[asterisk-users] Members in 'Unknown' status in output of 'queue show'

[James FitzGibbon]

Does anyone know what can cause queue members to go into a status of
"Unknown"?

pbxtel-01*CLI> queue show

cs        has 2 calls (max unlimited) in 'rrmemory' strategy (24s holdtime),
W:0, C:447, A:20, SL:91.7% within 60s
   Members:
      SIP/1405 (dynamic) (Unknown) has taken no calls yet
      SIP/1420 (dynamic) (paused) (Not in use) has taken no calls yet
      SIP/1442 (dynamic) (paused) (Unknown) has taken 2 calls (last was 101
secs ago)
      SIP/1440 (dynamic) (In use) has taken 2 calls (last was 3071 secs ago)
      SIP/1428 (dynamic) (paused) (Not in use) has taken 2 calls (last was
10818 secs ago)
      SIP/1404 (dynamic) (paused) (Not in use) has taken 2 calls (last was
2228 secs ago)
      SIP/1429 (dynamic) (paused) (Unknown) has taken 2 calls (last was 953
secs ago)
      SIP/1432 (dynamic) (Unavailable) has taken 5 calls (last was 1229 secs
ago)
      SIP/1430 (dynamic) (In use) has taken 2 calls (last was 22744 secs
ago)
      SIP/1435 (dynamic) (In use) has taken 3 calls (last was 13511 secs
ago)
      SIP/1434 (dynamic) (Unknown) has taken 6 calls (last was 9504 secs
ago)
      SIP/1424 (dynamic) (In use) has taken 4 calls (last was 16373 secs
ago)
      SIP/1408 (dynamic) (paused) (Not in use) has taken 2 calls (last was
8685 secs ago)
      SIP/1203 (dynamic) (In use) has taken 3 calls (last was 16425 secs
ago)
      SIP/1410 (dynamic) (Unknown) has taken 2 calls (last was 8629 secs
ago)
   Callers:
      1. Zap/50-1 (wait: 11:15, prio: 0)
      2. Zap/36-1 (wait: 0:41, prio: 0)

That's just one queue, but I had nearly all my agents just go into Unknown
status.  This is on * 1.4.10.1.  I had this happen once in the past, but
couldn't reproduce it in the lab.

When this happens, 'ringinuse=no' stops working, because app_queue considers
"Unknown" to be a valid state to dispatch a caller to.  So my agents start
getting flooded with calls while already on the phone, then the call-limit
I've configured in sip.conf kicks in and my console fills up with this:

pbxtel-01*CLI>
[Aug 29 16:44:04] ERROR[22621]: chan_sip.c:3169 update_call_counter: Call to
peer '1405' rejected due to usage limit of 2

pbxtel-01*CLI>
[Aug 29 16:44:04] ERROR[22621]: chan_sip.c:3169 update_call_counter: Call to
peer '1410' rejected due to usage limit of 2

pbxtel-01*CLI>
[Aug 29 16:44:04] ERROR[22762]: chan_sip.c:3169 update_call_counter: Call to
peer '1405' rejected due to usage limit of 2

pbxtel-01*CLI>
[Aug 29 16:44:04] ERROR[22762]: chan_sip.c:3169 update_call_counter: Call to
peer '1410' rejected due to usage limit of 2

pbxtel-01*CLI>
[Aug 29 16:44:04] ERROR[22686]: chan_sip.c:3169 update_call_counter: Call to
peer '1405' rejected due to usage limit of 2

pbxtel-01*CLI>
[Aug 29 16:44:04] ERROR[22686]: chan_sip.c:3169 update_call_counter: Call to
peer '1410' rejected due to usage limit of 2

I had to restart Asterisk to clear the states - sip reloads, app_queue
reloads didn't do anything.

Any thoughts as to where to start debugging this?  I killed * instead of
stopping it so that I got a core file.  There is nothing in the log to
indicate what went wrong prior to the first instance of "...rejected due to
usage limit".

Anything else I should gather before submitting a bug?

Thanks

-- 
j.
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