<html><body bgcolor="#FFFFFF"><div>There is a fix <a href="https://issues.asterisk.org/view.php?id=19318"><a href="https://issues.asterisk.org/view.php?id=19318">https://issues.asterisk.org/view.php?id=19318</a></a><br><br><div>--</div>Sent from my iPhone</div><div><br>On May 20, 2011, at 4:40 PM, satish patel <<a href="mailto:satish_lx@hotmail.com">satish_lx@hotmail.com</a>> wrote:<br><br></div><div></div><blockquote type="cite"><div>
Hey Eric,<br><br>I do have qualify=yes. Am i missing something ?<br><br>[seb-exten](!) ; Template<br>type=friend<br>host=dynamic<br>context=from-sip<br>qualify=yes<br>dtmfmode=rfc2833<br>nat=no<br>cc_agent_policy=generic<br>cc_monitor_policy=generic<a title="Send (Ctrl+Enter)" href="javascript:;" onclick="ComposePage._sending = true;" id="SendMessage"><span class="Label"></span></a><br><br>[7022](seb-exten)<br>callerid="Rover Conference" <7022><br>accountcode="Rover Conference"<br>mailbox=7022@default<br><br>[7023](seb-exten)<br>callerid="Faire Conference" <7023><br>accountcode="Faire Conference"<br>mailbox=7023@default<br><br><br><br>> From: <a href="mailto:EWieling@nyigc.com"><a href="mailto:EWieling@nyigc.com">EWieling@nyigc.com</a></a><br>> To: <a href="mailto:asterisk-users@lists.digium.com"><a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a></a><br>> Date: Fri, 20 May 2011 15:15:45 -0400<br>> Subject: Re: [asterisk-users] Restart asterisk destroy all registered SIP peers<br>> <br>> <br>> <br>> > -----Original Message-----<br>> > From: <a href="mailto:asterisk-users-bounces@lists.digium.com"><a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a></a><br>> > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of<br>> > satish patel<br>> > Sent: Friday, May 20, 2011 3:10 PM<br>> > To: asterisk-users<br>> > Subject: Re: [asterisk-users] Restart asterisk destroy all<br>> > registered SIP peers<br>> ><br>> > Issue is we are running customer support queue and if by<br>> > chance if i need to restart asterisk then they will not able<br>> > to get call until phone get register :( Let me check polycom<br>> > default timeout and set to min.<br>> <br>> Asterisk should cache the registrations across a restart and reboot. I belive this feature was added in 1.4.<br>> <br>> You should not need to set a low registration timeout. If you set it because of NAT issues, setting qualify=yes will keep the translations open.<br>> <br>> --<br>> _____________________________________________________________________<br>> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</a> --<br>> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> <a href="http://www.asterisk.org/hello"><a href="http://www.asterisk.org/hello">http://www.asterisk.org/hello</a></a><br>> <br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users"><a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a></a><br>
</div></blockquote><blockquote type="cite"><div><span>--</span><br><span>_____________________________________________________________________</span><br><span>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</a> --</span><br><span>New to Asterisk? Join us for a live introductory webinar every Thurs:</span><br><span> <a href="http://www.asterisk.org/hello"><a href="http://www.asterisk.org/hello">http://www.asterisk.org/hello</a></a></span><br><span></span><br><span>asterisk-users mailing list</span><br><span>To UNSUBSCRIBE or update options visit:</span><br><span> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users"><a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a></a></span></div></blockquote></body></html>