18 Dec
2008
18 Dec
'08
11:14 a.m.
I want to authenticate in openser some users which are in asterisk conf (to do load
balancing with another asterisk).
If I delete the user from the asterisk config and insecure=invite for the openser,
asterisk accept the call, but, I want that asterisk see that it is that user which is
doing the call.
Cordialement,
BERGANZ François
http://www.acropolistelecom.net
Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
-----Message d'origine-----
De : Andrew V.Statsenko [mailto:alter@siptalk.ru]
Envoyé : jeudi 18 décembre 2008 11:00
À : BERGANZ François
Objet : ***SPAM*** Re: [Kamailio-Users] openser and asterisk auth problem.
В Чтв, 18/12/2008 в 10:31 +0100, BERGANZ François пишет:
-----I have that error:
<--- SIP read from UDP://192.168.1.156:5060
--->
INVITE sip:33662971130@192.168.1.156 SIP/2.0
Record-Route: <sip:192.168.1.156;lr=on;ftag=0c46027a>
Via: SIP/2.0/UDP 192.168.1.156;branch=z9hG4bK20d2.de427962.0
Via: SIP/2.0/UDP
192.168.1.60:43490;branch=z9hG4bK-d8754z-c15d2350d0742a11-1---d8754z-;rport=43490
Max-Forwards: 70
Contact: <sip:6833211245@192.168.1.60:43490>
To: "33662971130"<sip:33662971130@192.168.1.156>
From: "6833211245"<sip:6833211245@192.168.1.156>;tag=0c46027a
Call-ID: YTkxOWU5MGJjZjJhMWFmODZhMzZkMTU1YzUzYjUyMTc.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1100l stamp 47546
Content-Length: 231
[..]
<------------->
--- (14 headers 10 lines) ---
Sending to 192.168.1.156 : 5060 (no NAT)
Using INVITE request as basis request - YTkxOWU5MGJjZjJhMWFmODZhMzZkMTU1YzUzYjUyMTc.
Found user '6833211245' for '6833211245'
This is the key trouble in your configuration: Asterisk trying to
authenticate this call as call from _user_ (INVITE->401->INVITE+auth),
but not as _peer_.
Will you describe us where you want to authenticate you users :
- at the OpenSER frontend (BTW, I don't see auth headers in first
INVITE)
- or in the Asterisk backend ?
Which service logic you are building ?
--
WBR,
Alter
18 Dec
18 Dec
11:26 a.m.
New subject: [Kamailio-Users] openser and asterisk auth problem.
18/12/2008 в 11:14 +0100, BERGANZ François wrote:
I want to authenticate in openser some users
OK, frontend will auth users - need some code in INVITE route in the
kamailio config.
which are in asterisk conf (to do load balancing with
another asterisk).
Also little code in kamailio config.
If I delete the user from the asterisk config and
insecure=invite for the openser,
OK, also little code in kamailio for 'From' user must be same as 'Auth'
user
asterisk accept the call, but, I want that asterisk
see that it is
that user which is doing the call.
You can take this information (that it is that user) in your Asterisk
dialplan through 'From' header
--
WBR,
Alter
11:31 a.m.
New subject: [Kamailio-Users] openser and asterisk auth problem.
Can you explain me a little more please?
Cordialement,
BERGANZ François
http://www.acropolistelecom.net
Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
-----Message d'origine-----
De : users-bounces(a)lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] De la part
de Andrew V.Statsenko
Envoyé : jeudi 18 décembre 2008 11:27
À : users(a)lists.kamailio.org
Objet : Re: [Kamailio-Users] openser and asterisk auth problem.
18/12/2008 в 11:14 +0100, BERGANZ François wrote:
I want to authenticate in openser some users
OK, frontend will auth users - need some code in INVITE route in the
kamailio config.
which are in asterisk conf (to do load balancing with
another asterisk).
Also little code in kamailio config.
If I delete the user from the asterisk config and
insecure=invite for the openser,
OK, also little code in kamailio for 'From' user must be same as 'Auth'
user
asterisk accept the call, but, I want that asterisk
see that it is
that user which is doing the call.
You can take this information (that it is that user) in your Asterisk
dialplan through 'From' header
--
WBR,
Alter
_______________________________________________
Users mailing list
Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
11:41 a.m.
New subject: [Kamailio-Users] openser and asterisk auth problem.
Hi,
in asterisk sip.conf file ...
[openser]
type = friend
host=ip_address_of_open
insecure = very
username=dummy_name
proxy= host
outbound_proxy=host
nat = yes;
canreinvnit =no;
And In Asterisk CTL > show sip peers
You will see the openser peer is registered....
And See openSER debug whether Option Header and Signaling Happing properly
or not.
In openSER.cfg check asterisk ip is restricting or not ?
Thanks &Regards
Ravi Prakash Sunkara
VoIP Architect & JAVA-SIP Developer
+91-9999882776
2008/12/18 BERGANZ François <francois(a)acropolistelecom.net>
I want to authenticate in openser some users which are in asterisk conf (to
do load balancing with another asterisk).
If I delete the user from the asterisk config and insecure=invite for the
openser, asterisk accept the call, but, I want that asterisk see that it is
that user which is doing the call.
Cordialement,
BERGANZ François
http://www.acropolistelecom.net
Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
-----Message d'origine-----
De : Andrew V.Statsenko [mailto:alter@siptalk.ru]
Envoyé : jeudi 18 décembre 2008 11:00
À : BERGANZ François
Objet : ***SPAM*** Re: [Kamailio-Users] openser and asterisk auth problem.
В Чтв, 18/12/2008 в 10:31 +0100, BERGANZ François пишет:
-----I have that error:
<--- SIP read from UDP://192.168.1.156:5060
--->
INVITE sip:33662971130@192.168.1.156 <sip%3A33662971130(a)192.168.1.156>SIP/2.0
Record-Route: <sip:192.168.1.156;lr=on;ftag=0c46027a>
Via: SIP/2.0/UDP 192.168.1.156;branch=z9hG4bK20d2.de427962.0
Via: SIP/2.0/UDP 192.168.1.60:43490
;branch=z9hG4bK-d8754z-c15d2350d0742a11-1---d8754z-;rport=43490
Max-Forwards: 70
Contact: <sip:6833211245@192.168.1.60:43490>
To:
"33662971130"<sip:33662971130@192.168.1.156<sip%3A33662971130@192.168.1.156>
From:
"6833211245"<sip:6833211245@192.168.1.156<sip%3A6833211245@192.168.1.156>
;tag=0c46027a
Call-ID: YTkxOWU5MGJjZjJhMWFmODZhMzZkMTU1YzUzYjUyMTc.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1100l stamp 47546
Content-Length: 231
[..]
<------------->
--- (14 headers 10 lines) ---
Sending to 192.168.1.156 : 5060 (no NAT)
Using INVITE request as basis request -
YTkxOWU5MGJjZjJhMWFmODZhMzZkMTU1YzUzYjUyMTc.
Found user '6833211245' for
'6833211245'
This is the key trouble in your configuration: Asterisk trying to
authenticate this call as call from _user_ (INVITE->401->INVITE+auth),
but not as _peer_.
Will you describe us where you want to authenticate you users :
- at the OpenSER frontend (BTW, I don't see auth headers in first
INVITE)
- or in the Asterisk backend ?
Which service logic you are building ?
--
WBR,
Alter
_______________________________________________
Users mailing list
Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
12:23 p.m.
New subject: [Kamailio-Users] openser and asterisk auth problem.
18/12/2008 в 11:31 +0100, BERGANZ François wrote:
Can you explain me a little more please?
Well, I will try, but I not sure that I can ;-) Also, sory for my bad
english.
If I understand your task correctly (not sure) , in my point of view you
can make this service logic:
- user 1001 send initial INVITE to your service
- Kamailio frontend ask your OSS/BSS and check that user 1001 in service
(if not - reply Forbidden)
- Kamailio send 407 Proxy authorize to the user
- user send INVITE with Authorization header
- Kamailio check authorization & verify that Authorization: Digest
username="1001".. header & From: 1001@... headers have same user.
- Kamailio make a decision to which Asterisk backend send this call and
proxy it
Here the end of Kamailio frontend tasks.
- Asterisk start to execute dialplan
- Asterisk take user info 1001 from 'From' header and write this to CDR
- Asterisk terminate this call to the needed peer, IVR, queue or other
App Server as you need.
Here the end of Asterisk backend tasks.
In this case Asterisk working only as transit switch, codes translator
or application server.
If you are not very familiar with Kamailio, may be you take a look for
http://siprouter.teigre.com/doc/gettingstarted/ - we are need to talk at
same language ;-)
P.S.
BTW, this backend (Asterisk) is not very scalable. IMHO.
--
WBR,
Alter
5633
days inactive
5633
days old
4 comments
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participants (3)
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BERGANZ François -
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