OK ... I have made some tests and now I`m able to use Dialplan module on Opensips-cp ... and are working good.<div><br></div><div>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.</div>
<div><br></div><div>How can I debug cdrtool to see what is going on?</div><div><br></div><div>Thanks</div><div><br></div><div class="gmail_extra"><br clear="all"><div>Willian Mazzardo<br>Depto TI - SYSSVOIP<br><a href="http://www.syssvoip.com.br">www.syssvoip.com.br</a><br>
55 3537 2030</div><br>
<br><br><div class="gmail_quote">2012/12/17 Bogdan-Andrei Iancu <span dir="ltr"><<a href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<u></u>
<div bgcolor="#ffffff" text="#000000">
<tt>Hi Willian,<br>
<br>
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.<br>
<br>
Regards,<br>
</tt>
<pre cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a></pre><div><div class="h5">
<br>
On 12/17/2012 08:19 PM, Willian Mazzardo - SYSSVOIP wrote:
<blockquote type="cite">Hi Bogdan ... sorry for this ...
<div><br>
</div>
<div>I've initiated some tests with Opensips ... and almost
everything is working ...</div>
<div><br>
</div>
<div>Now, i`m trying do a separate route for internal accounts
calls and PSTN calls.</div>
<div><br>
</div>
<div>I`ve this script on INVITE:</div>
<div>
<div><br>
</div>
<div> if (is_method("INVITE")) {</div>
<div><br>
</div>
<div> if(uri=~"^sip:[55910][0-9][0-9][0-9][0-9]@*") {</div>
<div> xlog("Willian: passou por aqui PONTO A PONTO");</div>
<div> route(3);</div>
<div><br>
</div>
<div> setflag(1); # do accounting</div>
<div><br>
</div>
<div> }else{</div>
<div><br>
</div>
<div> xlog("Willian: passou por aqui SAIDA");</div>
<div><br>
</div>
<div> rewritehostport("<a href="http://177.126.178.106:5060" target="_blank">177.126.178.106:5060</a>");</div>
<div> route(5);</div>
<div><br>
</div>
<div> setflag(1); # do accounting</div>
<div><br>
</div>
<div> }</div>
<div><br>
</div>
<div> setflag(1); # do accounting</div>
<div> }</div>
</div>
<div><br>
</div>
<div>My internal accounts start with 55910XXXX and my PSTN calls
are Country Code + Region Code ... like for Brazil =
<a href="tel:555588889999" value="+555588889999" target="_blank">555588889999</a></div>
<div><br>
</div>
<div>Is this INVITE section right?</div>
<div><br>
</div>
<div>Thanks.</div>
<div><br>
</div>
<div><br>
</div>
<div class="gmail_extra"><br clear="all">
<div>Willian Mazzardo<br>
Depto TI - SYSSVOIP<br>
<a href="http://www.syssvoip.com.br" target="_blank">www.syssvoip.com.br</a><br>
<a href="tel:55%203537%202030" value="+555535372030" target="_blank">55 3537 2030</a></div>
<br>
<br>
<br>
<div class="gmail_quote">2012/12/15 Bogdan-Andrei Iancu <span dir="ltr"><<a href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>></span><br>
<blockquote class="gmail_quote" style="margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<div>
<div>Hi,</div>
<div><br>
</div>
<div>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</div>
<div><br>
</div>
<div>Regards,</div>
<div>Bogdan</div>
<div><br>
</div>
<div><br>
</div>
<div>
<div style="font-size:75%;color:rgb(87,87,87)">Sent
from Samsung Mobile</div>
</div>
<div> <br>
Willian Mazzardo - SYSSVOIP <<a href="mailto:willian@syssvoip.com.br" target="_blank">willian@syssvoip.com.br</a>>
wrote:<br>
Hi all..
<div><br>
</div>
<div>I`m a very new user coming from Asterisk, and I
want to do some test with Kamailio billing / cdr my
calls.</div>
<div><br>
</div>
<div>I have installed CDRTool and Kamailio with a
working cfg who route any call to my SIP Provider.</div>
<div><br>
</div>
<div>But, when I do some call and hang up later... the
system doesn't create any log into radacct* tables.</div>
<div><br>
</div>
<div>I checked every configuration in
/etc/cdrtool/global.inc and seems to be OK.</div>
<div><br>
</div>
<div>I think maybe is an kamailio routing issue, like no
flag or something.</div>
<div><br>
</div>
<div>Can anyone help me with this?</div>
<div><br>
</div>
<div>Thanks in advice.</div>
<div><br>
</div>
<div><br clear="all">
<div>Willian Mazzardo<br>
Depto TI - SYSSVOIP<br>
<a href="http://www.syssvoip.com.br" target="_blank">www.syssvoip.com.br</a><br>
<a href="tel:55%203537%202030" value="+555535372030" target="_blank">55 3537 2030</a></div>
<br>
</div>
</div>
</div>
</blockquote>
</div>
<br>
</div>
</blockquote>
</div></div></div>
</blockquote></div><br></div>