16 Nov
2014
16 Nov
'14
8:25 p.m.
Because both kamailio and asterisk use the same db table for
authentication, see the auth_db module parameters in kamailio config.
The REGISTER request from sip user is authenticated by kamailio using
auth_db module and upon success kamailio generates REGISTER request back to
asterisk (using the credentials sent by sip user for authentication with
kamailio), this request is now authenticated by asterisk using realtime sip
users interface.
Thank you.
On Sun, Nov 16, 2014 at 2:53 PM, Mahmoud Ramadan Ali <
cisco.and.more.blog(a)gmail.com> wrote:
Hi Muhammad,
If the users MUST authenticate to Kamailio first,This means that Kamailio
should be aware of the SIP users exist in the Asterisk DB to be able to
authenticate them and NOT receive 401 Unauthorized error message from
Kamailio.
My question now might be simple but it a point of confusion to me and it
is how to tell Kamailio about the SIP users in the Asterisk DB ?!
Best Regards,
On Sun, Nov 16, 2014 at 3:01 PM, Muhammad Shahzad <shaheryarkh(a)gmail.com>
wrote:
This seems to be fine. The user MUST authenticate
to Kamailio, only then
Kamailio will create REGISTER request that is send to asterisk. That's the
key security feature behind the idea.
Look at the register architecture diagram,
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb…
Thank you.
On Sat, Nov 15, 2014 at 10:31 PM, Mahmoud Ramadan Ali <
cisco.and.more.blog(a)gmail.com> wrote:
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Hi Dears,
I'm trying to configure Kamailio as SBC in multi home mode for Asterisk
by authenticating the inbound SIP registration requests,i'm following this
tutorial
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
to achieve this goal. i have modified the necessary changes like the
Asterisk DB URL and the SIP table name and Username and password column and
verified the connection.
My topology like this *Asterisk (192.168.100.10)
<----Internal:192.168.100.1---->Kamailio<---External:192.168.50.1-----> SIP
Phone (192.168.50.2)*
But when trying to register a SIP phone Kamailio does NOT forward the
authentication request to Asterisk and sends 401 Unauthorized error
message.I've attached my config file if any one wants to check it and
thanks in advance.
Best Regards
U 192.168.50.2:37297 -> 192.168.50.1:5060
REGISTER sip:192.168.50.1;transport=UDP SIP/2.0.
Via: SIP/2.0/UDP 192.168.50.2:37297
;branch=z9hG4bK-d8754z-a46e0c7c9d98fe52-1---d8754z-;rport;transport=UDP.
Max-Forwards: 70.
Contact: <sip:1001@192.168.50.2:37297
;rinstance=1d7c44dbcb8a7a2f;transport=UDP>.
To: <sip:1001@192.168.50.1;transport=UDP>.
From: <sip:1001@192.168.50.1;transport=UDP>;tag=1d222e19.
Call-ID: NTc2NDBjMGQ2YWFmZjdmNWI0MzVmN2Y4NzYyODJlMTc..
CSeq: 2 REGISTER.
Expires: 70.
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE.
Supported: replaces, norefersub, extended-refer, timer,
X-cisco-serviceuri.
User-Agent: Z 3.2.21357 r21367.
Authorization: Digest
username="1001",realm="192.168.50.1",nonce="VGfAuFRnv4wMvoTG7wA9tqYD9fgZDe3D",uri="sip:192.168.50.1;transport=UDP",response="8bbd01d879250585eafee4f510689f73",algorithm=MD5.
Allow-Events: presence, kpml.
Content-Length: 0.
#
U 192.168.50.1:5060 -> 192.168.50.2:37297
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP 192.168.50.2:37297
;branch=z9hG4bK-d8754z-a46e0c7c9d98fe52-1---d8754z-;rport=37297;transport=UDP.
To: <sip:1001@192.168.50.1
;transport=UDP>;tag=b27e1a1d33761e85846fc98f5f3a7e58.fe8b.
From: <sip:1001@192.168.50.1;transport=UDP>;tag=1d222e19.
Call-ID: NTc2NDBjMGQ2YWFmZjdmNWI0MzVmN2Y4NzYyODJlMTc..
CSeq: 2 REGISTER.
WWW-Authenticate: Digest realm="192.168.50.1",
nonce="VGfAuFRnv4wMvoTG7wA9tqYD9fgZDe3D".
Server: kamailio (4.1.6 (i386/linux)).
Content-Length: 0.
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