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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Check reinvite and NAT settings on the line as well as the SIP peers.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>You can use a stun client from inside your network to see what’s going on with the NAT<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p>&nbsp;</o:p></span></p><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Chris Budinick<br><b>Sent:</b> Tuesday, November 13, 2012 4:59 PM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] Sending calls from behind NAT<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p>&nbsp;</o:p></p><div><p class=MsoNormal style='margin-bottom:12.0pt'><span style='color:black'>I'm with Duncan, you need a public IP address, not private. <o:p></o:p></span></p><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:10.0pt;font-family:"Verdana","sans-serif";color:#1F497D'>Chris Budinick</span><span style='color:black'><o:p></o:p></span></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:10.0pt;font-family:"Verdana","sans-serif";color:#1F497D'>Network Technician</span><span style='color:black'><o:p></o:p></span></p><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Verdana","sans-serif";color:#1F497D'>RAINIER CONNECT</span></b><span style='color:black'><br><br><img width=192 height=116 id="_x0000_i1025" src="cid:image001.jpg@01CDC1C1.BE48BFA0"><br><br><o:p></o:p></span></p></div><div class=MsoNormal align=center style='text-align:center'><span style='color:black'><hr size=3 width="100%" align=center id=zwchr></span></div><div><p class=MsoNormal><b><span style='font-family:"Helvetica","sans-serif";color:black'>From: </span></b><span style='font-family:"Helvetica","sans-serif";color:black'>&quot;Duncan Turnbull&quot; &lt;<a href="mailto:duncan@e-simple.co.nz">duncan@e-simple.co.nz</a>&gt;<br><b>To: </b>&quot;Asterisk Users Mailing List - Non-Commercial Discussion&quot; &lt;<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>&gt;<br><b>Sent: </b>Tuesday, November 13, 2012 1:29:28 PM<br><b>Subject: </b>Re: [asterisk-users] Sending calls from behind NAT<br><br><br>On 14/11/2012, at 10:16 AM, bilal ghayyad &lt;<a href="mailto:bilmar_gh@yahoo.com">bilmar_gh@yahoo.com</a>&gt; wrote:<br><br>&gt; Dears;<br>&gt; <br>&gt; It seems my service provider is requesting a complicated settings to allow me to send from behind NAT. <br>&gt; <br>&gt; What they said:<br>&gt; <br>&gt; &quot;It shouldn't matter as long as you are handling the NAT correctly your end. We do not fix NAT so if you're sending internal addresses in your INVITEs or SDP then things will fail but if you're handling it correctly, we shouldn't tell the difference&quot;.<br>&gt; <br>&gt; <br>&gt; Really, I did not understand what exactly they need. But maybe what they need is to see my public IP address without the private IP address (this what I understood if I am right).<br>&gt; <br>&gt; I tried to use the following in the [general] settings in the sip.conf<br>&gt; <br>&gt; localnet=192.168.10.2/255.255.255.254<br>&gt; externadd =196.40.164.239<br>&gt; <br>This should be externip not externadd<br><br>You are still sending them your local address<br><br>Cheers Duncan<br><br><br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</a> --<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br>&nbsp;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; <a href="http://www.asterisk.org/hello">http://www.asterisk.org/hello</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br>&nbsp;&nbsp; <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></span></p></div><p class=MsoNormal><span style='color:black'><o:p>&nbsp;</o:p></span></p></div></div></body></html>