<div dir="ltr"><div>Please get a debug log of such a call and put it on <a href="http://pastebin.freeswitch.org">pastebin.freeswitch.org</a>.<br></div>-MC<br></div><div class="gmail_extra"><br><br><div class="gmail_quote">
On Tue, Apr 30, 2013 at 12:33 AM, Farrukh Ali <span dir="ltr"><<a href="mailto:smfarrukh@live.com" target="_blank">smfarrukh@live.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<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>
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