[Spce-user] routing inbound calls to asterisk trunk

[gerry kernan]

Hi
 
I have a query about the following , I have an asterisk server setup as a sunscriber, but when inbound calls come in for this subscriber the calls when sent to asterisk doesn't show the correct TO address. It shows s@ instead of the subscribers DDI, do I need to change a parameter for the subscriber?
 
 
 
Trace from asterisk server of incoming call to subscriber with DDI 35315549455  
 
 
 
SIP/2.0 200 OK
Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bKfb85.6e2280d002ea4c36f147696c1ee9e983.0;received=
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKX05QAaGg;rport=5080
Record-Route: <sip:;r2=on;lr=on;ftag=165C656A-53623188000D846C-57EFE700;ngcplb=yes;socket=sip::5060>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=165C656A-53623188000D846C-57EFE700;ngcplb=yes;socket=sip::5060>
From: <sip:0861709790 at sip.itel.ie>;tag=165C656A-53623188000D846C-57EFE700
To: <sip:s at sip.itel.ie>;tag=as742b4f58
Call-ID: eKcfsPwj.Z8Hy0 at 89.234.66.178
CSeq: 10 INVITE
Server: FPBX-2.8.1(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:s at 192.168.0.7:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 255
 
Best Regards,
 
Gerry Kernan
InfinityIT
 
Suite 17 The Mall | Beacon Court | Sandyford | Dublin 18
p: +35312930090 | f: +35312930137 | m: +353861709790
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