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    <div class="moz-cite-prefix">Hi.<br>
      <br>
      Thank you.<br>
      <br>
      You mean do each call separately? That works without a glitch,
      nothing peculiar.<br>
      <br>
      Thx,<br>
      <br>
      <br>
      BC<br>
      <br>
      <br>
      <br>
      On 09/25/12 23:28, Danny Nicholas wrote:<br>
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    <blockquote cite="mid:017101cd9b64$b3eb9c90$1bc2d5b0$@debsinc.com"
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        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Do
            the call both ways again and check(post) the CLI output.<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
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            <p class="MsoNormal"><b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">From:</span></b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">
                <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>
                [<a class="moz-txt-link-freetext" href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</a>] <b>On
                  Behalf Of </b>Bart Coninckx<br>
                <b>Sent:</b> Tuesday, September 25, 2012 4:23 PM<br>
                <b>To:</b> Asterisk Users Mailing List - Non-Commercial
                Discussion<br>
                <b>Subject:</b> [asterisk-users] no audio while call
                forwarding, yes audio with followme<o:p></o:p></span></p>
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        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <p class="MsoNormal" style="margin-bottom:12.0pt">Hi all,<br>
          <br>
          the subject says it all.<br>
          Technical details: <br>
          - Asterisk 1.8.7.1 <br>
          - Behind NAT <br>
          - Using external SIP provider<br>
          <br>
          The call forwarding is tested both with this functionality on
          the phone and with configuration in the dialplan. In the
          latter case a database variable is set to the external number,
          if set a Dial command calls this number. So really nothing
          fancy (actually I followed the example on <a
            moz-do-not-send="true"
            href="http://www.voip-info.org/wiki/view/Asterisk+call+forwarding">http://www.voip-info.org/wiki/view/Asterisk+call+forwarding</a>
          ).<br>
          <br>
          sip.conf has nat=yes, externip= ... and I tried every setting
          of directmedia in the providers configuration part. <br>
          <br>
          Followme works flawlessly, so I'm really wondering if this is
          a NAT issue. <br>
          <br>
          <br>
          Can anyone point me into a certain direction?<br>
          <br>
          <br>
          Thx!!!!<br>
          <br>
          <br>
          BC<o:p></o:p></p>
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