[OpenSIPS-Users] Users Digest, Vol 20, Issue 85

Ahmed Munir ahmedmunir007 at gmail.com
Tue Mar 23 09:14:47 CET 2010


Hi Bogdan,

Thanks for your reply. As you suggested about check_source_address()
function, I get its return value using $avp(i:checksrc) as listed down
below;

        $avp(s:checksrc) = check_source_address("0");

 log("#################################################################################\n");
        xlog("Check Source Address from Address TABLE Where Value 1 is Equal
to True: $(avp(s:checksrc))\n");

 log("#################################################################################\n");

        if($avp(s:checksrc)!=1)
        {
               if(is_method("INVITE"))
               {
                       log("#################### CHECK SOURCE ADDRESS
######################");
                       route(1);
                       setflag(1);
               }
        }
        else
        {
               t_reply("403","Forbidden");
               exit;
        }

But the problem I'm facing is when I enlist IP in address table i.e.
11.22.33.44, call is rejected when else condition is used, when else
condition is commented call is made. But on other hand when I remove the IP
as mentioned from address table, it should reject the call (commenting else
condition), unfortunately the call is made.

Kindly assist me how can I permit or deny calls on IP bases, when user is
not registered from OpenSIPS but sending calls from GW to OpenSIPs?


Date: Mon, 22 Mar 2010 00:09:43 +0200
> From: Bogdan-Andrei Iancu <bogdan at voice-system.ro>
> Subject: Re: [OpenSIPS-Users] Check Live Peers on OpenSIPS
> To: OpenSIPS users mailling list <users at lists.opensips.org>
> Message-ID: <4BA69927.2050102 at voice-system.ro>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Hi Ahmed
>
> Ahmed Munir wrote:
> > Hi Bogdan,
> >
> > Thanks for your suggestion, few things I want to ask from you;
> >
> > 1- Can I use rewritehostport(); function instead of $rd='11.22.33.44'
> > and append it to t_relay()? Like;
> >
> > setflag(2);
> > rewritehostport("11.22.33.55:5060 <http://203.215.179.34:5060>");
> > t_relay();
> > route(1);
> > exit;
>
> Yes, that is correct.
> >
> > 2- When using check_source_address() function of permissions module,
> > I'm facing weird problem. On machine A I've installed OpenSIPS ver
> > 1.6.1 svn one, I used this function to permitted certain source IPs as
> > I listed in address table. On machine B (currently working on it using
> > Radius) I've installed same version of OpenSIPS as on machine A, when
> > I call its check_source_address() function in INVITE section, it is
> > working as it worked on machine A. Machine A settings are listed below;
> >
> >
> > if(is_method("INVITE") && check_source_address("0"))
> > {
> >        log("#################### CHECK SOURCE ADDRESS
> > ######################");
> >        route(1);
> >        setflag(1);
> > }
> >
> >
> > Machine B description I'm mentioning below;
> >
> > 2-1- If user registered him/her self on SIP phone their source IP not
> > going to be checked, and make calls to each other.
> > 2-2- If user A is on GW calls user B who is located and Registered on
> >  OpenSIPS, user A GW's source IP must be checked by OpenSIPs, if the
> > IP exists on address table, call is permitted if not deny the call.
> >
> > Problems;
> >
> > When I user A and user B registered on OpenSIPs (using Radius) they
> > can call each other, but if a user A calling from GW to user B who is
> > registered on OpenSIPs, calls is made even the address is not listed
> > on address table. And also in logs I see that that permissions module
> > shows that it doesn't find any IP enlisted in its hash table, but
> > still permitting it.
> The function just checks if the source IP is in the table, but does not
> take any action - you need to so this manually from the script, based on
> the return code (true or false) of the function.
>
> Regards,
> Bogdan
> > The configuration of machine B is listed below;
> >
> > [........]
> >
> > Kindly assist me, how can I permit or deny user from source IP ?
> > Because on machine A, check_source_address() function is working
> > perfectly but I haven't integrated FreeRadius with OpenSIPs. Please
> > sort out my problem as your earliest.
> >
> >
> >
> >
> >     Date: Thu, 18 Mar 2010 18:38:29 +0200
> >     From: Bogdan-Andrei Iancu <bogdan at voice-system.ro
> >     <mailto:bogdan at voice-system.ro>>
> >     Subject: Re: [OpenSIPS-Users] Check Live Peers on OpenSIPS
> >     To: OpenSIPS users mailling list <users at lists.opensips.org
> >     <mailto:users at lists.opensips.org>>
> >     Message-ID: <4BA25705.10506 at voice-system.ro
> >     <mailto:4BA25705.10506 at voice-system.ro>>
> >     Content-Type: text/plain; charset=ISO-8859-1; format=flowed
> >
> >     Hi Ahmed,
> >
> >     Ahmed Munir wrote:
> >     > Hi Bogdan,
> >     >
> >     > Thanks for reply. I forgot to mention earlier that for I'm using
> >     > OpenSIPS + FreeRadius, where radius is doing accounting and
> >     > authentication. I used aaa_does_uri_exist() function as well, but
> >     > seems not working or making mistake while implementing it. On other
> >     > hand using lookup("location",m) function, on retcode = -1, I
> >     > redirected the INVITE to GW, using Dispatcher.  But though
> >     thanks for
> >     > your suggestion and I'll consider it.
> >     >
> >     > Few things I want to ask you, as I listed below;
> >     > 1-How can I forward SIP INVITE request to other SIP machine in
> state
> >     > full manner ?
> >     simply do:
> >        # set new destination in RURI
> >        $rd= "11.22.33.44";
> >        # send it out in stateful mode
> >        t_relay();
> >        exit;
> >
> >     > 2- While accounting using radius, when user A (registered on
> >     OpenSIPS)
> >     > calls the user B who is located at GW side, accounting doesn't take
> >     > place.  On the other hand when user B (from GW) calls user A (to
> >     > OpenSIPS), accounting take place. I want to know its cause?
> >     Because I
> >     > want its accounting on both sides.
> >     take care and check where you set in script the acc flag - maybe
> >     you are
> >     setting it only if lookup is successful.
> >
> >     Regards,
> >     Bogdan
> >     >
> >     > Kindly advise me at your earliest.
> >     >
> >     >
> >     >     ------------------------------
> >     >
> >     >     Message: 6
> >     >     Date: Thu, 18 Mar 2010 10:23:27 +0200
> >     >     From: Bogdan-Andrei Iancu <bogdan at voice-system.ro
> >     <mailto:bogdan at voice-system.ro>
> >     >     <mailto:bogdan at voice-system.ro <mailto:bogdan at voice-system.ro
> >>>
> >     >     Subject: Re: [OpenSIPS-Users] Check Live Peers on OpenSIPS
> >     >     To: OpenSIPS users mailling list <users at lists.opensips.org
> >     <mailto:users at lists.opensips.org>
> >     >     <mailto:users at lists.opensips.org
> >     <mailto:users at lists.opensips.org>>>
> >     >     Message-ID: <4BA1E2FF.3060702 at voice-system.ro
> >     <mailto:4BA1E2FF.3060702 at voice-system.ro>
> >     >     <mailto:4BA1E2FF.3060702 at voice-system.ro
> >     <mailto:4BA1E2FF.3060702 at voice-system.ro>>>
> >     >     Content-Type: text/plain; charset=ISO-8859-1; format=flowed
> >     >
> >     >     Hi Ahmed,
> >     >
> >     >     if the destination number (called number) is not a local
> >     subscriber (a
> >     >     SIP user), you simply route the call to a PSTN GW (you do this
> >     >     re-route
> >     >     from the script)
> >     >
> >     >     To check if a user is a local subscriber, you can either check
> a
> >     >     pattern
> >     >     (like all my local users are alphanumeric, or all starts
> >     with 3345*,
> >     >     etc), either simply check if the user does exists in the
> >     subscriber
> >     >     table (see the URI module, the db_does_uri_exists() function:
> >     >
> >      http://www.opensips.org/html/docs/modules/1.6.x/uri.html#id271131
> >     >
> >     >     Regards,
> >     >     Bogdan
> >     >
> >     >     Ahmed Munir wrote:
> >     >     > Hi,
> >     >     >
> >     >     > I want to know how can I check the peers of source and
> >     destination
> >     >     > phones? Like if both phones are located (registered) on one
> >     >     > UAS(OpenSIPS) can call SIP-SIP, if any one phone is
> registered
> >     >     on UAS
> >     >     > and other is on PSTN, call will be re-routed to SIP-PSTN.
> >     In case of
> >     >     > SIP-SIP, lookup("location") function works and I need to know
> >     >     how can
> >     >     > I forward call to SIP-PSTN ?
> >     >     >
> >     >     > Kindly advise me the method/ function can used for it.
> >     >     >
> >     >     > --
> >     >     > Regards,
> >     >     >
> >     >     > Ahmed Munir
> >     >     >
> >     >     >
> >     >     >
> >     >
> >
> ------------------------------------------------------------------------
> >     >     >
> >     >     > _______________________________________________
> >     >     > Users mailing list
> >     >     > Users at lists.opensips.org <mailto:Users at lists.opensips.org>
> >     <mailto:Users at lists.opensips.org <mailto:Users at lists.opensips.org>>
> >     >     > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >     >     >
> >     >
> >     >
> >     >     --
> >     >     Bogdan-Andrei Iancu
> >     >     www.voice-system.ro <http://www.voice-system.ro>
> >     <http://www.voice-system.ro>
> >     >
> >     >
> >     >
> >     >
> >     > --
> >     > Regards,
> >     >
> >     > Ahmed Munir
> >     >
> >     >
> >     >
> >
> ------------------------------------------------------------------------
> >     >
> >     > _______________________________________________
> >     > Users mailing list
> >     > Users at lists.opensips.org <mailto:Users at lists.opensips.org>
> >     > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >     >
> >
> >
> >     --
> >     Bogdan-Andrei Iancu
> >     www.voice-system.ro <http://www.voice-system.ro>
> >
> >
> >
> >
> >
> > --
> > Regards,
> >
> > Ahmed Munir
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
>
> --
> Bogdan-Andrei Iancu
> www.voice-system.ro
>
>
>
>
-- 
Regards,

Ahmed Munir
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