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    <div class="moz-cite-prefix">Hi all,<br>
    </div>
    <blockquote
      cite="mid:1378583082.18520.YahooMailNeo@web126202.mail.ne1.yahoo.com"
      type="cite">
      <div style="color:#000; background-color:#fff; font-family:arial,
        helvetica, sans-serif;font-size:10pt">hi Markus,<br>
        how do you determine if it's a complete number? Do you expect
        the outgoing ISDN channel to tell you that?<br>
      </div>
    </blockquote>
    Sorry that I don't have all details; I am not as familary as I want
    to with the ISDN protocols. But I know that each number goes one by
    one to the outgoing ISDN and then outgoint ISDN tells you back when
    you have reached a destination. If needed I can make a log when I
    have my Siemens Hipath placed behind my Freeswitch in NT mode; there
    you can see the numbers comming one by one.<br>
    <blockquote
      cite="mid:1378583082.18520.YahooMailNeo@web126202.mail.ne1.yahoo.com"
      type="cite">
      <div style="color:#000; background-color:#fff; font-family:arial,
        helvetica, sans-serif;font-size:10pt"><br>
        I also wonder what happens if you attach an ISDN-SIP gateway,
        like Patton. Will you have a new SIP message (which?) on every
        dialed digit?<br>
      </div>
    </blockquote>
    Normaly all the products do the "overlap dialing" crap, as mentioned
    in my first email. I don't know for sure but I think there is no
    alternative, especially if you make only SIP and not ISDN.<br>
    <br>
    No comments about my design suggestion?<br>
    <br>
    Regards,<br>
    Markus<br>
    <blockquote
      cite="mid:1378583082.18520.YahooMailNeo@web126202.mail.ne1.yahoo.com"
      type="cite">
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        <div><br>
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          <div style="font-family: times new roman, new york, times,
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            <div dir="ltr">
              <hr size="1"> <font size="2" face="Arial"> <b><span
                    style="font-weight:bold;">From:</span></b> Markus
                M&uuml;ller <a class="moz-txt-link-rfc2396E"
                  href="mailto:freeswitchdev@priv.de">&lt;freeswitchdev@priv.de&gt;</a><br>
                <b><span style="font-weight: bold;">To:</span></b> <a
                  class="moz-txt-link-abbreviated"
                  href="mailto:freeswitch-dev@lists.freeswitch.org">freeswitch-dev@lists.freeswitch.org</a>
                <br>
                <b><span style="font-weight: bold;">Sent:</span></b>
                Saturday, September 7, 2013 1:52 PM<br>
                <b><span style="font-weight: bold;">Subject:</span></b>
                [Freeswitch-dev] Call relaying (ISDN &lt;&gt; Freeswitch
                &lt;&gt; ISDN)<br>
              </font> </div>
            <div class="y_msg_container"><br>
              Hello Freeswitch Developers,<br>
              <br>
              ISDN in germany (maybe also somewhere else) has a feature,
              which colides<br>
              with the design of a dialplan: it sends the numbers the
              user types into<br>
              his phone LIVE (!) through the ISDN network. If the number
              is complete,<br>
              the ISDN network tells it to the caller; Only now the call
              get<br>
              established (means the dialplan gets invoked). So you have
              to do an own<br>
              step (live and interactive determination of the number)
              BEFORE the<br>
              dialplan comes in line.<br>
              <br>
              Because this is object not supported by freeswith, in the
              following<br>
              situation<br>
              <br>
              User &lt;-&gt; Analog Phone &lt;-&gt; Siemens Hipath
              &lt;-&gt; ISDN &lt;-&gt; [FreeTDM &lt;-&gt;<br>
              Freeswitch &lt;-&gt; FreeTDM] &lt;-&gt; ISDN &lt;-&gt;
              World &lt;-&gt; ISDN Destiation<br>
              <br>
              you have to do "overlap dialing". Means, freeswitch waits
              some seconds<br>
              until the user has entered the last number and then it
              goes directly to<br>
              the dialplan.<br>
              <br>
              -&gt; This is not what I need!<br>
              <br>
              I want that it works as if there is no FreeSwitch in
              between. Means,<br>
              freeswitch should relay each number the user types into
              his phone to the<br>
              ISDN on the remote side, and make the dialplan stuff AFTER
              the number<br>
              has been dicovered for completeness.<br>
              <br>
              How do you think this should be implemented? If nobody has
              an object, I<br>
              would code this the following way into FreeTDM: If I get a
              call with a<br>
              starting number I know that it must go to the external
              ISDN (means: a<br>
              second dialplan), I first relay the typed numbers and
              determine the full<br>
              number. Only now, when I got the full number, I would give
              this call to<br>
              the higher layers.<br>
              <br>
              What you think about this?<br>
              <br>
              Regards,<br>
              Markus Mueller<br>
              <br>
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      <pre wrap="">_________________________________________________________________________
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