what is the lowest value I can use for sip-force-expires ?<div><br></div><div>Thanks</div><div>Lloyd<br>
<br><br><div class="gmail_quote">On Thu, Apr 8, 2010 at 9:29 PM, Aloysius Lloyd <span dir="ltr"><<a href="mailto:lloyd.aloysius@gmail.com">lloyd.aloysius@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div><br></div><div>Yes Asterisk config using the qualify=yes. </div><div><br></div><div>I will modify the Registration time out by 50sec and see how it is behaving.</div><div><br></div><div>Thanks</div><div>Lloyd</div><div>
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<br><br><div class="gmail_quote">On Thu, Apr 8, 2010 at 7:23 PM, David Ponzone <span dir="ltr"><<a href="mailto:david.ponzone@gmail.com" target="_blank">david.ponzone@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div style="word-wrap:break-word">On most low-end routers, the NAT table will expire UDP translations after 60 sec.<div>Did you configure your phones to send a NAT keep-alive every X seconds, with X < 60 ?</div><div>You can also use sip-force-expires on the FS side.</div>
<div><br></div><div>In your Asterisk config, do you use qualify=yes ?</div><div><br></div><div><div><span style="border-collapse:separate;color:rgb(0, 0, 0);font-family:Helvetica;font-size:14px;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:normal;text-align:auto;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><div style="word-wrap:break-word">
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<div><font face="'Helvetica Neue'"><font color="#1C00FF">David Ponzone </font><font color="#000000" size="3"><span style="font-size:12px">Direction Technique</span></font></font></div><div><font face="'Helvetica Neue'"><font size="3"><span style="font-size:13px">email: <a href="mailto:david.ponzone@ipeva.fr" target="_blank">david.ponzone@ipeva.fr</a></span></font></font></div>
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<div style="text-decoration:underline;text-align:justify"><font color="#C0C0C0"><i><br></i></font></div></span></div></span></font></div></span></font></div></div></span><br></div></span><br> </div><br><div><div>Le 08/04/2010 à 23:00, Aloysius Lloyd a écrit :</div>
<br><blockquote type="cite"><div><div></div><div>Hi All,<div><br></div><div>I am having lots of problem with NAT and FreeSWITCH.</div><div><br></div><div>FreeeSWITCH running Public IP.</div><div><br></div><div>
Phones Connecting to FreeeSWITCH through NAT. Cable Provider Router/Modem : SMC Gateway.</div> <div><br></div><div>The Phone Losing the Registrations. The Environment working for Asterisk without any problem.</div><div>
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</div><div>Please note I am not comparing Asterisk Vs FreeSWITCH. I am trying make FreeSWITCH work.</div> <div><br></div><div>How to solve this problem.</div><div><br clear="all">Thanks,<br>Lloyd </div></div></div><div>
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