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">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 = 555588889999</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">www.syssvoip.com.br</a><br>
55 3537 2030</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:0 0 0 .8ex;border-left:1px #ccc solid;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:#575757">Sent from Samsung Mobile</div></div><div class="im"> <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>