[asterisk-users] route INVITE sip:s at sip.test.fr

[Marc LEURENT]

Good evening!
I was wondering one thing,
I'm using freepbx to configure my asterisk server and I have a problem
with some inbound calls.

When I receive a call to an INVITE sip:01xxxxxx at myip.com I an set an
inbound route! It matches a DID number.

How can I route an INVITE sip:s at myip.com? The number only appear in the
To: Section.

Thanks!

Example:

With this one, I cannot route it (there is only the number to be reached
in the To: section)
#
U 217.36.112.145:5060 -> 192.168.95.235:5060
INVITE sip:s at 192.168.95.235:5060;transport=udp SIP/2.0.
Allow: UPDATE,REFER,INFO.
Call-ID: 02975-TP-0223ae6d-6daf01263 at sip.lecom.com.
Contact: <sip:217.66.118.145:5060>.
Content-Type: application/sdp.
CSeq: 34878212 INVITE.
From: "0614740696"
<sip:0614740696 at sip.lecom.com;user=phone>;tag=02975-US-0223ae6e-67d6c4495.
Max-Forwards: 31.
To: <sip:0170080048 at 127.0.0.1;user=phone>.
User-Agent: Cirpack/v4.41c (gw_sip).
Via: SIP/2.0/UDP 217.36.112.145:5060;branch=z9hG4bK-744D-33B812.
Content-Length: 303.
.



Whereas with this one I can do it! (there is a number in the INVITE)
#
U 87.98.202.114:5060 -> 192.168.95.235:5060
INVITE sip:0170704626 at 192.168.95.235 SIP/2.0.
Via: SIP/2.0/UDP 87.98.202.114:5060;branch=z9hG4bK1fd2c6b4;rport.
From: "0158136741" <sip:0158136740 at 87.98.201.114>;tag=as25391ca7.
To: <sip:0170704626 at 192.168.95.235>.
Contact: <sip:0158136741 at 87.98.201.114>.
Call-ID: 4091f4686a9bbc4c5223fe9c6cf60a62 at 87.98.202.114.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX.
Max-Forwards: 70.
Date: Tue, 13 Nov 2007 18:07:00 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Content-Type: application/sdp.
Content-Length: 233.
.