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                            Asterisk Project : Asterisk 13 Application_Page
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            Created by <span class='author'> wikibot</span> on Aug 08, 2014
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                    <h1 id="Asterisk13Application_Page-Page()">Page()</h1>

<h3 id="Asterisk13Application_Page-Synopsis">Synopsis</h3>

<p>Page series of phones</p>

<h3 id="Asterisk13Application_Page-Description">Description</h3>

<p>Places outbound calls to the given <em>technology</em>/<em>resource</em> and dumps them into a conference bridge as muted participants. The original caller is dumped into the conference as a speaker and the room is destroyed when the original caller leaves.</p>


<h3 id="Asterisk13Application_Page-Syntax">Syntax</h3>


<div class="preformatted panel" style="border-width: 1px;"><div class="preformattedContent panelContent">
<pre>Page(Technology/Resource&amp;[Technology2/Resource2[&amp;...]],[options,[timeout]])</pre>
</div></div>

<h5 id="Asterisk13Application_Page-Arguments">Arguments</h5>

<ul>
	<li><code>Technology/Resource</code>
	<ul>
		<li><code>Technology/Resource</code> - Specification of the device(s) to dial. These must be in the format of <code>Technology/Resource</code>, where <em>Technology</em> represents a particular channel driver, and <em>Resource</em> represents a resource available to that particular channel driver.</li>
		<li><code>Technology2/Resource2</code> - Optional extra devices to dial in parallel<br />
If you need more than one, enter them as Technology2/Resource2&amp; Technology3/Resourse3&amp;.....</li>
	</ul>
	</li>
	<li><code>options</code>
	<ul>
		<li><code>b</code> - Before initiating an outgoing call, Gosub to the specified location using the newly created channel. The Gosub will be executed for each destination channel.
		<ul>
			<li><code>context</code></li>
			<li><code>exten</code></li>
			<li><code>priority</code>
			<ul>
				<li><code>arg1</code></li>
				<li><code>argN</code></li>
			</ul>
			</li>
		</ul>
		</li>
		<li><code>B</code> - Before initiating the outgoing call(s), Gosub to the specified location using the current channel.
		<ul>
			<li><code>context</code></li>
			<li><code>exten</code></li>
			<li><code>priority</code>
			<ul>
				<li><code>arg1</code></li>
				<li><code>argN</code></li>
			</ul>
			</li>
		</ul>
		</li>
		<li><code>d</code> - Full duplex audio</li>
		<li><code>i</code> - Ignore attempts to forward the call</li>
		<li><code>q</code> - Quiet, do not play beep to caller</li>
		<li><code>r</code> - Record the page into a file ( <code>CONFBRIDGE(bridge,record_conference)</code>)</li>
		<li><code>s</code> - Only dial a channel if its device state says that it is <code>NOT_INUSE</code></li>
		<li><code>A</code> - Play an announcement to all paged participants
		<ul>
			<li><code>x</code> - The announcement to playback to all devices</li>
		</ul>
		</li>
		<li><code>n</code> - Do not play announcement to caller (alters <code>A<img class="emoticon emoticon-cross" src="images/icons/emoticons/error.png" data-emoticon-name="cross" alt="(error)"/></code> behavior)</li>
	</ul>
	</li>
	<li><code>timeout</code> - Specify the length of time that the system will attempt to connect a call. After this duration, any page calls that have not been answered will be hung up by the system.</li>
</ul>


<h3 id="Asterisk13Application_Page-SeeAlso">See Also</h3>

<ul>
	<li><a href="Asterisk-13-Application_ConfBridge_29394402.html">Asterisk 13 Application_ConfBridge</a></li>
</ul>


<h3 id="Asterisk13Application_Page-ImportVersion">Import Version</h3>

<p>This documentation was imported from Asterisk Version SVN-branch-13-r420538</p>
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                    <p>Document generated by Confluence on Aug 11, 2014 13:45</p>
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