[OpenSIPS-Users] Kamailio 3.1 + CDRTool + CallControl

[Bogdan-Andrei Iancu]

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 <bogdan at opensips.org 
>> <mailto:bogdan at opensips.org>>
>>
>>     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 <bogdan at opensips.org
>>>     <mailto:bogdan at opensips.org>>
>>>
>>>         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 <willian at syssvoip.com.br
>>>         <mailto:willian at syssvoip.com.br>> 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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