Re: [OpenSIPS-Users] How to send SIP header 302 registration request to Asterisk

2009-12-18 Thread Muhammad Zulqarnain
Hello Ahmed!

Use rewritehost() and send 302 reply to user, I am using permission module 
and redirecting to zz.zz.zz.zz. ip if authenticated by permission module.

if (check_source_address(1,$avp(i:9))) {
xlog(L_INFO,GW sends INVITE and was found in DB: $avp(i:9)\n);
xlog(L_INFO,from user- $ru\n);

rewritehost(zz.zz.zz.zz);
sl_send_reply(302, Redirect);
exit;
}


Thanks
Regards
Muhammad Zulqarnain



Re: [OpenSIPS-Users] How to send SIP header 302 registration request to Asterisk
Friday, December 18, 2009 2:22:26 PM
From:
Ahmed Munir ahmedmunir...@gmail.com
To:
 users@lists.opensips.org
Hi,

Thanks for your reply, I'm attaching the file named trace. Where OpenSIPs IP 
is: yy.yy.yy.yy, Asterisk IP: zz.zz.zz.zz and my UAC IP: xx.xx.xx.xx.

As you can see I'm getting unauthourized error from Asterisk side, even using 
these credentials I can get registered on my Asterisk machine

Previously I forgot to ask, how can I set SIP header 302 on registeration 
section in OpenSIPs?

Kindly advise me.



 

Date: Thu, 17 Dec 2009 22:40:15 -0800
From: Jai Rangi jpra...@gmail.com
Subject: Re: [OpenSIPS-Users] How to send SIP header 302 registration
   request to Asterisk
To: OpenSIPS users mailling list users@lists.opensips.org
Message-ID:
   eb007ec0912172240hc779cbflbf3460b835f1f...@mail.gmail.com
Content-Type: text/plain; charset=iso-8859-1

Getting some ngrep traces will help some other to help you. Unauthorized
message is for useraccount or for opensip.
-Jai


On Thu, Dec 17, 2009 at 10:25 PM, Ahmed Munir 
ahmedmunir...@gmail.comwrote:

 Hi,

 I'm using OpenSIPs version 1.6, the module I'm using is dispatcher using
 mysql. My question is how can I send SIP header 302 registration request 
to
 Asterisk? Because Asterisk is sending me unAuthorized message to OpenSIPs.
 Even the credentials I'm using for Asterisk is the same as I'm using for
 OpenSIPs.


 Kindly advise me to resolve this issue.

 --
 Regards,

 Ahmed Munir



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Regards,

Ahmed Munir


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Re: [OpenSIPS-Users] IP Based Authentication on Asterisk with OpenSIP (Bogdan-Andrei Iancu)

2009-12-12 Thread Muhammad Zulqarnain
Hello!

Thanks Bodgan! 

I have added following to sip header which add source IP in sip packet before 
sending to asterisk.
append_hf(X-customer-ip: $si\r\n);
Sip trace got following packet which add X-customer-ip: x.x.x.x to sip header 
but I am wondering how to told/setup asterisk to read source IP from additional 
sip header instead of network level. Also I have another switch named 
Voipswitch on which I would not be able to customize any thing to honor 
incoming sip header for source ip validation so solution should work for all 
GW/Server other then asterisk only.

U 2009/12/12 18:58:32.829652 y.y.y.y:5060 - z.z.z.z:5060
INVITE sip:123456...@y.y.y.y SIP/2.0.
Record-Route: sip:y.y.y.y;lr;ftag=a16f2b41;did=882.11d0f5f.
To: sip:123456...@y.y.y.y.
From: hellosip:he...@y.y.y.y;tag=a16f2b41.
Via: SIP/2.0/UDP y.y.y.y;branch=z9hG4bKfb2e.a79595a4.0.
Via: SIP/2.0/UDP 
x.x.x.x:7266;received=x.x.x.x;branch=z9hG4bK-d87543-197592634-1--d87543-;rport=7266.
Call-ID: 9867b13bdc3a4f30.
CSeq: 1 INVITE.
Contact: sip:he...@x.x.x.x:7266.
Max-Forwards: 69.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, 
INFO.
Content-Type: application/sdp.
User-Agent: eyeBeam release 3007n stamp 17816.
Content-Length: 237.
X-customer-ip: x.x.x.x.

Please suggest solution with sample configs.

Thanks
Regards
Muhammad Zulqarnain



Hi Muhammad,

What you can do (and I know for sure it works with
Asterisk), is to configure opensips (LB) to add the real src IP (IP of
the client) into a SIP header (in the request) and send it to Asterisk
- asterisk will do IP based auth but will take IP from that header
(custom name) instead of taking it from network level.

Regards,
Bogdan

Muhammad Zulqarnain wrote:
 Hello!
 

This is Zulqarnain and my first post to this mailing list. I just start
using OpenSIP and trying to get myself familiar with OpenSIP and it's
config files, method etc.
 
 Well, I am trying to achieve following:
 Client IP: x.x.x.x
 OpenSIP: y.y.y.y
 Asterisk: z.z.z.z
 
 1) Client - OpenSIP (Load_balancer) - Asterisk1, Asterisk2
 
  a) If Client Request is Register - Relay to Asterisk (It is
working fine as peer is defined with user/passwd auth, host=dynamic)
 
  b) If Client Request is INVITE - Relay to Asterisk (Asterisk
Reject as peer is defined as IP Auth, type=peer, host=x.x.x.x)
 

b scenario failed as Asterisk read OpenSIP IP y.y.y.y and reject the
call as Asterisk is expecting Source IP x.x.x.x instead of y.y.y.y
 
 I tried following to fix but no success:
 

1) 302 Redirect for GW Clients, who can only authenticate by IP and
don't support registration with user/passwd. But my DID Provider don't
support 302 redirect although i have tested with Eyebeam Client in peer
to peer mode and it works. But as it didn't work with my provider so I
tried another way:
 
 2) uac_replace_from 
uac_replace_to function of uac.so module, I tried to replace from/to
field with client IP but still Invite is from OpenSIP IP y.y.y.y and
again Asterisk/Voipswitch (another switch) reject call.
 

If required I can post my config file as well here or pastebin for your
expert reviews. I would like to know if it is possible to achive above
scenario with OpenSIP, if Yes then how, any example would appreciated.
 

Another small problem is that while setting up 302 redirect I wanted to
pass $du value to rewritehost() as $du have the address return by
load_balancer of destination IP and rewritehost() only accept string.
Looking for suggestion on this as well.
 
 Hopefully to receive positive reply by community.
 
 Thanks
 Regards,
 Muhammad Zulqarnain
 MSN: zulqarn...@globalitvision.com
 
 
 
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-- Bogdan-Andrei Iancu
www.voice-system.ro



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[OpenSIPS-Users] IP Based Authentication on Asterisk with OpenSIP

2009-12-11 Thread Muhammad Zulqarnain
Hello!

This is Zulqarnain and my first post to this mailing list. I just start using 
OpenSIP and trying to get myself familiar with OpenSIP and it's config files, 
method etc.

Well, I am trying to achieve following:
Client IP: x.x.x.x
OpenSIP: y.y.y.y
Asterisk: z.z.z.z

1) Client - OpenSIP (Load_balancer) - Asterisk1, Asterisk2
   a) If Client Request is Register - Relay to Asterisk (It is working fine as 
peer is defined with user/passwd auth, host=dynamic)
   b) If Client Request is INVITE - Relay to Asterisk (Asterisk Reject as peer 
is defined as IP Auth, type=peer, host=x.x.x.x)

b scenario failed as Asterisk read OpenSIP IP y.y.y.y and reject the call as 
Asterisk is expecting Source IP x.x.x.x instead of y.y.y.y

I tried following to fix but no success:

1) 302 Redirect for GW Clients, who can only authenticate by IP and don't 
support registration with user/passwd. But my DID Provider don't support 302 
redirect although i have tested with Eyebeam Client in peer to peer mode and it 
works. But as it didn't work with my provider so I tried another way:

2) uac_replace_from  uac_replace_to function of uac.so module, I tried to 
replace from/to field with client IP but still Invite is from OpenSIP IP 
y.y.y.y and again Asterisk/Voipswitch (another switch) reject call. 

If required I can post my config file as well here or pastebin for your expert 
reviews. I would like to know if it is possible to achive above scenario with 
OpenSIP, if Yes then how, any example would appreciated.

Another small problem is that while setting up 302 redirect I wanted to pass 
$du value to rewritehost() as $du have the address return by load_balancer of 
destination IP and rewritehost() only accept string. Looking for suggestion on 
this as well.

Hopefully to receive positive reply by community.

Thanks
Regards,
Muhammad Zulqarnain
MSN: zulqarn...@globalitvision.com



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