17 May
2010
17 May
'10
11:10 p.m.
Hello.
Is possible use some module in kamailio to authenticate against a
Asterisk Box for example?. What I need is to have a Sip Client
Registered in Kamailio, but when this client is called I need to send
the call to Asterisk which will ask for authentication (user/pass). Is
possible to do this with kamailio?. If so, can someone give an example?
Thanks in advance,
Regards,
Ricardo Martinez.-
17 May
17 May
11:12 p.m.
On 05/17/2010 04:10 PM, Ricardo Martinez wrote:
Is possible use some module in kamailio to
authenticate against a
Asterisk Box for example?. What I need is to have a Sip Client
Registered in Kamailio, but when this client is called I need to send
the call to Asterisk which will ask for authentication (user/pass). Is
possible to do this with kamailio?. If so, can someone give an example?
http://www.kamailio.org/docs/modules/3.0.x/modules_k/uac.html#id2885303
--
Alex Balashov - Principal
Evariste Systems LLC
1170 Peachtree Street
12th Floor, Suite 1200
Atlanta, GA 30309
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/
18 May
18 May
1:30 a.m.
Hello Alex.
Thanks for your answer.
I was reading the documentation, but I'm still not getting the idea on
how to use it.
Can you give me an example...
I'm sending the INVITE to the asterisk box.. the asterisk responds a
"SIP/2.0 401 Unauthorized."
From where do I get the user/pass to authenticate the
user?
Hope you can guide me here.
Thanks in advance
Regards,
Ricardo.-
-----Mensaje original-----
De: sr-users-bounces(a)lists.sip-router.org
[mailto:sr-users-bounces@lists.sip-router.org] En nombre de Alex
Balashov
Enviado el: lunes, 17 de mayo de 2010 16:13
Para: sr-users(a)lists.sip-router.org
Asunto: Re: [SR-Users] Kamailio as a UAC
On 05/17/2010 04:10 PM, Ricardo Martinez wrote:
Is possible use some module in kamailio to
authenticate against a
Asterisk Box for example?. What I need is to have a Sip Client
Registered in Kamailio, but when this client is called I need to send
the call to Asterisk which will ask for authentication (user/pass). Is
possible to do this with kamailio?. If so, can someone give an
example?
http://www.kamailio.org/docs/modules/3.0.x/modules_k/uac.html#id2885303
--
Alex Balashov - Principal
Evariste Systems LLC
1170 Peachtree Street
12th Floor, Suite 1200
Atlanta, GA 30309
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
1:34 a.m.
On 05/17/2010 06:30 PM, Ricardo Martinez wrote:
I'm sending the INVITE to the asterisk box.. the
asterisk responds a
"SIP/2.0 401 Unauthorized."
From where do I get the user/pass to authenticate the user?
In your initial request route, arm a failure route and catch the 401
challenge:
route {
...
t_on_failure("FAILURE_HANDLE");
if(!t_relay())
sl_reply_error();
}
failure_route[FAILURE_HANDLE] {
if(t_is_cancelled())
exit;
if(t_check_status("401")) {
uac_auth();
}
...
t_relay();
}
}
--
Alex Balashov - Principal
Evariste Systems LLC
1170 Peachtree Street
12th Floor, Suite 1200
Atlanta, GA 30309
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/
6:28 p.m.
I'm using the exact same script.. I even have a log line in the
uac_auth(). So is catching the 401.
But Kamailio is not sending the new INVITE with the credentials..
What else I'm missing...??
I also have defined :
modparam("uac","credential","username:domain:password")
modparam("uac","auth_realm_avp","$avp(i:10)")
modparam("uac","auth_username_avp","$avp(i:11)")
modparam("uac","auth_password_avp","$avp(i:12)")
This is the debug..
10.0.0.207 : GW
10.0.0.212 : Kamailio
10.0.0.197 : Asterisk
U 10.0.0.207:5060 -> 10.0.0.212:5060
INVITE sip:5501234567@10.0.0.212 SIP/2.0.
Via: SIP/2.0/UDP 10.0.0.207:5060;branch=z9hG4bKd24b5e47a416.
From: <sip:2408100@10.0.0.207>;tag=d24b5e47a4.
To: <sip:5501234567@10.0.0.212>.
Call-ID: d291f24b-58b6-5e54-8047-0002a4005672(a)10.0.0.207.
CSeq: 16 INVITE.
Supported: timer, replaces.
Min-SE: 1800.
Date: Tue, 18 May 2010 13:10:42 GMT.
User-Agent: AddPac AP2520 8.30V.
Contact: <sip:2408100@10.0.0.207>.
Accept: application/sdp.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO.
Content-Type: application/sdp.
Content-Length: 286.
Max-Forwards: 70.
.
v=0.
o=2408100 1274188242 1274188242 IN IP4 10.0.0.207.
s=AddPac Gateway SDP.
c=IN IP4 10.0.0.207.
t=1274188242 0.
m=audio 23032 RTP/AVP 0 8 18 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:20.
U 10.0.0.212:5060 -> 10.0.0.207:5060
SIP/2.0 100 trying -- your call is important to us.
Via: SIP/2.0/UDP 10.0.0.207:5060;branch=z9hG4bKd24b5e47a416.
From: <sip:2408100@10.0.0.207>;tag=d24b5e47a4.
To: <sip:5501234567@10.0.0.212>.
Call-ID: d291f24b-58b6-5e54-8047-0002a4005672(a)10.0.0.207.
CSeq: 16 INVITE.
Server: kamailio (3.0.1 (i386/linux)).
Content-Length: 0.
.
U 10.0.0.212:5060 -> 10.0.0.197:5060
INVITE sip:5501234567@10.0.0.197 SIP/2.0.
Record-Route: <sip:10.0.0.212;lr=on;ftag=d24b5e47a4;did=c3a.e176f9e4>.
Via: SIP/2.0/UDP 10.0.0.212;branch=z9hG4bK8a9d.47d682e.0.
Via: SIP/2.0/UDP 10.0.0.207:5060;branch=z9hG4bKd24b5e47a416.
From: <sip:2408100@10.0.0.207>;tag=d24b5e47a4.
To: <sip:5501234567@10.0.0.212>.
Call-ID: d291f24b-58b6-5e54-8047-0002a4005672(a)10.0.0.207.
CSeq: 16 INVITE.
Supported: timer, replaces.
Min-SE: 1800.
Date: Tue, 18 May 2010 13:10:42 GMT.
User-Agent: AddPac AP2520 8.30V.
Contact: <sip:2408100@10.0.0.207>.
Accept: application/sdp.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO.
Content-Type: application/sdp.
Content-Length: 286.
Max-Forwards: 69.
P-hint: outbound->inbound .
.
v=0.
o=2408100 1274188242 1274188242 IN IP4 10.0.0.207.
s=AddPac Gateway SDP.
c=IN IP4 10.0.0.207.
t=1274188242 0.
m=audio 23032 RTP/AVP 0 8 18 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:20.
U 10.0.0.197:5060 -> 10.0.0.212:5060
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP
10.0.0.212;branch=z9hG4bK8a9d.47d682e.0;received=10.0.0.212.
Via: SIP/2.0/UDP 10.0.0.207:5060;branch=z9hG4bKd24b5e47a416.
From: <sip:2408100@10.0.0.207>;tag=d24b5e47a4.
To: <sip:5501234567@10.0.0.212>;tag=as6f2d46d3.
Call-ID: d291f24b-58b6-5e54-8047-0002a4005672(a)10.0.0.207.
CSeq: 16 INVITE.
Server: Asterisk PBX 1.6.1.19.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO.
Supported: replaces, timer.
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="66243944".
Content-Length: 0.
.
U 10.0.0.212:5060 -> 10.0.0.197:5060
ACK sip:5501234567@10.0.0.197 SIP/2.0.
Via: SIP/2.0/UDP 10.0.0.212;branch=z9hG4bK8a9d.47d682e.0.
From: <sip:2408100@10.0.0.207>;tag=d24b5e47a4.
To: <sip:5501234567@10.0.0.212>;tag=as6f2d46d3.
Call-ID: d291f24b-58b6-5e54-8047-0002a4005672(a)10.0.0.207.
CSeq: 16 ACK.
Content-Length: 0.
Max-Forwards: 69.
.
U 10.0.0.212:5060 -> 10.0.0.207:5060
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP 10.0.0.207:5060;branch=z9hG4bKd24b5e47a416.
From: <sip:2408100@10.0.0.207>;tag=d24b5e47a4.
To: <sip:5501234567@10.0.0.212>;tag=as6f2d46d3.
Call-ID: d291f24b-58b6-5e54-8047-0002a4005672(a)10.0.0.207.
CSeq: 16 INVITE.
Server: Asterisk PBX 1.6.1.19.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO.
Supported: replaces, timer.
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="66243944".
Content-Length: 0.
.
U 10.0.0.207:5060 -> 10.0.0.212:5060
ACK sip:5501234567@10.0.0.212 SIP/2.0.
Via: SIP/2.0/UDP 10.0.0.207:5060;branch=z9hG4bKd24b5e47a416.
From: <sip:2408100@10.0.0.207>;tag=d24b5e47a4.
To: <sip:5501234567@10.0.0.212>;tag=as6f2d46d3.
Call-ID: d291f24b-58b6-5e54-8047-0002a4005672(a)10.0.0.207.
CSeq: 16 ACK.
Content-Length: 0.
Max-Forwards: 70.
.
Ricardo.-
5302
days inactive
5303
days old
4 comments
2 participants
participants (2)
-
Alex Balashov -
Ricardo Martinez