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    Hello<br>
    <br>
    With or without the extension parameter, it's exactly the same.<br>
    <br>
    Thanks<br>
    <br>
    On 01/05/2015 20:04, Stanislav Sinyagin wrote:
    <blockquote
cite="mid:CACOYK=hhrQ-dJoF_vpYKkceh1M=iHcc_AhYj2Rb1+4_LCNCgFw@mail.gmail.com"
      type="cite">
      <p dir="ltr">Remove the extension parameter and see if it helps.</p>
      <div class="gmail_quote">On May 1, 2015 6:11 PM, "Tanguy" &lt;<a
          moz-do-not-send="true" href="mailto:phenix@vfemail.net">phenix@vfemail.net</a>&gt;
        wrote:<br type="attribution">
        <blockquote class="gmail_quote" style="margin:0 0 0
          .8ex;border-left:1px #ccc solid;padding-left:1ex">
          <div text="#000000" bgcolor="#FFFFFF"> Hello, <br>
            <br>
            My provider did not send correct DID number in the INVITE
            packet but i can use "To" argument<br>
            <br>
            <tt>INVITE <a moz-do-not-send="true"
href="mailto:sip:gw+0a96c3d3-0b0e-4864-b9ec-759fa4422429@92.222.18.xxx:5080;transport=udp;gw=0a96c3d3-0b0e-4864-b9ec-759fa4422429"
                target="_blank">sip:gw+0a96c3d3-0b0e-4864-b9ec-759fa4422429@92.222.18.xxx:5080;transport=udp;gw=0a96c3d3-0b0e-4864-b9ec-759fa4422429</a>
              SIP/2.0.<br>
              Call-ID: <a moz-do-not-send="true"
                href="mailto:25016-VB-188fd96b-526e3dbd4@sip.ovh.fr"
                target="_blank">25016-VB-188fd96b-526e3dbd4@sip.ovh.fr</a>.<br>
              Contact: &lt;sip:<a moz-do-not-send="true"
                href="http://10.7.1.60:5060" target="_blank">10.7.1.60:5060</a>&gt;.<br>
              Content-Type: application/sdp.<br>
              CSeq: 403749831 INVITE.<br>
              From: "0967212xxx"
              <a moz-do-not-send="true"
                href="mailto:sip:0967212xxx@sip.ovh.fr;user=phone"
                target="_blank">&lt;sip:0967212xxx@sip.ovh.fr;user=phone&gt;</a>;tag=25016-VE-188fd96c-18bb43586.<br>
              Max-Forwards: 27.<br>
              Record-Route: &lt;sip:91.121.129.20:5060;lr&gt;.<br>
              <b>To: <a moz-do-not-send="true"
                  href="mailto:sip:0557590xxx@10.7.1.60;user=phone"
                  target="_blank">&lt;sip:0557590xxx@10.7.1.60;user=phone&gt;</a>.</b><br>
              Via: SIP/2.0/UDP
              91.121.129.20:5060;branch=z9hG4bK-WGZO-1fe73949-2df58378.</tt><br>
            <br>
            Using asterisk i can bypass the issue using something like&nbsp;
            <tt>exten =&gt;
              s,1,Goto(from-trunk,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1)</tt>
            but i am unable to do the same under freeswitch.<br>
            <br>
            My trunk configuration seems correct, as you can see i used&nbsp;
            auto_to_user, but the destination number remains <tt>0033972480xxx

              when i call 0557590xxx.</tt><br>
            <br>
            2015-05-01 18:02:12.235827 [INFO] mod_dialplan_xml.c:635
            Processing 0967212xxx &lt;0967212xxx&gt;-&gt;0033972480xxx
            in context public<br>
            <br>
            <br>
            <tt>&lt;include&gt;<br>
              &nbsp;&nbsp;&nbsp; &lt;gateway
              name="0a96c3d3-0b0e-4864-b9ec-759fa4422429"&gt;<br>
              &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;param name="username" value="0033972480xxx"/&gt;<br>
              &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;param name="password" value="xxxxxxx"/&gt;<br>
              &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;param name="proxy" value="<a
                moz-do-not-send="true" href="http://sip.ovh.fr"
                target="_blank">sip.ovh.fr</a>"/&gt;<br>
              &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;param name="expire-seconds" value="800"/&gt;<br>
              &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;param name="register" value="true"/&gt;<br>
              &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;param name="retry-seconds" value="30"/&gt;<br>
              &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;param name="extension" value="auto_to_user"/&gt;<br>
              &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;param name="context" value="public"/&gt;<br>
              &nbsp;&nbsp;&nbsp; &lt;/gateway&gt;<br>
              &lt;/include&gt;</tt><br>
            <br>
            I tried to edit my inbound dialplan manually, it works
            using&nbsp; &lt;condition field="${sip_to_user}"
            expression="0557590xxx" &gt; but i prefer a proper way to do
            this because i will also use telcos with normal invite
            packets<br>
            <br>
            I how i can copy $sip_to_header to destination for this
            specific trunk ?<br>
            <br>
            Please note that i use fusionpbx.<br>
            <br>
            Best regards, sorry for my bad English<br>
            &nbsp;<br>
            <br>
          </div>
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