Hi Lloyd, as you are using Aastra phone 9143i behind NAT, plesse enable &quot;rport&quot; under &quot;Network&quot; settings of Aastra phone,and restart the phone, most likely you should be able to bet back to work unless your SMC Gateway really sucks.<br>
Please advise if this helps.<br><br>Chris<br><br><br><div class="gmail_quote">On Fri, Apr 9, 2010 at 3:40 PM, Aloysius Lloyd <span dir="ltr">&lt;<a href="mailto:lloyd.aloysius@gmail.com">lloyd.aloysius@gmail.com</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">I could not find a SIP ALG Setting. I setup the <span style="font-family: arial,sans-serif; font-size: 13px; border-collapse: collapse;">sip-force-expires and ping for the user directory. Only one time registering then lost the connection.</span><div>

<font face="arial, sans-serif"><span style="border-collapse: collapse;"><br></span></font></div><div><font face="arial, sans-serif"><span style="border-collapse: collapse;">Here is the sofia profile internal status</span></font></div>

<div><font face="arial, sans-serif"><span style="border-collapse: collapse;"><br></span></font></div><div><font face="arial, sans-serif"><span style="border-collapse: collapse;"><div>
Call-ID:        7307ef8fa6044407</div><div>User:           <a href="mailto:202@abc.com" target="_blank">202@abc.com</a></div><div>Contact:        &quot;Mike Derouin&quot; &lt;sip:202@A.B.C.D:5060;transport=udp;srcadr=<a href="http://192.168.1.51:5060" target="_blank">192.168.1.51:5060</a>&gt;</div>

<div>Agent:          Aastra 9143i/<a href="http://2.5.2.30" target="_blank">2.5.2.30</a></div><div>Status:         Registered(UDP)(unknown) EXP(2010-04-09 15:31:21)</div><div>Host:           TestSrv</div><div>IP:             A.B.C.D</div>

<div>Port:           5060</div><div>Auth-User:      202</div><div>Auth-Realm:     <a href="http://abc.com" target="_blank">abc.com</a></div><div>MWI-Account:    <a href="mailto:202@abc.com" target="_blank">202@abc.com</a></div>
<div><br></div><div>Please let me know how to fix this issue.</div>
<div><br></div><div>Thanks</div><div>Lloyd</div></span></font><div><div></div><div class="h5">
<br><br><div class="gmail_quote">On Fri, Apr 9, 2010 at 1:37 AM, David Ponzone <span dir="ltr">&lt;<a href="mailto:david.ponzone@gmail.com" target="_blank">david.ponzone@gmail.com</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">

<div style="word-wrap: break-word;">I dont think there is a lowest value, but 30 seconds is reasonable in most cases.<div>You can also add a ping parameter with value 30, in the user config with the variables.</div><div>
The result is a SIP OPTIONS sent to the phone every X sec.</div>
<div>That&#39;s quite the equivalent of qualify=yes in Asterisk.</div><div><div><br><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-indent: 0px; text-transform: none; white-space: normal; word-spacing: 0px;"><div style="word-wrap: break-word;">

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<div><font face="&#39;Helvetica Neue&#39;"><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="&#39;Helvetica Neue&#39;"><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>

<div><font face="&#39;Helvetica Neue&#39;"><font size="3"><span style="font-size: 13px;">tel:      01 74 03 18 97</span></font></font></div><div><font face="&#39;Helvetica Neue&#39;"><font size="3"><span style="font-size: 13px;">gsm:   06 66 98 76 34</span></font></font></div>

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<div style="margin: 0px; color: rgb(0, 34, 243);"><span style="text-decoration: underline;"><br></span></div><div style="margin: 0px; color: rgb(0, 34, 243);">
<span><div style="margin: 0px; text-align: justify; color: rgb(192, 192, 192);"><i>Ce message et toutes les pièces jointes sont confidentiels et établis à l&#39;intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d&#39;altération. </i><b><i>IPeva</i></b><i> décline toute responsabilité au titre de ce message s&#39;il a été altéré, déformé ou falsifié. Si vous n&#39;êtes pas destinataire de ce message, merci de le détruire immédiatement et d&#39;avertir l&#39;expéditeur.</i></div>

<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><div>
Le 09/04/2010 à 03:31, Aloysius Lloyd a écrit :</div>
<div><div></div><div><br><blockquote type="cite">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">&lt;<a href="mailto:lloyd.aloysius@gmail.com" target="_blank">lloyd.aloysius@gmail.com</a>&gt;</span> wrote:<br>

<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; 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> <div></div><div><div> <br></div><div><br> <br><br><div class="gmail_quote">On Thu, Apr 8, 2010 at 7:23 PM, David Ponzone <span dir="ltr">&lt;<a href="mailto:david.ponzone@gmail.com" target="_blank">david.ponzone@gmail.com</a>&gt;</span> wrote:<br>

<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; 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 &lt; 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-indent: 0px; text-transform: none; white-space: normal; word-spacing: 0px;"><div style="word-wrap: break-word;">

 <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-indent: 0px; text-transform: none; white-space: normal; word-spacing: 0px;"><div style="word-wrap: break-word;">

 <div><font face="&#39;Helvetica Neue&#39;"><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="&#39;Helvetica Neue&#39;"><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>

 <div><font face="&#39;Helvetica Neue&#39;"><font size="3"><span style="font-size: 13px;">tel:      01 74 03 18 97</span></font></font></div><div><font face="&#39;Helvetica Neue&#39;"><font size="3"><span style="font-size: 13px;">gsm:   06 66 98 76 34</span></font></font></div>

 <div><font face="&#39;Helvetica Neue&#39;"><br></font></div><div><font color="#1c00ff" face="&#39;Helvetica Neue&#39;">Service Client<span> </span></font><font face="&#39;Helvetica Neue&#39;"><font color="#ff0000">IP</font></font><font color="#1c00ff" face="&#39;Helvetica Neue&#39;">eva</font></div>

 <div><font color="#1c00ff" face="&#39;Helvetica Neue&#39;"><span style="color: rgb(0, 0, 0); font-family: Helvetica;"><div><font face="&#39;Helvetica Neue&#39;"><font size="3"><span style="font-size: 13px;">tel:      0811 46 26 26</span></font></font></div>

 <div><font face="&#39;Helvetica Neue&#39;" size="3"><span style="font-size: 13px;"><div style="margin: 0px; color: rgb(0, 34, 243);"><span style="text-decoration: underline;"><a>www.ipeva.fr</a></span><span style="color: rgb(101, 104, 149);">  -   <span style="color: rgb(0, 34, 243); text-decoration: underline;"><a>www.ipeva-studio.com</a></span></span></div>

 <div style="margin: 0px; color: rgb(0, 34, 243);"><span style="text-decoration: underline;"><br></span></div><div style="margin: 0px; color: rgb(0, 34, 243);">
 <span><div style="margin: 0px; text-align: justify; color: rgb(192, 192, 192);"><i>Ce message et toutes les pièces jointes sont confidentiels et établis à l&#39;intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d&#39;altération. </i><b><i>IPeva</i></b><i> décline toute responsabilité au titre de ce message s&#39;il a été altéré, déformé ou falsifié. Si vous n&#39;êtes pas destinataire de ce message, merci de le détruire immédiatement et d&#39;avertir l&#39;expéditeur.</i></div>

 <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> <br> </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> _______________________________________________<br>FreeSWITCH-users mailing list<br><a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a><br>

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