Asterisk queue not forwarding call to extension

dms*CLI> queue show software
software has 1 calls (max unlimited) in 'rrmemory' strategy (0s holdtime, 0s talktime), W:0, C:0, A:1, SL:0.0%, SL2:0.0% within 0s
   Members:
      Agent 5001 (PJSIP/5001) (ringinuse disabled) (realtime) (Unavailable) has taken no calls yet (login was 229 secs ago)
      Agent 6002 (PJSIP/6002) (ringinuse disabled) (realtime) (Not in use) has taken no calls yet (login was 229 secs ago)
      Agent 6001 (PJSIP/6001) (ringinuse disabled) (realtime) (Not in use) has taken no calls yet (login was 229 secs ago)
   Callers:
      1. PJSIP/my_sip_trunk-00000003 (wait: 2:54, prio: 0)

dms*CLI>

Why call not connect with 6001 or 6002? what is the reason?

Are you able to dial between extensions directly, without the queue ?

It might help to post your /etc/asterisk/queues.conf file.

When I dial directly, It was succeeded. But when I using queue, 1 st call successfully transfer to agent, but from next call, it will not transferred.

This is my CLI when has a call

software has 1 calls (max unlimited) in 'leastrecent' strategy (7s holdtime, 2s talktime), W:0, C:1, A:0, SL:0.0%, SL2:0.0% within 0s
   Members:
      PJSIP/6002 (ringinuse disabled) (Unavailable) has taken no calls yet (login was 228 secs ago)
      PJSIP/6001 (ringinuse disabled) (Not in use) has taken 1 calls (last was 54 secs ago) (login was 228 secs ago)
   Callers:
      1. PJSIP/my_sip_trunk-00000002 (wait: 0:09, prio: 0)

And this is the queue.conf file

[general]
persistentmembers = yes
monitor-type = MixMonitor

[software]

context = from_external
strategy = leastrecent
timeout = 15
retry = 1
wrapuptime = 300
ringinuse = no
leavewhenempty = no
joinempty = yes

member => PJSIP/6001
member => PJSIP/6002

I solved this. After reducing wrapuptime from 300 to 10, it was succeeded.

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