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    <p>I'd ask provider to make a change and send destination number in
      SIP URI like most others do. <br>
    </p>
    <p>Especially if later you get different numbers from a different
      provider, you might benefit from having it all standard from day
      1.<br>
    </p>
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</pre>
    <div class="moz-cite-prefix">On 16-09-07 11:38 AM, Allan Kristensen
      wrote:<br>
    </div>
    <blockquote
cite="mid:CAA4dF92vdg+cK7QiMCvyxGh5_yrfe7uwiCVtAWrHkk6eA7mruA@mail.gmail.com"
      type="cite">
      <div dir="ltr">Hello,
        <div><br>
        </div>
        <div>I've just setup a sip trunk with a new provider, however
          there are some issues receiving calls.</div>
        <div><br>
        </div>
        <div>The INVITE looks like this:</div>
        <div><br>
        </div>
        <div><font face="monospace, monospace">   INVITE <a
              moz-do-not-send="true"
              href="http://sip:NCdWXsdSsaF2B@192.168.0.190:52284">sip:NCdWXsdSsaF2B@192.168.0.190:52284</a>
            SIP/2.0<br>
          </font></div>
        <div><font face="monospace, monospace">   ....</font></div>
        <div>
          <div><font face="monospace, monospace">   From:
              <a class="moz-txt-link-rfc2396E" href="mailto:sip:+XXXXXXXXXX@x.x.x.x">&lt;sip:+XXXXXXXXXX@x.x.x.x&gt;</a>;tag=Ud8ghaXg01821203</font></div>
          <div><font face="monospace, monospace">   To:
              <a class="moz-txt-link-rfc2396E" href="mailto:sip:+YYYYYYYYYY@x.x.x.x">&lt;sip:+YYYYYYYYYY@x.x.x.x&gt;</a></font></div>
        </div>
        <div><font face="monospace, monospace">   ...</font></div>
        <div><br>
        </div>
        <div>The invite has my account name, instead of the destination
          number and the real destination number is in the "To:" field
          (which it normally is I guess ;-)<br>
        </div>
        <div><br>
        </div>
        <div>Then Freeswitch does this of cause:</div>
        <div><br>
        </div>
        <div><font face="monospace, monospace">[INFO]
            mod_dialplan_xml.c:637 Processing
            +XXXXXXXXXX &lt;+XXXXXXXXXX&gt;-&gt;NCdWXsdSsaF2B in context
            inbound<br>
          </font></div>
        <div><br>
        </div>
        <div>That is undesirable, so Is there any way for Freeswitch to
          grab the destination number from the to field instead of the
          INVITE string ?</div>
        <div>I could make a special sip profile for that provider with
          it's own dialplan and then set destination_number from the
          "sip_to_user" variable and transfer to the real dialplan, but
          it just seems so messy...</div>
        <div>Or is it my provider that is doing it wrong?<br>
        </div>
        <div><br>
        </div>
        <div>Best regards,</div>
        <div>  Allan</div>
      </div>
      <br>
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      <br>
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