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<DIV><FONT size=2>ok.</FONT></DIV>
<DIV><FONT size=2> no nat is used. but for sure it could be a client
problem</FONT></DIV>
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style="BORDER-LEFT: #000000 2px solid; PADDING-LEFT: 5px; PADDING-RIGHT: 0px; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="FONT: 10pt arial; BACKGROUND: #e4e4e4; font-color: black"><B>From:</B>
<A title=anthony.minessale@gmail.com
href="mailto:anthony.minessale@gmail.com">Anthony Minessale</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=freeswitch-users@lists.freeswitch.org
href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Thursday, March 25, 2010 2:29
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Freeswitch-users] send fake
UDP packets</DIV>
<DIV><BR></DIV>hanging up a leg will automatically hang up b leg, if it doesnt
you still have unresolved nat problems.<BR><BR>make your phone register every
20 sec to keep the sip path alive.<BR><BR><BR>
<DIV class=gmail_quote>On Thu, Mar 25, 2010 at 1:20 PM, Madovsky <SPAN
dir=ltr><<A
href="mailto:infos@madovsky.org">infos@madovsky.org</A>></SPAN> wrote:<BR>
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class=gmail_quote>
<DIV bgcolor="#ffffff">
<DIV><FONT size=2>I use an audio stream relayed by a client that has no
audio DSP card.</FONT></DIV>
<DIV><FONT size=2>so when I stop to talk, no UDP audio packet is sent to FS.
Do you know</FONT></DIV>
<DIV><FONT size=2>a trick to send fake UDP packet to keep the call alive
? (I need to keep rtp-timeout short to hang up leg B when leg A hangs
up)</FONT></DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT size=2>Thanks</FONT></DIV>
<DIV><FONT
size=2><BR>F</FONT></DIV></DIV><BR>_______________________________________________<BR>FreeSWITCH-users
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clear=all><BR>-- <BR>Anthony Minessale II<BR><BR>FreeSWITCH <A
href="http://www.freeswitch.org/">http://www.freeswitch.org/</A><BR>ClueCon <A
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#freeswitch<BR><BR>FreeSWITCH Developer Conference<BR><A
href="mailto:sip%3A888@conference.freeswitch.org">sip:888@conference.freeswitch.org</A><BR><A
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