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

Is this INVITE section right?

Thanks.



Willian Mazzardo
Depto TI - SYSSVOIP
www.syssvoip.com.br <http://www.syssvoip.com.br>
55 3537 2030



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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