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<tt>Take a look at <a class="moz-txt-link-freetext" href="http://www.opensips.org/Resources/DocsTutRadius">http://www.opensips.org/Resources/DocsTutRadius</a><br>
<br>
And be sure first that OpenSIPS (properly configured) is sending
the ACC request to the RADIUS server.<br>
<br>
Regards,</tt><br>
<pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a></pre>
<br>
On 12/18/2012 03:58 AM, Willian Mazzardo - SYSSVOIP wrote:
<blockquote
cite="mid:CAL2ykgH-hkBA-ABwkLBgyFH8bkpGuLNAyZLcxD3UKo=at5nCDA@mail.gmail.com"
type="cite">
<p dir="ltr">Yes... I follow the tutorial in CDR tool website.</p>
<p dir="ltr">There is any way to check if everything is ok?<br>
</p>
<p dir="ltr">Thanks<br>
</p>
<div class="gmail_quot<blockquote class=" style="margin: 0pt
0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204);
padding-left: 1ex;">
<div bgcolor="#ffffff" text="#000000"> <tt>It might be a silly
question, but have you configured the accounting via radius
backend ?<br>
<br>
Regards,</tt><br>
<pre cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a moz-do-not-send="true" href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a></pre>
<br>
On 12/17/2012 11:49 PM, Willian Mazzardo - SYSSVOIP wrote:
<blockquote type="cite">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 moz-do-not-send="true"
href="http://www.syssvoip.com.br" target="_blank">www.syssvoip.com.br</a><br>
<a moz-do-not-send="true" href="tel:55%203537%202030"
value="+555535372030" target="_blank">55 3537 2030</a></div>
<br>
<br>
<br>
<div class="gmail_quote">2012/12/17 Bogdan-Andrei Iancu <span
dir="ltr"><<a moz-do-not-send="true"
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 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 moz-do-not-send="true" href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a></pre>
<div>
<div> <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
moz-do-not-send="true"
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
moz-do-not-send="true"
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 moz-do-not-send="true"
href="http://www.syssvoip.com.br"
target="_blank">www.syssvoip.com.br</a><br>
<a moz-do-not-send="true"
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
moz-do-not-send="true"
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
moz-do-not-send="true"
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 moz-do-not-send="true"
href="http://www.syssvoip.com.br"
target="_blank">www.syssvoip.com.br</a><br>
<a moz-do-not-send="true"
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>
</blockquote>
</div>
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