[asterisk-users] Asterisk Manager gives event PeerStatus: Unregistered Cause: Expired

[Jonas Kellens]

Hello


I see the following event from the Asterisk Manager :

2021-06-30 11:20:55
Array
(
     [0] => Event: PeerStatus
     [1] => Privilege: system,all
     [2] => SystemName: tstv7
     [3] => ChannelType: SIP
     [4] => Peer: SIP/testacc7700921
     [5] => PeerStatus: Unregistered
     [6] => Cause: Expired
)

But I see no SIP REGISTER with SIP-header Expires:0 (so an UNregister if 
you like) in my SIP debug.


What I do see is a SIP OPTION, following a SIP 200 OK (so this is the 
qualify frequenty) at 11:20:45


[Jun 30 11:20:45] VERBOSE[1581] chan_sip.c: Reliably Transmitting (NAT) 
to my.lo.cal.ip:55014:
OPTIONS sip:testacc7700921 at 192.168.1.9:5060 SIP/2.0
Via: SIP/2.0/UDP my.aste.risk.ip:5060;branch=z9hG4bK357689ec;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at my.aste.risk.ip>;tag=as0da9cbe1
To: <sip:testacc7700921 at 192.168.1.9:5060>
Contact: <sip:asterisk at my.aste.risk.ip:5060>
Call-ID: 0b8c7f5745051ae73193231f5a487b11 at my.aste.risk.ip:5060
CSeq: 102 OPTIONS
User-Agent: TSTv7
Date: Wed, 30 Jun 2021 09:20:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[Jun 30 11:20:45] VERBOSE[1581] chan_sip.c:
<--- SIP read from UDP:my.lo.cal.ip:55014 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP my.aste.risk.ip:5060;branch=z9hG4bK357689ec;rport=5060
From: "asterisk" <sip:asterisk at my.aste.risk.ip>;tag=as0da9cbe1
To: <sip:testacc7700921 at 192.168.1.9:5060>;tag=2924269434
Call-ID: 0b8c7f5745051ae73193231f5a487b11 at my.aste.risk.ip:5060
CSeq: 102 OPTIONS
User-Agent: Yealink SIP-T46G 28.83.0.120
Content-Length: 0



In sip.conf I have the following config concerning SIP registration expiry :

maxexpiry=3600
;minexpiry=60
;defaultexpiry=120
;submaxexpiry=3600
;subminexpiry=60
qualifyfreq=120


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 ??



Kind regards.


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