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📁 KphoneSI (kpsi) is a SIP (Session Initiation Protocol) user agent for Linux, with which you can in
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN"><html><head>  <meta http-equiv="CONTENT-TYPE" content="text/html; charset=utf-8">  <title>How to Call</title>  <meta name="GENERATOR" content="OpenOffice.org 1.1.1  (Linux)">  <meta name="CREATED" content="20051015;8391400">  <meta name="CHANGED" content="20051015;8460200">  <style type="text">	<!--		@page { size: 21cm 29.7cm; margin-left: 2cm; margin-right: 2cm; margin-top: 0.8cm; margin-bottom: 0.8cm }		TD P.cjk { font-weight: medium }		TD P.ctl { font-weight: medium }		H1.western { font-family: "Nimbus Sans L" }		P.cjk { font-weight: medium }		P.ctl { font-weight: medium }	-->	</style></head><body dir="ltr" lang="en-US"><a href="index.html">back to the ToC</a><h1 class="western" style="margin-top: 0cm; margin-bottom: 0.2cm; page-break-before: always;">KPhoneISUser Manual:&nbsp;How to Call</h1><p class="western" style="margin-bottom: 0cm;"><span style="font-weight: bold;">Dial</span><br></p><p class="western" style="margin-bottom: 0cm;">To start a session (asto call someone) you may proceed as follows:</p><br>You have three main choices:<br><ol>  <li>First select the URI, then pressthe session icon of your session type (audio, video, ...)</li>  <li>Pressthe session icon of your session type (audio, video, ...) directly<br>  </li>  <li>Click a contact,using the right mouse button and select a session type</li></ol><p class="western" style="margin-bottom: 0cm;">In all cases the <span style="font-weight: bold;">callwidget pops up</span>, in the cases one and three the URI is stillfilled in.<br></p><p class="western" style="margin-bottom: 0cm;">The selection of teh URImay be achieved by:<br></p><ol>  <li>Using the phone book to select one; the dial field will be filledautomatically<br>  </li>  <li>entering one into the dial field directly<br>  </li></ol><p class="western" style="margin-bottom: 0cm;"> </p><img style="width: 408px; height: 320px;" alt="dial0" src="dial0.png"><br><br><h2 style="font-weight: bold;"><small><small style="font-weight: normal;">After pressing the session icon, thepartner is invited</small></small></h2><img style="width: 824px; height: 321px;" alt="dial 1" src="dial1.png"><br><br>if she accepts (presses the Accept button) the call is switched through.<br><br>The CallWidgets are hidden afterwards. The may begot back by double clicking the appropriate line in the active calllist.<br><br><p><small><small><span style="font-weight: normal;"></span></small><big><big><span style="font-weight: bold;">Exceptions:</span></big><br></big></small></p><p class="western">If you do not select a URI in the main widget, youmay also enter it in the call widget:</p><ul>  <li>    <p class="western" style="margin-bottom: 0.1cm;">typing it<br>    </p>  </li>  <li>    <p class="western" style="margin-bottom: 0.1cm;">selecting from therecent call list (press arrow down and select from the pull down)</p>  </li>  <li>    <p class="western" style="margin-bottom: 0.1cm;">using the phonebook press the icon select and return with a double click</p>  </li></ul><p class="western"><br>Afterwards press the dial button. On the callee'sside the call widget appears (possibly accompanied by the dialtone)</p>If the callee presses the Accept Button the callis established. <br><br>The CallWidgets are hidden afterwards. The may begot back by double clicking the appropriate line in the active calllist.<br><br clear="left"><br><h2>Ways to enter the SIP-URI</h2><p class="western">We have the following possibilities:</p><ol>  <li>    <p class="western" style="margin-bottom: 0.1cm;">userid</p>  </li>  <li>    <p class="western" style="margin-bottom: 0.1cm;"><a href="mailto:userid@host">userid@host</a></p>  </li>  <li>    <p class="western" style="margin-bottom: 0.1cm;"><a href="mailto:userid@host">userid@host:port</a></p>  </li>  <li>    <p class="western" style="margin-bottom: 0.1cm;">tel: phone number</p>  </li>  <li>    <p class="western" style="margin-bottom: 0.1cm;">userid,DTMF-string</p>  </li>  <li>    <p class="western" style="margin-bottom: 0.1cm;">userid@host,DTMF-string</p>  </li></ol><p class="western" style="margin-bottom: 0cm;">(1) is completed to<a href="mailto:userid@proxy">userid@proxy</a> automatically; (5) to<a href="mailto:userid@proxy">userid@proxy</a>,DTMF-string.</p><p class="western" style="margin-bottom: 0cm;">The DTMF-String is sentin digital form, as soon a the call is active. </p><p class="western" style="margin-bottom: 0cm;">(4) can be used toaddress a telephone gateway, if the proxy supports this.</p><br><h2>Password Entry</h2>The password is asked, if the proxy agent is requesting for one. Then awindow pops up:<br><img style="width: 179px; height: 206px;" alt="password" src="enterpw.png"><br><p>The first four lines contain information about your connection:<br></p><ul>  <li><span style="font-style: italic;">Authentication for Entry: </span>yourtab in the kphonesirc database</li>  <li><span style="font-style: italic;">SipUri : </span>Your sipuri</li>  <li><span style="font-style: italic;">Server :</span> The proxyagent, your are connected to.</li></ul>The entry fields:<br><br><ul>  <li><span style="font-style: italic;">Usename :</span> enter youruser name here. If kphone still knows your user name, it will display it</li>  <li><span style="font-style: italic;">Password:</span> enter yourpassword here</li>  <li><span style="font-style: italic;">Save password:</span> select,if you want your password written into the kphonesirc database</li></ul><p>You may solicit your entries by pressing OK, or cancel them, bypressing Cancel<br></p><h2><br></h2><h2>Remote Start </h2><h3>Start a 鈥濺emote-Application鈥

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