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    <div class="moz-cite-prefix">Try to set the local port of the SIP
      device to something else than 5060 (15060). Routers with NAT often
      change the outgoing SIP port during NAT (e.g. if 5060 is already
      used by another device). Sometimes (PfSense) they are even
      puzzling the port by default for security reasons, which cause
      similar problems as you describe.<br>
      <br>
      Best regards<br>
      Peter<br>
      <br>
      On 01/15/18 11:46, Alex Lake wrote:<br>
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      cite="mid:44c7e6c4-683e-913d-1932-181f0c64dbb4@digitalmail.com"
      type="cite">
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      <div class="moz-cite-prefix">I'm beginning to think that this high
        port number thing is a red herring. The router in question is a
        Meraki and it seems to have trouble allowing more than one SIP
        client to register with freeswitch (maybe a NAT/ALG issue)- is
        this some kind of FAQ? <br>
        <br>
        On 11/01/2018 15:47, Alex Lake wrote:<br>
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        <p>Hi,</p>
        <p>Got a problem with a handset that doesn't seem to be
          registering on our Freeswitch server. The diagnostics from a
          member of Cisco support said...</p>
        <p>"<span class="mediumtext">At packet 422 we see the phone <a
              class="moz-txt-link-rfc2396E"
              href="mailto:0_4147057716@192.168.40.26"
              moz-do-not-send="true">"0_4147057716@192.168.40.26"</a>
            send a registration request to 5.9.36.113:5060 from
            175.159.77.82:32060 - We do not see a reply to this from the
            SIP provider.<br>
            <br>
            We responses to requests from <a
              href="mailto:0_359240637@192.168.40.26"
              moz-do-not-send="true">0_359240637@192.168.40.26</a> with
            source port 5060, which suggests to me that the SIP provider
            is not replying to the high UDP port requests. As the MX is
            a NAT device, subsequent SIP requests will always have a
            high UDP port."</span></p>
        <p><span class="mediumtext">I'm not sure where we might
            configure acceptable ranges of SIP source or if it's
            something else.</span></p>
        <p><span class="mediumtext">Thanks for any suggestions...</span></p>
        <p><span class="mediumtext">Alex<br>
          </span></p>
        <br>
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        <br>
        <pre wrap="">_________________________________________________________________________
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      <p><br>
      </p>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_________________________________________________________________________
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Official FreeSWITCH Sites
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    <br>
    <br>
    <pre class="moz-signature" cols="72">-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
<a class="moz-txt-link-freetext" href="mailto:lists">mailto:lists</a> (att) telefaks.de
Internet: <a class="moz-txt-link-abbreviated" href="http://www.telefaks.de">www.telefaks.de</a>

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