<div dir="ltr"><div><div>Hi Nick ... <br><br></div>I have tried with your modparam set but no look. If I register with dns name, the register not work.<br><br></div><div>What I have? maybe help you canhelp me with what I need ;)<br>
<br></div><div>I have Asterisk to handle registration, routes and billing, but my customer base are increasing, and I want use Opensips to handle REGISTER and route PSTN calls to one of my asterisk box. (load balance) but this is later.<br>
<br></div><div>My customers have Asterisk, Linksys PAP2 adapters and sofpthones, and all of them are registering with my asterisk. Some with IP Address, others with one of my dns names.<br><br></div><div>I want to implement IM, Video calls, and presence to offer to my customers something different from others Voip company.<br>
<br></div><div>I expect use opensips to call SIP > SIP customers directly, without asterisk using rtpproxy when its needed, and PSTN calls routing to asterisk until I found some billing system who I can comprehend and know how to use.<br>
<br></div><div>Thats all ;)<br><br><br></div></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><div class="gmail_quote">2013/7/18 Nick Khamis <span dir="ltr"><<a href="mailto:symack@gmail.com" target="_blank">symack@gmail.com</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">There is also:<div><br><div><ol style="margin:0px;padding:0px 0px 0px 48px;color:rgb(172,172,172);font-family:Consolas,Menlo,Monaco,'Lucida Console','Liberation Mono','DejaVu Sans Mono','Bitstream Vera Sans Mono',monospace,serif;font-size:12px;line-height:21px;background-color:rgb(248,248,248)">
<li><div style="padding:0px 5px;vertical-align:top;color:rgb(0,0,0);border-left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);background-color:rgb(255,255,255)">modparam("auth_db|usrloc|uri", "use_domain", 1)</div>
</li></ol></div></div><div>Please change that to 0. It's been a while since I have dealt with REGISTER authentiacation issues. Are you sure</div><div>you need it? It's quite a resourceful process as the number of clients increase. What we do now, is use:</div>
<div><br></div><div>1) The address table</div><div>2) Dialplan</div><div>3) Dynamic Routing</div><div>4) IPTables</div><div><br></div><div>To enforce who's INVITE gets processed by our servers. No registration required.<br>
</div><div><br></div><div>If you really want to handle REGISTER, I will take a closer look. Until then maybe look at Chapter 5 (Page 90),</div><div>of <a href="ftp://115.146.120.141/voIP/Building%20Telephony%20Systems%20with%20OpenSIPS%201.6.pdf" target="_blank">ftp://115.146.120.141/voIP/Building%20Telephony%20Systems%20with%20OpenSIPS%201.6.pdf</a>. I know the</div>
<div>answer is in there because I dealt with your issue a long time ago.</div><div><br></div><div>Kind Regards,</div><div><br></div><div>Nick</div></div>
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