[OpenSIPS-Users] Kamailio 3.1 + CDRTool + CallControl
Willian Mazzardo - SYSSVOIP
willian at syssvoip.com.br
Mon Dec 17 20:49:00 CET 2012
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
55 3537 2030
2012/12/17 Bogdan-Andrei Iancu <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 Developerhttp://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");
> 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
>
> Is this INVITE section right?
>
> Thanks.
>
>
>
> Willian Mazzardo
> Depto TI - SYSSVOIP
> www.syssvoip.com.br
> 55 3537 2030
>
>
>
> 2012/12/15 Bogdan-Andrei Iancu <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> 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
>> 55 3537 2030
>>
>>
>
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