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    Thanks for the helps<br>
    <br>
    I use channel originate command to achieve this.<br>
    <br>
    Command:<br>
    <br>
    asteriskCLI&gt; channel originate SIP/201 application ConfBrigde
    1234<br>
    <br>
    This will make a call to the 201 user and when connected,it will be
    routed to conference room .<br>
    <br>
    Thanks<br>
    NIkhil<br>
    <br>
    On 06/15/2011 02:17 PM, virendra bhati wrote:
    <blockquote
      cite="mid:BANLkTi=hqyyOGe7fs2MCssm3eAdN0pDk6Q@mail.gmail.com"
      type="cite">
      <div dir="ltr">Hi, <br>
        <br>
        You may used the Page() function of asterisk. Which will work
        the same as you are required at this moment.<br>
        <br>
        &nbsp;<br>
        <br>
        <div class="gmail_quote">On Wed, Jun 15, 2011 at 12:51 PM, Alex
          Balashov <span dir="ltr">&lt;<a moz-do-not-send="true"
              href="mailto:abalashov@evaristesys.com">abalashov@evaristesys.com</a>&gt;</span>
          wrote:<br>
          <blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
            0.8ex; border-left: 1px solid rgb(204, 204, 204);
            padding-left: 1ex;">On 06/15/2011 01:34 AM, Nikhil wrote:<br>
            <blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
              0.8ex; border-left: 1px solid rgb(204, 204, 204);
              padding-left: 1ex;">
              Hi<br>
              Asterisk support dialout conference?.My requirement is
              when type a CLI<br>
              command with argument as a number ,asterisk should able to
              make a call<br>
              to that number and when connected ,that channel should
              entered in to<br>
              the conference room,like this I should able to add
              multiple users into<br>
              the conference.I am using ConfBridge application for
              asterisk version<br>
              1.6.2<br>
            </blockquote>
            <br>
            This is something that can be accomplished with the manager
            interface or call files.<br>
            <br>
            -- <br>
            Alex Balashov - Principal<br>
            Evariste Systems LLC<br>
            260 Peachtree Street NW<br>
            Suite 2200<br>
            Atlanta, GA 30303<br>
            Tel: +1-678-954-0670<br>
            Fax: +1-404-961-1892<br>
            Web: <a moz-do-not-send="true"
              href="http://www.evaristesys.com/" target="_blank">http://www.evaristesys.com/</a><br>
            <br>
            --<br>
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            every Thurs:<br>
            &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;<a moz-do-not-send="true"
              href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
            <br>
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            To UNSUBSCRIBE or update options visit:<br>
            &nbsp;<a moz-do-not-send="true"
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              target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
          </blockquote>
        </div>
        <br>
        <br clear="all">
        <br>
        -- <br>
        <div dir="ltr"><br>
          <br>
          <br>
          -----<br>
          Thanks and regards<br>
          <br>
          &nbsp;Virendra Bhati<br>
          +91-9172341457<br>
          Asterisk Engineer<br>
        </div>
        <br>
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To UNSUBSCRIBE or update options visit:
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    </blockquote>
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