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<body class='hmmessage'><div dir='ltr'>Yes BDF, you are right, but I have been using Asterisk for a while and never faced such problem, there has to be some mechanism Freeswitch is lagging, beside this Freeswitch is quiet flexible to use due to its unique XML configuration style, <br>And thanks for response<br><br>Regards<br>Muhammad Farrukh<br><br><div><div id="SkyDrivePlaceholder"></div><hr id="stopSpelling">Date: Thu, 2 May 2013 08:51:42 -0400<br>From: bdfoster@davri.com<br>To: freeswitch-users@lists.freeswitch.org<br>Subject: Re: [Freeswitch-users] Dead channels in freeswitch<br><br><p dir="ltr">Just a note:</p>
<p dir="ltr">In your setup, a timeout if some sort is your only option. It may not be efficient, it may not be perfect, but neither are your WiFi/Mobile clients apparently. </p>
<p dir="ltr">-BDF</p>
<div class="ecxgmail_quote">On May 2, 2013 7:09 AM, "Farrukh Ali" <<a href="mailto:smfarrukh@live.com">smfarrukh@live.com</a>> wrote:<br><blockquote class="ecxgmail_quote" style="border-left:1px #ccc solid;padding-left:1ex;">
<div><div dir="ltr">Thanks everyone, and Steve timers might not be efficient solution but for now it will be good for testing purpose, could you please tell me exactly which file and parameter should I look for to change, is this configuration located in sofia.conf.xml ? kindly reply!<br>
and thanks to all again!<br><br>Regards,<br>Muhammad Farrukh<br><br><div><div></div><hr>From: <a href="mailto:steveayre@gmail.com" target="_blank">steveayre@gmail.com</a><br>Date: Tue, 30 Apr 2013 17:20:36 +0100<br>To: <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>
Subject: Re: [Freeswitch-users] Dead channels in freeswitch<br><br><div>SIP over UDP has no way of knowing the other side has lost connectivity (eg wifi down). Sip droid will know the local network is down and be able to hang up the call because of that. FreeSWITCH has no such knowledge, so won't know the call is hung up and will continue sending RTP.</div>
<div><br></div><div>3 possibilities are:</div><div>1) Use session timers</div><div>2) There is a parameter to hang up a call is no RTP has been received for a certain period of time</div><div><span>3) Use SIP/TCP, keepalives might help (but the default is measured in hours)</span></div>
<div><span><br></span></div><div>Note that all of these are timeouts - you simply won't be able to know the call has ended immediately.</div><div><br>Steve</div><div><br></div><div><br></div><div><br>On 30 Apr 2013, at 08:33, Farrukh Ali <<a href="mailto:smfarrukh@live.com" target="_blank">smfarrukh@live.com</a>> wrote:<br>
<br></div><blockquote><div>
<div dir="ltr">Hi,<br><br>I need some help, there is an issue in Freeswitch with SIP clients, when a call is not hung up properly Freeswitch does not close its RTP channels, and it starts to get load on bandwidth after too many dead channels are generated, Improper hang up means if during call sip client went out of the range of WiFi, the client i.e SipDroid shows call hang up but freeswitch continues to send voice packets. Kindly reply<br>
</div>
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