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"><<a href="mailto:david@styleflare.com">david@styleflare.com</a>></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 "media" 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 ->Kamailio -> Asterisk ->Kamailio ->
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>
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