[Kamailio-Users] SIP Re-Routing using Private Header

[Geoffrey Mina]

Hello,
I have a problem which I am not sure the best way to solve with
Kamailio.  I have many asterisk servers which use Kamailio as their
outbound gateway to route calls to the PSTN.  This works great, I use
the LCR engine to control routing.

What I want to do is have the ability to dial a random SIP URI from my
asterisk servers, but route the call through my Kamailio server for
accounting and security purposes.  My asterisk servers are not allowed
SIP messaging from anything other than my Kamailio gateway.  What I am
considering doing is something like this:

Since asterisk is fairly limited in your ability to route calls, I
need to do a little magic to make the call route through a proxy.
Maybe I'm wrong, but I haven't yet been able to figure it out.  My
theory is that I will add a special header at the asterisk level and
send the invite to Kamailio.

[test]
exten => 1,1,SipAddHeader("P-Forward-URI: bob at somedomain.com")
exten => 1,n,Dial(SIP/forward at kamailio,30)


[From Asterisk To Kamailio]
INVITE sip:forward at 10.1.1.1 SIP/2.0
Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bK776asdhds
Max-Forwards: 70
To: <sip:forward at 10.1.1.1>
From: "" <sip:5555551212 at pc33.atlanta.com>;tag=1928301774
Call-ID: a84b4c76e66710 at pc33.atlanta.com
CSeq: 314159 INVITE
Contact: <sip:5555551212 at pc33.atlanta.com>
Content-Type: application/sdp
Content-Length: 142
P-Forward-URL: bob at somedomain.com


if(is_present_hf("P-Forward-URL")){
   //what do i do here to rewrite the To and INVITE parts before doing
record_route() and t_relay()
}


Maybe I'm totally off track here, but this is all I have come up with
so far!  Perhaps there is a mechanism in SIP which already allows me
to do this, and I don't know about it... I don't know what I don't
know :)

Thanks,
Geoff