[OpenSIPS-Users] Kamailio 3.1 + CDRTool + CallControl

[Bogdan-Andrei Iancu]

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 <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>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20121217/c7e4902a/attachment-0001.htm>