[bug]: queue member->lastcall reset without reason

[ruslan-voip]

Severity

Trivial

Versions

20.6.0

Components/Modules

apps/app_queue.c

Operating Environment

any

Frequency of Occurrence

Constant

Issue Description

queue members are visible in log output

call scenario

• incoming 1st call into queue1
• 1st call is answered by member PJSIP/6001
• PJSIP/6001 does attended transfer of 1st call into queue2 that is 2nd call
• 2nd call is answered by member PJSIP/6002
• PJSIP/6001 has finished attended transfer
• you can see in log output that in both queues were set lastcall=0
• 2nd call is still running

; queue config
; it does not matter if shared_lastcall= yes|no 
[queue1]
musicclass = default
announce = queue-markq
strategy = ringall
wrapuptime= 10

[queue2]
musicclass = default
announce = queue-markq
strategy = ringall
wrapuptime= 50

it is a problem because wrapup time does not work properly in this case

Relevant log output

CLI> queue show
queue1 has 0 calls (max unlimited) in 'ringall' strategy (4s holdtime, 9s talktime), W:0, C:6, A:2, SL:0.0%, SL2:0.0% within 0s
   Members:
      PJSIP/6001 (ringinuse enabled) (dynamic) (in call) (In use) has taken 6 calls (last was 144 secs ago) (login was 621 secs ago)
   No Callers

queue2 has 0 calls (max unlimited) in 'ringall' strategy (11s holdtime, 6s talktime), W:0, C:2, A:1, SL:0.0%, SL2:0.0% within 0s
   Members:
      PJSIP/6002 (ringinuse enabled) (dynamic) (in call) (In use) has taken 2 calls (last was 59 secs ago) (login was 621 secs ago)
   No Callers

    -- Stopped music on hold on PJSIP/6003-00000016
    -- Channel PJSIP/6001-00000018 left 'native_rtp' basic-bridge <f3060611-7d47-49ec-b1e6-bb38d163a6e2>
    -- Channel Local/_attended@transfer-00000000;2 swapped with PJSIP/6001-00000018 into 'native_rtp' basic-bridge <f3060611-7d47-49ec-b1e6-bb38d163a6e2>
  == Spawn extension (from-internal, 8502, 1) exited non-zero on 'PJSIP/6001-00000018'
    -- Channel PJSIP/6001-00000017 left 'native_rtp' basic-bridge <5415f7ee-646f-4239-87b1-d0c0c9ed4bf8>
    -- Channel Local/_attended@transfer-00000000;1 swapped with PJSIP/6001-00000017 into 'native_rtp' basic-bridge <5415f7ee-646f-4239-87b1-d0c0c9ed4bf8>
CLI> queue show
queue1 has 0 calls (max unlimited) in 'ringall' strategy (4s holdtime, 21s talktime), W:0, C:7, A:2, SL:0.0%, SL2:0.0% within 0s
   Members:
      PJSIP/6001 (ringinuse enabled) (dynamic) (Not in use) has taken 7 calls (last was 3 secs ago) (login was 634 secs ago)
   No Callers

queue2 has 0 calls (max unlimited) in 'ringall' strategy (11s holdtime, 8s talktime), W:0, C:3, A:1, SL:0.0%, SL2:0.0% within 0s
   Members:
      PJSIP/6002 (ringinuse enabled) (dynamic) (In use) has taken 3 calls (last was 3 secs ago) (login was 634 secs ago)
   No Callers

Asterisk Issue Guidelines

• Yes, I have read the Asterisk Issue Guidelines

debug_log_123456.gz

[jcolp]

Please attach a debug log according to https://docs.asterisk.org/Operation/Logging/Collecting-Debug-Information/?h=collecting with the scenario.

[ruslan-voip]

debug_log_123456.gz