Monitor realtime peers?

Asterisk Asterisk Support
1

Hi,

I cannot monitor realtime peers corectly. If a phone is disconnected from the network, I will never see it with sip show peers unless someone tries to call the phone.
I thought asterisk cached peers in memory upon registration or phone call.

How can I make sure asterisk will update the cache when the phone registers?

Here is my config for one of my peer:

Name : 01-A-1245782
Description :
Realtime peer: Yes, cached
Secret :
MD5Secret :
Remote Secret:
Context : default
Record On feature : automon
Record Off feature : automon
Subscr.Cont. :
Language : fr
Tonezone :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Named Callgr :
Nam. Pickupgr:
MOH Suggest :
Mailbox : 2740@default
VM Extension : *97
LastMsgsSent : 32767/65535
Call limit : 2147483647
Max forwards : 0
Dynamic : Yes
Callerid : “Test Asterisk” <>
MaxCallBR : 384 kbps
Expire : -1
Insecure : no
Force rport : Auto (No)
Symmetric RTP: No
ACL : No
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: -1
DirectMedia : Yes
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : Yes
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : info
Timer T1 : 500
Timer B : 32000
ToHost :
Addr->IP : (null)
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 2740
SIP Options : (none)
Codecs : (ulaw)
Codec Order : (ulaw:20)
Auto-Framing : No
Status : UNREACHABLE
Useragent :
Reg. Contact :
Qualify Freq : 60000 ms
Keepalive : 0 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No

Here is my realtime settings:

rtsavesysname = yes
limitonpeers = yes
rtcachefriends=yes
rtupdate=yes
qualifyfreq=60

2

Update:

I have just discovered that Asterisk does update the cache if the phone registers but will stop sending
sip OPTIONS packets to the phone after a sip reload.(Confirmed in wireshark)

So, if I do a sip reload and an IP phone is disconnected and no one calls the phone, asterisk will never know the phone is disconnected.

Is this normal behavior?

Should Asterisk keeps checking the phone status after a sip reload even if it did not received a registration request?