<div dir="ltr"><div class="gmail_extra">set opensips peer to insecure=port,invite</div><div class="gmail_extra"><br><br><div class="gmail_quote">On Wed, Jul 17, 2013 at 1:12 PM, Willian Mazzardo - SYSSVOIP <span dir="ltr"><<a href="mailto:willian@syssvoip.com.br" target="_blank">willian@syssvoip.com.br</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hi Stephens... how do I do this?<br></div><div class="gmail_extra"><div class="im"><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>55 3537 2030</div>
<br><br></div><div><div class="h5"><div class="gmail_quote">2013/7/17 Stephen Vigus <span dir="ltr"><<a href="mailto:svigus@gmail.com" target="_blank">svigus@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">Hi Willian<div><br></div><div>You most likely need to configure Asterisk to not authenticate SIP requests coming from Opensips.</div><div><br></div><div>Regards</div><div>Stephen</div>
<div><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote"><div><div>On Wed, Jul 17, 2013 at 3:32 AM, Willian Mazzardo - SYSSVOIP <span dir="ltr"><<a href="mailto:willian@syssvoip.com.br" target="_blank">willian@syssvoip.com.br</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div><div dir="ltr"><div><div>Hi all..<br><br></div>I know this is a very simple scenario, all PSTN calls be routed to asterisk to do the billing job, but im having some problems, this is my scenario:<br>
<br></div><div>Sip Client (10.0.0.3) > Opensips (10.1.1.2) > Asterisk (10.1.1.247) ..... > PSTN<br>
</div><div><div><div><div><br>Calls between sip clients on Opensips are working, but when I try to call over Asterisk, I have Proxy authentication problem.<br><br>Here is my logs:<br><br>Opensips: <a href="http://pastebin.com/SWpuRHku" target="_blank">http://pastebin.com/SWpuRHku</a><br>
Asterisk: <a href="http://pastebin.com/6jp50LSS" target="_blank">http://pastebin.com/6jp50LSS</a><br><br>[opensips]<br>host=10.1.1.2<br>type=friend<br>context=callingcard<br>qualify=no<br>insecure=very<br>fromdomain=10.1.1.2<br>
<br><br>
Route: <a href="http://pastebin.com/mLgpXiNx" target="_blank">http://pastebin.com/mLgpXiNx</a><br>
<br>Can someone help me on this?<br><br>Thanks<br><br><br>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>
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<br></blockquote></div><br><br clear="all"><div><br></div>-- <br>Dani Popa
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