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    <tt>He Eric,<br>
      <br>
      In the script (generated by menuconfig), you have an IF block with
      :<br>
          append_hf("P-hint: outbound\r\n");<br>
      <br>
      That block is executed if one of your users is dialing an external
      SIP domain. So, that is the place where the calls will break out
      to public internet. <br>
      <br>
      As your OpenSIPS has a private IP (not routable in internet), you
      need to configure it to advertise the public IP of your NAT via
      the set_advertise_address() - see:<br>
         
      <a class="moz-txt-link-freetext" href="http://www.opensips.org/Documentation/Script-CoreFunctions-2-2#toc46">http://www.opensips.org/Documentation/Script-CoreFunctions-2-2#toc46</a><br>
      <br>
      Put there the public IP of your NAT. Also, be sure you create in
      your router (running the NAT) a port forwarding from the public IP
      port 5060 UDP to the private IP of your OpenSIPS port 5060 UDP.<br>
      <br>
      Best regards,<br>
    </tt>
    <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>
    <div class="moz-cite-prefix">On 15.09.2016 13:10, Eric Freeman
      wrote:<br>
    </div>
    <blockquote
cite="mid:DM3PR12MB076490A13985682AA8D2708087F00@DM3PR12MB0764.namprd12.prod.outlook.com"
      type="cite">
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      <div>Yes.  <br>
        <br>
        <div class="acompli_signature">Get <a moz-do-not-send="true"
            href="https://aka.ms/o0ukef">Outlook for iOS</a></div>
        <br>
      </div>
      <hr tabindex="-1" style="display:inline-block; width:98%">
      <div id="divRplyFwdMsg" dir="ltr"><font style="font-size:11pt"
          color="#000000" face="Calibri, sans-serif"><b>From:</b>
          Bogdan-Andrei Iancu <a class="moz-txt-link-rfc2396E" href="mailto:bogdan@opensips.org">&lt;bogdan@opensips.org&gt;</a><br>
          <b>Sent:</b> Thursday, September 15, 2016 4:57:45 AM<br>
          <b>To:</b> Eric Freeman; OpenSIPS users mailling list<br>
          <b>Subject:</b> Re: [OpenSIPS-Users] Routing in opensips</font>
        <div> </div>
      </div>
      <div><tt>Hello Eric,<br>
          <br>
          So, you OpenSIPS is running behind a NAT ?<br>
          <br>
          Regards,<br>
        </tt>
        <pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a moz-do-not-send="true" class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a></pre>
        <div class="moz-cite-prefix">On 14.09.2016 18:43, Eric Freeman
          wrote:<br>
        </div>
        <blockquote type="cite">
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            <p>Thank you. I am using the residential script and my
              device is registered. When I register two devices I can
              call between them. But, when I want to send the traffic
              out to the Internet I receive these types of errors. My
              device is a 10 address trying to call public IPs on the
              Internet. It appears I need to add a default route in
              opensips? It seems like it can't send the traffic to the
              Internet or a default route. The opensips server itself
              can get onto the Internet. </p>
            <p><br>
            </p>
            <p style="margin-right:0px; margin-left:0px; font-size:18px;
              line-height:normal; font-family:Monaco;
              color:rgb(4,51,255)">
              <span style="color:rgb(0,0,0); font-size:11pt;
                font-family:Verdana,Geneva,sans-serif">Aug 22 15:12:55
                cd-ubuntu-opensips /usr/local/sbin/opensips[11973]:
                DBG:core:get_hdr_field</span></p>
            <p style="margin-right:0px; margin-left:0px; font-size:18px;
              line-height:normal; font-family:Monaco;
              color:rgb(4,51,255)">
              <span style="color:rgb(0,0,0); font-size:11pt;
                font-family:Verdana,Geneva,sans-serif">: found end of
                header</span></p>
            <p style="margin-right:0px; margin-left:0px; font-size:18px;
              line-height:normal; font-family:Monaco;
              color:rgb(4,51,255)">
              <span style="color:rgb(0,0,0); font-size:11pt;
                font-family:Verdana,Geneva,sans-serif">Aug 22 15:12:55
                cd-ubuntu-opensips /usr/local/sbin/opensips[11973]:
                DBG:rr:find_first_rout</span></p>
            <p style="margin-right:0px; margin-left:0px; font-size:18px;
              line-height:normal; font-family:Monaco;
              color:rgb(4,51,255)">
              <span style="color:rgb(0,0,0); font-size:11pt;
                font-family:Verdana,Geneva,sans-serif">e: No Route
                headers found</span></p>
            <p style="margin-right:0px; margin-left:0px; font-size:18px;
              line-height:normal; font-family:Monaco;
              color:rgb(4,51,255)">
              <span style="color:rgb(0,0,0); font-size:11pt;
                font-family:Verdana,Geneva,sans-serif">Aug 22 15:12:55
                cd-ubuntu-opensips /usr/local/sbin/opensips[11973]:
                DBG:rr:loose_route: Th</span></p>
            <p style="margin-right:0px; margin-left:0px; font-size:18px;
              line-height:normal; font-family:Monaco;
              color:rgb(4,51,255)">
              <span style="color:rgb(0,0,0); font-size:11pt;
                font-family:Verdana,Geneva,sans-serif">ere is no Route
                HF</span></p>
            <p style="margin-right:0px; margin-left:0px; font-size:18px;
              line-height:normal; font-family:Monaco;
              color:rgb(4,51,255)">
              <span style="color:rgb(0,0,0); font-size:11pt;
                font-family:Verdana,Geneva,sans-serif">Aug 22 15:12:55
                cd-ubuntu-opensips /usr/local/sbin/opensips[11973]:
                DBG:dialog:w_create_di</span></p>
            <p><span style="color:rgb(0,0,0); font-size:11pt;
                font-family:Verdana,Geneva,sans-serif"></span><span
                style="color:rgb(0,0,0); font-size:11pt;
                font-family:Verdana,Geneva,sans-serif"></span><span
                style="color:rgb(0,0,0); font-size:11pt;
                font-family:Verdana,Geneva,sans-serif"></span><span
                style="color:rgb(0,0,0); font-size:11pt;
                font-family:Verdana,Geneva,sans-serif"></span><span
                style="color:rgb(0,0,0); font-size:11pt;
                font-family:Verdana,Geneva,sans-serif"></span><span
                style="color:rgb(0,0,0); font-size:11pt;
                font-family:Verdana,Geneva,sans-serif"></span><span
                style="color:rgb(0,0,0); font-size:11pt;
                font-family:Verdana,Geneva,sans-serif"></span></p>
            <p style="margin-right:0px; margin-left:0px; font-size:18px;
              line-height:normal; font-family:Monaco;
              color:rgb(4,51,255)">
              <span style="color:rgb(0,0,0); font-size:11pt;
                font-family:Verdana,Geneva,sans-serif">alog2: bye on
                timeout activated</span></p>
            <p><br>
            </p>
            <p>Thank you,</p>
            <p><br>
            </p>
            <div id="Signature">
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                  <p>Eric Freeman</p>
                  <p><span id="ms-rterangepaste-start"></span></p>
                  <div style="color:rgb(34,34,34);
                    font-family:arial,sans-serif; font-size:12.8px">
                    <br>
                  </div>
                  <div style="color:rgb(34,34,34);
                    font-family:arial,sans-serif; font-size:12.8px">
                  </div>
                  <span id="ms-rterangepaste-end"></span></div>
              </div>
            </div>
          </div>
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          <div id="divRplyFwdMsg" dir="ltr"><font style="font-size:11pt"
              color="#000000" face="Calibri, sans-serif"><b>From:</b>
              Bogdan-Andrei Iancu
              <a moz-do-not-send="true" class="moz-txt-link-rfc2396E"
                href="mailto:bogdan@opensips.org">&lt;bogdan@opensips.org&gt;</a><br>
              <b>Sent:</b> Wednesday, September 14, 2016 10:47:23 AM<br>
              <b>To:</b> OpenSIPS users mailling list; Eric Freeman<br>
              <b>Subject:</b> Re: [OpenSIPS-Users] Routing in opensips</font>
            <div> </div>
          </div>
          <div><tt>Hi Eric,<br>
              <br>
              A good starting point is the residential script that can
              be generated via the menuconfig tool. See:<br>
                  <a moz-do-not-send="true"
                class="moz-txt-link-freetext"
                href="http://www.opensips.org/Documentation/Generating-Configs-2-2">
http://www.opensips.org/Documentation/Generating-Configs-2-2</a><br>
              <br>
              Then have your unit registered with your OpenSIPS.<br>
              <br>
              Best regards,<br>
            </tt>
            <pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a moz-do-not-send="true" class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a></pre>
            <div class="moz-cite-prefix">On 12.09.2016 18:15, Eric
              Freeman wrote:<br>
            </div>
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                <p>I am very new to opensips and I want to use it for
                  one application. I have installed version: opensips
                  2.2.1 (x86_64/linux) </p>
                <p>on Ubuntu 14.04.2 LTS</p>
                <p>Release: 14.04</p>
                <p><br>
                </p>
                <p>I have a video conference unit and I want to be able
                  make SIP calls to other video conference</p>
                <p>units over the Internet. </p>
                <p><br>
                </p>
                <p>I believe I need to set up some type of routing to
                  route calls from this device out</p>
                <p>to the Internet. I believe I need to add some
                  commands to the route section of the configuration
                  file. Can someone please let me know what commands I
                  need to the config to allow the unit to call other
                  units via sip over the Internet. I am using the
                  “default routing” from the residential script. </p>
                <p><br>
                </p>
                <p><span style="font-size:10pt">I have registered the
                    video conference unit to the opensips server</span><br>
                </p>
                <p><br>
                </p>
                <p><span style="font-size:10pt">Contact:: <a
                      moz-do-not-send="true"
                      class="moz-txt-link-abbreviated"
                      href="mailto:sip:LifeSize@10.89.71.12;transport=UDP">
                      sip:LifeSize@10.89.71.12;transport=UDP</a> Q=</span><br>
                </p>
                <p><br>
                </p>
                <p>from config</p>
                <p>listen=udp:10.88.23.13:5060   # CUSTOMIZE ME</p>
                <p><br>
                </p>
                <p>listen=tcp:10.88.23.13:5060   # CUSTOMIZE ME </p>
                <p><br>
                </p>
                <p><span style="font-size:10pt">Please let me know if
                    you need more information.</span><br>
                </p>
                <p><br>
                </p>
                <p>Thank you,</p>
                <p>Eric Freeman</p>
                <div id="Signature">
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              <pre>_______________________________________________
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<a moz-do-not-send="true" class="moz-txt-link-abbreviated" href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a>
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</pre>
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