[Freeswitch-users] SIP Trunk vs routing incoming call
Robert M. Niczyporuk
rniczypo at gmail.com
Tue Jul 27 12:05:03 PDT 2010
Hello,
I've got simple problem.
My ITSP, when sends me an INVITE message, puts a destination DID (e.g.
223517864, 223517865) only in a field 'To:" of the INVITE message.
So it looks like:
---------------------------------------------------------------------------------------------------------------------------------------------------
Session Initiation Protocol
Request-Line: INVITE
sip:alamakota@[.....deleted.......];transport=udp;line=eccbc87e
SIP/2.0
Method: INVITE
Request-URI: sip:alamakota@
[.....deleted.......];transport=udp;line=eccbc87e
[Resent Packet: False]
Message Header
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bKedjaoo2068qhogsrm5o1.1
From: <sip:555555555 at appsm.ngn.yyyy.pl<sip%3A555555555 at appsm.ngn.yyyy.pl>
>;tag=SD3lb8e01-118520006-1280255979518-
To: "ab"<sip:*223517864*@yyyyy.pl;line=eccbc87e>
Call-ID: SD3lb8e01-b5526ec70ec5846b8f02e61df8d8eb32-aovtv90
CSeq: 177862656 INVITE
Contact: <sip:x.x.x.x:5060;transport=udp>
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Accept:
multipart/mixed,application/media_control+xml,application/sdp
Supported: timer
Min-SE: 60
Max-Forwards: 9
Content-Type: application/sdp
Content-Length: 332
Message Body
----------------------------------------------------------------------------------------------------------------------------------------------------------
In Request-Line there's always the same string (in this case it's
alamakota at ...........),
independently of callee DID.
Question is: How to route those calls based on 'To' field (more precisely -
based on User Part of the 'To' field) ?????
Dialplan with simple condition "destination_number" of course doesn't
work....
Is there a solution?
(Using Asterisk I'm able to analyse whole INVITE message, substract 'To'
field and then route via value in this field...)
Thanks for any help
--
rniczypo
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