Take a look at http://www.opensips.org/Resources/DocsTutRadius
And be sure first that OpenSIPS (properly configured) is sending the ACC
request to the RADIUS server.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 12/18/2012 03:58 AM, Willian Mazzardo - SYSSVOIP wrote:
Yes... I follow the tutorial in CDR tool website.
There is any way to check if everything is ok?
Thanks
It might be a silly question, but have you configured the accounting
via radius backend ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 12/17/2012 11:49 PM, Willian Mazzardo - SYSSVOIP wrote:
OK ... I have made some tests and now I`m able to use Dialplan module
on Opensips-cp ... and are working good.
Now i`m trying make work CDRTool on this scenario ... but no luck ...
cdrtool daemon is running, freeradius too ... but no data on
radacct201212 table on radius database.
How can I debug cdrtool to see what is going on?
Thanks
Willian Mazzardo
Depto TI - SYSSVOIP
www.syssvoip.com.br <http://www.syssvoip.com.br>
55 3537 2030 <tel:55%203537%202030>
2012/12/17 Bogdan-Andrei Iancu <[email protected]
<mailto:[email protected]>>
Hi Willian,
Assuming that route(3) is doing routing to register subscribers
and route(5) is doing routing to PSTN and inside these routes you
do the t_relay(), I would suggest moving the setflag for
accounting before triggering those routes. The main idea is to
have the setflag done before the call is forwarded to whatever
destination.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 12/17/2012 08:19 PM, Willian Mazzardo - SYSSVOIP wrote:
Hi Bogdan ... sorry for this ...
I've initiated some tests with Opensips ... and almost
everything is working ...
Now, i`m trying do a separate route for internal accounts calls
and PSTN calls.
I`ve this script on INVITE:
if (is_method("INVITE")) {
if(uri=~"^sip:[55910][0-9][0-9][0-9][0-9]@*") {
xlog("Willian: passou por aqui PONTO A PONTO");
route(3);
setflag(1); # do accounting
}else{
xlog("Willian: passou por aqui SAIDA");
rewritehostport("177.126.178.106:5060
<http://177.126.178.106:5060>");
route(5);
setflag(1); # do accounting
}
setflag(1); # do accounting
}
My internal accounts start with 55910XXXX and my PSTN calls are
Country Code + Region Code ... like for Brazil = 555588889999
<tel:555588889999>
Is this INVITE section right?
Thanks.
Willian Mazzardo
Depto TI - SYSSVOIP
www.syssvoip.com.br <http://www.syssvoip.com.br>
55 3537 2030 <tel:55%203537%202030>
2012/12/15 Bogdan-Andrei Iancu <[email protected]
<mailto:[email protected]>>
Hi,
This is a mailing list for opensips project, and we do offer
support and help for opensips. So either you redirect your
question to the right mailing list, either you start using
opensips
Regards,
Bogdan
Sent from Samsung Mobile
Willian Mazzardo - SYSSVOIP <[email protected]
<mailto:[email protected]>> wrote:
Hi all..
I`m a very new user coming from Asterisk, and I want to do
some test with Kamailio billing / cdr my calls.
I have installed CDRTool and Kamailio with a working cfg who
route any call to my SIP Provider.
But, when I do some call and hang up later... the system
doesn't create any log into radacct* tables.
I checked every configuration in /etc/cdrtool/global.inc and
seems to be OK.
I think maybe is an kamailio routing issue, like no flag or
something.
Can anyone help me with this?
Thanks in advice.
Willian Mazzardo
Depto TI - SYSSVOIP
www.syssvoip.com.br <http://www.syssvoip.com.br>
55 3537 2030 <tel:55%203537%202030>
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