Thanks,<div>I found solution - in Asterisk - rtptimeout</div><div>I am using rtpproxy with Kamailio, there is timeout socket, but for this moment, it looks complicated for me.</div><div><br><br><div class="gmail_quote">On Mon, Feb 13, 2012 at 5:53 PM, David <span dir="ltr">&lt;<a href="mailto:david@styleflare.com">david@styleflare.com</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
  
    
  
  <div bgcolor="#FFFFFF" text="#000000">
    Doubt it.<br>
    <br>
    You would need your &quot;media&quot; gateway to detect such a case.<div><div class="h5"><br>
    <br>
    <br>
    <br>
    On 2/13/12 10:44 AM, Stoyan Mihaylov wrote:
    </div></div><blockquote type="cite"><div><div class="h5">Problem - if connection drop, call can persist.
      <div>In Asterisk there is silencedetecthangup - which should cause
        hangup, if there is full silence for desired period of time.</div>
      <div>Unfortunately it does not hangup.</div>
      <div>I mean:</div>
      <div>SIP client 1 -&gt;Kamailio -&gt; Asterisk -&gt;Kamailio -&gt;
        SIP client 2</div>
      <div>If I drop connection for one of SIP clients, I expect call
        should be automatically hangup after a time I set (20 sec).</div>
      <div>But call persists. In worst case - if connection drops for
        both clients, call will persist until Asterisk is restarted.</div>
      <div>I will continue to look how to solve problem with Asterisk,
        but I am curious if this can be done from Kamailio also.</div>
      <div>If I can cancel call from both places - it will be great.</div>
      <div>I need to ensure that if something wrong happens, call will
        be dropped within 30 sec maximum.</div>
      <div>Stoyan</div>
      <br>
      <fieldset></fieldset>
      <br>
      </div></div><pre>_______________________________________________
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    </blockquote>
    <br>
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