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:
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:
I found this example in a previous Serusers message.
append_rpid_hf("<sip:+",
"@zettou.net>;party=calling;id-type=subscriber;privacy=off;screen=yes");
I tried calling append_rpid_hf();, but it does nothing.
I also tried
append_rpid_hf("party=calling;id-type=subscriber;privacy=off;screen=yes"
); 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@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@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,
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
or
I
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@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
ATA
186
v2. 16.2 ata18x (030909a)..Authorization: Digest
username="test",realm="voip2.test.net",nonce="3f7c49b40e81572eff05bdf50c
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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