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    <div class="moz-cite-prefix">On 1/21/14, 4:38 PM, David Cunningham
      wrote:<br>
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cite="mid:CAHGbP-8sNup5s4LvbGxHmQYX6-_yebNroNo4DntLgDb6y_0y=g@mail.gmail.com"
      type="cite">
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        <div>Hi Andres,<br>
          <br>
        </div>
        Thanks for the idea. We did send bindaddr to the VPN address and
        restarted Asterisk, but unfortunately that didn't solve the
        issue. Asterisk didn't complain, but still the "sip set debug
        on" didn't show the packets.<br>
        <br>
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    Ok, that is progress though.  At this point we know that the OS is
    receiving the packet and Asterisk is listening on that interface and
    port.  I know you already removed the firewall so that would not be
    the issue.  My next guess is Asterisk is looking at the packet and
    dropping it because it believes it is not meant for it (Kamalio
    config issue), so try a simple test to confirm this.  Just configure
    a remote IP phone/softphone via that same VPN interface to simulate
    a remote SIP endpoint.  If the SIP phone works fine but Kamalio does
    not, that will clearly tell you where the problem lies.<br>
    <br>
    <br>
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cite="mid:CAHGbP-8sNup5s4LvbGxHmQYX6-_yebNroNo4DntLgDb6y_0y=g@mail.gmail.com"
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      <div class="gmail_extra"><br>
        <br>
        <div class="gmail_quote">On 22 January 2014 01:40, Andres <span
            dir="ltr"><<a moz-do-not-send="true"
              href="mailto:andres@telesip.net" target="_blank">andres@telesip.net</a>></span>
          wrote:<br>
          <blockquote class="gmail_quote" style="margin:0 0 0
            .8ex;border-left:1px #ccc solid;padding-left:1ex">
            David,<br>
            <br>
            It seems to me that Asterisk is not seeing/binding to your
            VPN interface.  You need to debug that first.  I would set
            en explicit bind statement in sip.conf to the VPN interface
            address and nothing else.  Then start your asterisk and
            watch the log messages.  It should confirm that it cannot
            bind to that address. If it does bind, then try your test
            again and asterisk should see the SIP packets coming in.<span
              class="HOEnZb"><font color="#888888"><br>
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          </blockquote>
        </div>
        <br>
        <br clear="all">
        <br>
        -- <br>
        David Cunningham, Voisonics<br>
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    <br>
    <br>
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