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    <h2><a href="https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_SIPPEER">Asterisk 11 Function_SIPPEER</a></h2>
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         <h1><a name="Asterisk11Function_SIPPEER-SIPPEER%28%29"></a>SIPPEER()</h1>

<h3><a name="Asterisk11Function_SIPPEER-Synopsis"></a>Synopsis</h3>

<p>Gets SIP peer information.</p>
<h3><a name="Asterisk11Function_SIPPEER-Description"></a>Description</h3>



<h3><a name="Asterisk11Function_SIPPEER-Syntax"></a>Syntax</h3>

<div class="preformatted panel" style="border-width: 1px;"><div class="preformattedContent panelContent">
<pre>SIPPEER(peername,item)</pre>
</div></div>
<h5><a name="Asterisk11Function_SIPPEER-Arguments"></a>Arguments</h5>


<ul>
        <li>peername</li>
        <li>item
        <ul>
                <li>ip - (default) The IP address.</li>
                <li>port - The port number.</li>
                <li>mailbox - The configured mailbox.</li>
                <li>context - The configured context.</li>
                <li>expire - The epoch time of the next expire.</li>
                <li>dynamic - Is it dynamic? (yes/no).</li>
                <li>callerid_name - The configured Caller ID name.</li>
                <li>callerid_num - The configured Caller ID number.</li>
                <li>callgroup - The configured Callgroup.</li>
                <li>pickupgroup - The configured Pickupgroup.</li>
                <li>codecs - The configured codecs.</li>
                <li>status - Status (if qualify=yes).</li>
                <li>regexten - Extension activated at registration.</li>
                <li>limit - Call limit (call-limit).</li>
                <li>busylevel - Configured call level for signalling busy.</li>
                <li>curcalls - Current amount of calls. Only available if call-limit is set.</li>
                <li>language - Default language for peer.</li>
                <li>accountcode - Account code for this peer.</li>
                <li>useragent - Current user agent header used by peer.</li>
                <li>maxforwards - The value used for SIP loop prevention in outbound requests</li>
                <li>chanvar<a href="/wiki/pages/createpage.action?spaceKey=AST&amp;title=name&amp;linkCreation=true&amp;fromPageId=20186946" class="createlink">name</a> - A channel variable configured with setvar for this peer.</li>
                <li>codec<a href="/wiki/pages/createpage.action?spaceKey=AST&amp;title=x&amp;linkCreation=true&amp;fromPageId=20186946" class="createlink">x</a> - Preferred codec index number <em>x</em> (beginning with zero).</li>
        </ul>
        </li>
</ul>



<h3><a name="Asterisk11Function_SIPPEER-ImportVersion"></a>Import Version</h3>

<p>This documentation was imported from Asterisk Version SVN-trunk-r370322</p>
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