The setup is a (VOIP) SIP trunk provider in the UK connecting via the internet to a FreeSWITCH PBX with the Blue.box web interface that is attached to the Internet via a NAT routed ADSL last mile. The local IP telephones connect to the FreeSWITCH PBX and the different trunks from the UK (using 3 trunks for 3 different projects) should appear on one of three of the 5 extension lines on a Cisco SPA 525G.

The trunk to the UK was registered correctly as far as FreeSWITCH/Blue.box was concerned but when you called the UK phone number from the PSTN then it rang but then went to disconnect tone. What was expected was that the call would use the SIP trunk to connect via the FreeSWITCH to the IP phone on the client site and light up the correct extension.

The problem is that in Blue.box I had configured the trunk Bind to the Authenticated SIP interface and not the Authenticated SIP NAT interface. Changed it to the Authenticated SIP NAT interface and that worked. The Authenticated SIP NAT port has port 5070, auto-detect IP address via UPnP whereas the Authenticate SIP port I have is on the fixed local LAN private IP address used for local IP phones to proxy register against. These SIP interfaces are defined in blue.box under Connectivity -> SIP Interface and the trunk is defined under Connectivity -> Trunk manager.

You will see the FreeSWITCH console log has,

2011-12-25 12:29:06.274781 [WARNING] sofia_reg.c:1445
SIP auth challenge (INVITE) on sofia profile 'sipinterface_4'
for [44XXXXXXXX@sipprovider.example.com] from ip X.Y.Z.A

for each inbound call attempt. Bit of a first day of setup gotcha.