[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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.opensips.org/pipermail/users/attachments/20100323/51efb22d/attachment-0001.htm
More information about the Users
mailing list