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    <div class="moz-cite-prefix">Hi Stanislav Sinyagin,<br>
      <br>
      what you mean is the "overlap dialing" feature, this is already
      implemented and working in freetdm. It waits some seconds after
      the last number and then dials.<br>
      <br>
      ====&gt;&gt; This is not what I am searching for ! This is also
      not what I am talking about ! Please accept this and make sure how
      the reality is before saying that I am telling wrong things.<br>
      <br>
      I am searching what I worte, if you believe or not. ISDN in
      germany currently works this way, and yes, there is a
      "completeness" of numbers which is interactively determined by
      sending each key through the ISDN network and getting back when a
      destination has been found.<br>
      <br>
      Regards,<br>
      Markus Mueller<br>
      <br>
    </div>
    <blockquote
      cite="mid:1378595113.21724.YahooMailNeo@web126203.mail.ne1.yahoo.com"
      type="cite">
      <div style="color:#000; background-color:#fff; font-family:arial,
        helvetica, sans-serif;font-size:10pt">I guess there's no such
        ting as "complete number", but a simple interdigit timeout. If
        you stop dialing and don't enter digits for 10 seconds, it
        automatically places the call.<br>
        <br>
        So, you need a gateway which supports such an interdigit
        timeout. In theory, you can implement that with an external
        script in FreeSWITCH. On practice, needs testing and a proof of
        concept.<br>
        <br>
        <br>
        <br>
        <div><span><br>
          </span></div>
        <div><br>
        </div>
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          font-size: 10pt;">
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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 10:08 PM<br>
                <b><span style="font-weight: bold;">Subject:</span></b>
                Re: [Freeswitch-dev] Call relaying (ISDN &lt;&gt;
                Freeswitch &lt;&gt; ISDN)<br>
              </font> </div>
            <div class="y_msg_container"><br>
              <div id="yiv1287044465">
                <div>
                  <div class="yiv1287044465moz-cite-prefix">Hi all,<br>
                  </div>
                  <blockquote 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 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 type="cite">
                    <div
                      style="color:#000;background-color:#fff;font-family:arial,
                      helvetica, sans-serif;font-size:10pt;">
                      <div><br>
                      </div>
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                        sans-serif;font-size:10pt;">
                        <div style="font-family:times new roman, new
                          york, times, serif;font-size:12pt;">
                          <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 moz-do-not-send="true"
                                rel="nofollow"
                                class="yiv1287044465moz-txt-link-rfc2396E"
                                ymailto="mailto:freeswitchdev@priv.de"
                                target="_blank"
                                href="mailto:freeswitchdev@priv.de">&lt;freeswitchdev@priv.de&gt;</a><br>
                              <b><span style="font-weight:bold;">To:</span></b>
                              <a moz-do-not-send="true" rel="nofollow"
                                class="yiv1287044465moz-txt-link-abbreviated"
ymailto="mailto:freeswitch-dev@lists.freeswitch.org" target="_blank"
                                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="yiv1287044465y_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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                    <br>
                    <pre>_________________________________________________________________________
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      <pre wrap="">_________________________________________________________________________
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