[OpenSIPS-Users] Kamailio 3.1 + CDRTool + CallControl

Bogdan-Andrei Iancu bogdan at opensips.org
Mon Dec 17 21:30:03 CET 2012


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
>
>
>
> 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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