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<p><font face="Helvetica, Arial, sans-serif">Hello</font></p>
<p><font face="Helvetica, Arial, sans-serif"><br>
</font></p>
<p><font face="Helvetica, Arial, sans-serif">I see the following
event from the Asterisk Manager :</font></p>
<p><font face="Helvetica, Arial, sans-serif">2021-06-30 11:20:55<br>
Array<br>
(<br>
[0] => Event: PeerStatus<br>
[1] => Privilege: system,all<br>
[2] => SystemName: tstv7<br>
[3] => ChannelType: SIP<br>
[4] => Peer: SIP/testacc7700921<br>
[5] => PeerStatus: Unregistered<br>
[6] => Cause: Expired<br>
)<br>
</font></p>
<p>But I see no SIP REGISTER with SIP-header Expires:0 (so an
UNregister if you like) in my SIP debug.</p>
<p><br>
</p>
<p>What I do see is a SIP OPTION, following a SIP 200 OK (so this is
the qualify frequenty) at 11:20:45</p>
<p><br>
</p>
<p>[Jun 30 11:20:45] VERBOSE[1581] chan_sip.c: Reliably Transmitting
(NAT) to my.lo.cal.ip:55014:<br>
OPTIONS <a class="moz-txt-link-abbreviated" href="mailto:sip:testacc7700921@192.168.1.9:5060">sip:testacc7700921@192.168.1.9:5060</a> SIP/2.0<br>
Via: SIP/2.0/UDP my.aste.risk.ip:5060;branch=z9hG4bK357689ec;rport<br>
Max-Forwards: 70<br>
From: "asterisk"
<a class="moz-txt-link-rfc2396E" href="mailto:sip:asterisk@my.aste.risk.ip"><sip:asterisk@my.aste.risk.ip></a>;tag=as0da9cbe1<br>
To: <a class="moz-txt-link-rfc2396E" href="mailto:sip:testacc7700921@192.168.1.9:5060"><sip:testacc7700921@192.168.1.9:5060></a><br>
Contact: <a class="moz-txt-link-rfc2396E" href="mailto:sip:asterisk@my.aste.risk.ip:5060"><sip:asterisk@my.aste.risk.ip:5060></a><br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:0b8c7f5745051ae73193231f5a487b11@my.aste.risk.ip:5060">0b8c7f5745051ae73193231f5a487b11@my.aste.risk.ip:5060</a><br>
CSeq: 102 OPTIONS<br>
User-Agent: TSTv7<br>
Date: Wed, 30 Jun 2021 09:20:45 GMT<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY, INFO, PUBLISH, MESSAGE<br>
Supported: replaces, timer<br>
Content-Length: 0<br>
<br>
<br>
---<br>
[Jun 30 11:20:45] VERBOSE[1581] chan_sip.c: <br>
<--- SIP read from UDP:my.lo.cal.ip:55014 ---><br>
SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP
my.aste.risk.ip:5060;branch=z9hG4bK357689ec;rport=5060<br>
From: "asterisk"
<a class="moz-txt-link-rfc2396E" href="mailto:sip:asterisk@my.aste.risk.ip"><sip:asterisk@my.aste.risk.ip></a>;tag=as0da9cbe1<br>
To: <a class="moz-txt-link-rfc2396E" href="mailto:sip:testacc7700921@192.168.1.9:5060"><sip:testacc7700921@192.168.1.9:5060></a>;tag=2924269434<br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:0b8c7f5745051ae73193231f5a487b11@my.aste.risk.ip:5060">0b8c7f5745051ae73193231f5a487b11@my.aste.risk.ip:5060</a><br>
CSeq: 102 OPTIONS<br>
User-Agent: Yealink SIP-T46G 28.83.0.120<br>
Content-Length: 0<br>
</p>
<p><br>
</p>
<p><br>
</p>
<p>In sip.conf I have the following config concerning SIP
registration expiry :</p>
<p>maxexpiry=3600<br>
;minexpiry=60<br>
;defaultexpiry=120<br>
;submaxexpiry=3600<br>
;subminexpiry=60<br>
qualifyfreq=120 </p>
<p><br>
</p>
<p>So my question is : what causes the Asterisk Manager to report a
"PeerStatus: Unregistered" if I find no such data in my SIP debug
information ??<br>
</p>
<p><br>
</p>
<p><br>
</p>
<p>Kind regards.</p>
<p><br>
</p>
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