Re: [OpenSIPS-Users] Check Live Peers on OpenSIPS
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 <mailto:bog...@voice-system.ro>>
> Subject: Re: [OpenSIPS-Users] Check Live Peers on OpenSIPS
> To: OpenSIPS users mailling list <mailto:users@lists.opensips.org>>
> Message-ID: <4ba25705.10...@voice-system.ro
> <mailto:4ba25705.10...@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
&g
Re: [OpenSIPS-Users] Check Live Peers on OpenSIPS
ou want to enable presence server
## and comment the next 'if' block
## NOTE: uncomment also the definition of route[2] from below
##if( is_method("PUBLISH|SUBSCRIBE"))
## route(2);
if (is_method("PUBLISH"))
{
sl_send_reply("503", "Service Unavailable");
exit;
}
if (is_method("REGISTER"))
{
route(2);
}
if ($rU==NULL) {
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}
# apply DB based aliases (uncomment to enable)
##alias_db_lookup("dbaliases");
# do lookup with method filtering
if (!lookup("location","m")) {
switch ($retcode) {
case -1:
log("# LOOKUP LOCATION FLAG -1
PASS ###");
setflag(2);
rewritehostport("11.22.33.44:5060");
log("### CALL ROUTING TO ROUTE 1
###");
route(1);
exit;
case -3:
log("# LOOKUP LOCATION FLAG -3
PASS ###");
t_newtran();
t_reply("404", "Not Found");
exit;
case -2:
log("# LOOKUP LOCATION FLAG -2
PASS ###");
sl_send_reply("405", "Method Not Allowed");
exit;
}
}
# when routing via usrloc, log the missed calls also
setflag(2);
log(" LOOKUP LOCATION FLAG 1 PASS ");
route(1);
}
route[1] {
# for INVITEs enable some additional helper routes
#if (is_method("INVITE") && check_source_address("0")) {
if (is_method("INVITE")) {
log("INVITE ROUTE 1
Function");
t_on_branch("2");
t_on_reply("2");
t_on_failure("1");
#ds_select_dst("1","4");
#forward();
}
if (!t_relay()) {
sl_reply_error();
};
exit;
}
route[2]
{
log("## AAA-REGISTRATION #");
if (!aaa_www_authorize("rose.abc.com"))
{
www_challenge("rose.abc.com", "1");
return;
}
if (!save("location"))
sl_reply_error();
exit;
}
branch_route[2] {
xlog("new branch at $ru\n");
}
onreply_route[2] {
xlog("incoming reply\n");
}
failure_route[1] {
if (t_was_cancelled()) {
exit;
}
}
Kindly assist me, how can I permit or deny user from source IP ? Because on
machine A, check_source_address() function is working perfectly but I
haven't integrated FreeRadius with OpenSIPs. Please sort out my problem as
your earliest.
> Date: Thu, 18 Mar 2010 18:38:29 +0200
> From: Bogdan-Andrei Iancu
> Subject: Re: [OpenSIPS-Users] Check Live Peers on OpenSIPS
> To: OpenSIPS users mailling list
> Message-ID: <4ba25705.10...@voice-system.ro>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Hi Ahmed,
>
> Ahmed Munir wrote:
> > Hi Bogdan,
> >
> > Thanks for reply. I forgot to mention earlier that for I'm using
> > OpenSIPS + FreeRadius, where radius is doing accounting and
> > authentication. I used aaa_does_uri_exist() function as well, but
> > seems not working or making mistake while implementing it. On other
> > hand using lookup("location",m) function, on retcode = -1, I
> > redirected the INVITE to GW, using Dispatcher. But though thanks for
> > your suggestion and I'll consider it.
> >
> > Few things I want to ask you, as I listed below;
> > 1-How can I forward SIP INVITE request to other SIP machine in state
> > full manner ?
> simply do:
># set new destination in RURI
>$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 G
Re: [OpenSIPS-Users] Check Live Peers on OpenSIPS
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 <mailto:bog...@voice-system.ro>>
> Subject: Re: [OpenSIPS-Users] Check Live Peers on OpenSIPS
> To: OpenSIPS users mailling list <mailto:users@lists.opensips.org>>
> Message-ID: <4ba1e2ff.3060...@voice-system.ro
> <mailto:4ba1e2ff.3060...@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@lists.opensips.org <mailto:Users@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@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
--
Bogdan-Andrei Iancu
www.voice-system.ro
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Check Live Peers on OpenSIPS
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 ?
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.
Kindly advise me at your earliest.
> --
>
> Message: 6
> Date: Thu, 18 Mar 2010 10:23:27 +0200
> From: Bogdan-Andrei Iancu
> Subject: Re: [OpenSIPS-Users] Check Live Peers on OpenSIPS
> To: OpenSIPS users mailling list
> Message-ID: <4ba1e2ff.3060...@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@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
>
> --
> Bogdan-Andrei Iancu
> www.voice-system.ro
>
>
>
>
> --
Regards,
Ahmed Munir
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Check Live Peers on OpenSIPS
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@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
--
Bogdan-Andrei Iancu
www.voice-system.ro
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Check Live Peers on OpenSIPS
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@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
