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    <tt>It might be a silly question, but have you configured the
      accounting via radius backend ?<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 12/17/2012 11:49 PM, Willian Mazzardo - SYSSVOIP wrote:
    <blockquote
cite="mid:CAL2ykgHiDyD-oRN13XD_ftKQ---jwGvZMSeL4EURajp4YL+dbQ@mail.gmail.com"
      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">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">&lt;<a moz-do-not-send="true"
              href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>&gt;</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 class="h5"> <br>
                  On 12/17/2012 08:19 PM, Willian Mazzardo - SYSSVOIP
                  wrote:
                  <blockquote type="cite">Hi Bogdan ... sorry for this
                    ...&nbsp;
                    <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>&nbsp; &nbsp;if (is_method("INVITE")) {</div>
                      <div><br>
                      </div>
                      <div>&nbsp; &nbsp; &nbsp; &nbsp;
                        if(uri=~"^sip:[55910][0-9][0-9][0-9][0-9]@*") {</div>
                      <div>&nbsp; &nbsp; &nbsp; &nbsp; xlog("Willian: passou por aqui PONTO
                        A PONTO");</div>
                      <div>&nbsp; &nbsp; &nbsp; &nbsp; route(3);</div>
                      <div><br>
                      </div>
                      <div>&nbsp; &nbsp; &nbsp; &nbsp; setflag(1); # do accounting</div>
                      <div><br>
                      </div>
                      <div>&nbsp; &nbsp; &nbsp; &nbsp; }else{</div>
                      <div><br>
                      </div>
                      <div>&nbsp; &nbsp; &nbsp; &nbsp; xlog("Willian: passou por aqui
                        SAIDA");</div>
                      <div><br>
                      </div>
                      <div>&nbsp; &nbsp; &nbsp; &nbsp; rewritehostport("<a
                          moz-do-not-send="true"
                          href="http://177.126.178.106:5060"
                          target="_blank">177.126.178.106:5060</a>");</div>
                      <div>&nbsp; &nbsp; &nbsp; &nbsp; route(5);</div>
                      <div><br>
                      </div>
                      <div>&nbsp; &nbsp; &nbsp; &nbsp; setflag(1); # do accounting</div>
                      <div><br>
                      </div>
                      <div>&nbsp; &nbsp; &nbsp; &nbsp; }</div>
                      <div><br>
                      </div>
                      <div>&nbsp; &nbsp; &nbsp; &nbsp; setflag(1); # do accounting</div>
                      <div>&nbsp; &nbsp; &nbsp; &nbsp; }</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">&lt;<a
                            moz-do-not-send="true"
                            href="mailto:bogdan@opensips.org"
                            target="_blank">bogdan@opensips.org</a>&gt;</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 &lt;<a
                                moz-do-not-send="true"
                                href="mailto:willian@syssvoip.com.br"
                                target="_blank">willian@syssvoip.com.br</a>&gt;

                              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>
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