dingyu
/
Asterisk


			
							123456789101112131415161718192021222324252627282930313233343536373839404142434445464748495051525354555657585960616263646566676869707172737475767778798081828384858687888990919293949596979899100101102103104105106107108109110111112113
							<!DOCTYPE html>
<html>
    <head>
        <title>Asterisk Project : Asterisk 13 Function_SIPPEER</title>
        <link rel="stylesheet" href="styles/site.css" type="text/css" />
        <META http-equiv="Content-Type" content="text/html; charset=UTF-8">
    </head>

    <body class="theme-default aui-theme-default">
        <div id="page">
            <div id="main" class="aui-page-panel">
                <div id="main-header">
                    <div id="breadcrumb-section">
                        <ol id="breadcrumbs">
                            <li class="first">
                                <span><a href="index.html">Asterisk Project</a></span>
                            </li>
                                                    <li>
                                <span><a href="Asterisk-13-Command-Reference_29394274.html">Asterisk 13 Command Reference</a></span>
                            </li>
                                                    <li>
                                <span><a href="Asterisk-13-Dialplan-Functions_29394287.html">Asterisk 13 Dialplan Functions</a></span>
                            </li>
                                                </ol>
                    </div>
                    <h1 id="title-heading" class="pagetitle">
                                                <span id="title-text">
                            Asterisk Project : Asterisk 13 Function_SIPPEER
                        </span>
                    </h1>
                </div>

                <div id="content" class="view">
                    <div class="page-metadata">
                        
        
            Created by <span class='author'> wikibot</span> on Aug 08, 2014
                        </div>
                    <div id="main-content" class="wiki-content group">
                    <h1 id="Asterisk13Function_SIPPEER-SIPPEER()">SIPPEER()</h1>

<h3 id="Asterisk13Function_SIPPEER-Synopsis">Synopsis</h3>

<p>Gets SIP peer information.</p>

<h3 id="Asterisk13Function_SIPPEER-Description">Description</h3>


<h3 id="Asterisk13Function_SIPPEER-Syntax">Syntax</h3>


<div class="preformatted panel" style="border-width: 1px;"><div class="preformattedContent panelContent">
<pre>SIPPEER(peername,item)</pre>
</div></div>

<h5 id="Asterisk13Function_SIPPEER-Arguments">Arguments</h5>

<ul>
	<li><code>peername</code></li>
	<li><code>item</code>
	<ul>
		<li><code>ip</code> - (default) The IP address.</li>
		<li><code>port</code> - The port number.</li>
		<li><code>mailbox</code> - The configured mailbox.</li>
		<li><code>context</code> - The configured context.</li>
		<li><code>expire</code> - The epoch time of the next expire.</li>
		<li><code>dynamic</code> - Is it dynamic? (yes/no).</li>
		<li><code>callerid_name</code> - The configured Caller ID name.</li>
		<li><code>callerid_num</code> - The configured Caller ID number.</li>
		<li><code>callgroup</code> - The configured Callgroup.</li>
		<li><code>pickupgroup</code> - The configured Pickupgroup.</li>
		<li><code>namedcallgroup</code> - The configured Named Callgroup.</li>
		<li><code>namedpickupgroup</code> - The configured Named Pickupgroup.</li>
		<li><code>codecs</code> - The configured codecs.</li>
		<li><code>status</code> - Status (if qualify=yes).</li>
		<li><code>regexten</code> - Extension activated at registration.</li>
		<li><code>limit</code> - Call limit (call-limit).</li>
		<li><code>busylevel</code> - Configured call level for signalling busy.</li>
		<li><code>curcalls</code> - Current amount of calls. Only available if call-limit is set.</li>
		<li><code>language</code> - Default language for peer.</li>
		<li><code>accountcode</code> - Account code for this peer.</li>
		<li><code>useragent</code> - Current user agent header used by peer.</li>
		<li><code>maxforwards</code> - The value used for SIP loop prevention in outbound requests</li>
		<li><code>chanvar<a class="createlink" href="/wiki/pages/createpage.action?spaceKey=AST&amp;title=name&amp;linkCreation=true&amp;fromPageId=29394333">name</a></code> - A channel variable configured with setvar for this peer.</li>
		<li><code>codec<a class="createlink" href="/wiki/pages/createpage.action?spaceKey=AST&amp;title=x&amp;linkCreation=true&amp;fromPageId=29394333">x</a></code> - Preferred codec index number <em>x</em> (beginning with zero).</li>
	</ul>
	</li>
</ul>


<h3 id="Asterisk13Function_SIPPEER-SeeAlso">See Also</h3>


<h3 id="Asterisk13Function_SIPPEER-ImportVersion">Import Version</h3>

<p>This documentation was imported from Asterisk Version SVN-branch-13-r420538</p>
                    </div>

                    
                </div>             </div> 
            <div id="footer">
                <section class="footer-body">
                    <p>Document generated by Confluence on Aug 11, 2014 13:48</p>
                </section>
            </div>
        </div>     </body>
</html>