacing an old installation (Asterisk 1.2 and FreePBX2.2.1) with a new one based on the current Asterisk 1.4 and latest FreePBX.

Whilst I’m pretty happy with most things I’m stumped by what would appear to be a change in the SIP INVITE affecting inbound routes.

I’ve set up incoming routes just as I had on the old install using the DID to recognise the number. This is using sipgate in the UK and worked with the old install on the basis of the sipgate account number. With the new asterisk/freepbx combo the DID’s are not recognised and unless I have a catch-all set up the system won’t allow the incoming calls.

Using the SIP debug messages it seems that asterisk 1.2 started the first SIP exchange with

“INVITE sip:[email protected] public ip address SIP/2.0” - where XXXXXXX is my sipgate account number

These calls are successfully routed as set up in Freepbx

In the new asterisk 1.4 system with the trunk set up on the same basis the first SIP exchange is

“INVITE sip:[email protected] public ip address SIP/2.0”

CAn anyone suggest how I can fix this, either (i) by changing the INVITE behaviour or (ii) by adjusting something else in the system.

Many thanks

I’m having similar issue, i’m not been able to catch the incoming calls. Everything else is working, outbound routes, extension to extension, queues, etc… But i was not able to rpute the incoming calls.

Thank you