[OpenSIPS-Users] Check Live Peers on OpenSIPS

Bogdan-Andrei Iancu bogdan at voice-system.ro
Sun Mar 21 23:09:43 CET 2010


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("203.215.179.34:5060 <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




More information about the Users mailing list