2 Oct
2003
2 Oct
'03
2:35 p.m.
At the top of the config I have the following statement....
if (method=="INVITE") {
record_route();
if (!radius_www_authorize("")) {
log(1,"radius auth failure");
www_challenge("voip2.test.net","0");
break;
};
append_rpid_hf();
};
If I call a phone with caller-id and my username is 256, it shows 256 on
the phone, if I call with a username of test, it shows 4534 on the
handset even though I have the sip-rpid set to 222.
Stephen
Subject: Re: [Serusers] caller-id with raius using sip-rpid
Retrieving caller-id when users register doesn't work because REGISTER
messages are processed by the server and the server generates a reply
only.
What you need is to insert Remote-Party-ID header field into INVITE
message, to make it work you must authenticate also INVITE messages.
My guess is that you do not authenticate INVITE messages and therefore
there is nothing append_rpid_hf can add.
Jan.
On 02-10 11:35, Steve Dolloff wrote:
append_rpid_hf("party=calling;id-type=subscriber;privacy=off;screen=yes"
not
> > responses.
> >
> > Jan.
> >
> > On 02-10 10:53, Steve Dolloff wrote:
> > > Ser doesn't appear to be passing the Caller-id to the ata at
auth
I want to retrieve the caller-id of a user from radius
when they
register and use it when I redirect an invite request to the voicemail
system so that the voicemail system can route the call based on
caller-id. I also want to be able to send it to the sip gateway when
I
am routing calls to the pstn.
I have this in my config currently and it doesn't appear to set the
caller-id in the invite message.
rewritehostport("209.242.10.153:5061");
append_rpid_hf();
t_relay();
Subject: Re: [Serusers] caller-id with raius using sip-rpid
append_rpid_hf has no parameters. Version with 2 parameters is in
unstable branch in the CVS only. What exactly do you want to do ?
Jan.
On 02-10 11:26, Steve Dolloff wrote:
"@zettou.net>;party=calling;id-type=subscriber;privacy=off;screen=yes");
I found this example in a previous Serusers
message.
append_rpid_hf("<sip:+",
I tried calling append_rpid_hf();, but it does nothing.
I also tried
> ); but I get an unknown command presumably
because I am not using
the
> right number of parameters.
>
> I only want to modify the calling-id info.
>
> Can anyone provide an example?
>
> -----Original Message-----
> From: Jan Janak [mailto:jan@iptel.org]
> Sent: Thursday, October 02, 2003 11:09 AM
> To: Steve Dolloff
> Cc: serusers(a)lists.iptel.org
> Subject: Re: [Serusers] caller-id with raius using sip-rpid
>
> If you want to add Remote-Party-ID header field then you have to
call
> append_rpid_hf function in your script.
>
> Jan.
>
> On 02-10 11:07, Steve Dolloff wrote:
> > OK, I am setting up a Voicemail system (using asterisk) and I am
> > currently doing a rewritehostport(ip:port) and then trelay() to
send
it
> to the voicemail system if an invite fails.
>
> Should I change something? See my ser.cfg and output from the call
to
the vm.
Here is the code from ser.cfg
rewritehostport("219.242.10.153:5061");
> t_relay();
>
> Here is the sip info from asterisk.
>
> INVITE sip:200@219.242.10.153:5061;user=phone SIP/2.0
> Max-Forwards: 10
> Record-Route: <sip:200@219.242.10.153;ftag=2236658534;lr=on>
> Via: SIP/2.0/UDP 219.242.10.153;branch=z9hG4bK13de.31901234.0
> Via: SIP/2.0/UDP 226.145.234.113:5060;rport=5060
> From: sip:test@voip2.test.net;tag=2236658534
> To: <sip:200@voip2.test.net;user=phone>
> Call-ID: 2218108971(a)226.145.234.113
> CSeq: 1 INVITE
> Contact: <sip:test@226.145.234.113:5060;transport=udp>
> User-Agent: Cisco ATA 186 v2.16.2 ata18x (030909a)
> Expires: 300
> Content-Length: 271
> Content-Type: application/sdp
>
>
>
> Subject: Re: [Serusers] caller-id with raius using sip-rpid
>
> Remote-Party-ID header field is inserted into forwarded requests, or
username="test",realm="voip2.test.net",nonce="3f7c49b40e81572eff05bdf50c
I
ATA
> am doing something wrong. Can anyone point
me in the right
direction?
> >
> > Thanks,
> >
> > Stephen
> >
> > I have the following entry in my freeradius users file.
> >
> > test(a)voip2.test.net Auth-Type := Digest, User-Password == "test"
> > Reply-Message = "Hello, test with digest", Sip-Rpid =
> > "8472222222"
> >
> > When I run a radclient test I get the correct info..
> >
> > radclient -f digest.test 219.242.10.153:1812 auth testing
> > Received response ID 134, code 2, length = 57
> > Reply-Message = "Hello, test with digest"
> > Sip-Rpid = "8472222222"
> >
> > This is the output from ngrep port 5060
> >
> > U 216.222.234.113:5060 -> 219.242.10.153:5060
> > REGISTER sip:voip2.test.net SIP/2.0..Via: SIP/2.0/UDP
> > 216.222.234.113:5060..
> > From: sip:test@voip2.test.net;tag=277486986..To:
> > sip:test@voip2.test.net..Cal
> > l-ID: 2687235586@216.222.234.113..CSeq: 2 REGISTER..Contact:
> > <sip:test@216.
> > 222.234.113:5060;transport=udp>;expires=120..User-Agent: Cisco 186
v2.
16.2 ata18x (030909a)..Authorization: Digest
> >
867a85bbb0da3c",uri="sip:voip2.test
> >
.net",response="1684410c130d6faa9a3c573365f36ab6"..Content-Length:
> > 0....
> > #
> > U 219.242.10.153:5060 -> 216.222.234.113:5060
> > SIP/2.0 200 OK..Via: SIP/2.0/UDP
> > 216.222.234.113:5060;rport=5060..From: sip
> > :test@voip2.test.net;tag=277486986..To:
> > sip:test@voip2.test.net;tag=b27e1a1d3
> > 3761e85846fc98f5f3a7e58.13c2..Call-ID:
> > 2687235586@216.222.234.113..CSeq: 2
> > REGISTER..Contact:
> > <sip:test@216.222.234.113:5060;transport=udp>;q=0.00;exp
> > ires=120..Server: Sip EXpress router (0.8.12dev-t16
> > (i386/linux))..Content-
> > Length: 0..Warning: 392 219.242.10.153:5060 "Noisy feedback
tells:
> > > pid=121
> > > 59 req_src_ip=216.222.234.113 req_src_port=5060
> > > in_uri=sip:voip2.test.net ou
> > > t_uri=sip:voip2.test.net via_cnt==1"....
> > >
> > > _______________________________________________
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> > > serusers(a)lists.iptel.org
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