[asterisk-users] Asterisk Queue's

Tobias Ahlander plyschen at gmail.com
Thu Aug 28 06:26:20 CDT 2008


Hello List,

I have a sample queue with two dynamic agents. When the first caller calls
in to the system, the first agents phone starts to ring. Then another caller
calls in to the queue, but the other phone doesn't start to ring until the
first agents pick up his queued call.

I want the second call to start ringing on the second agents phone right
away, since he's available.

Here's the output from the queue from the CLI:
kursk*CLI> queue show sales
sales        has 0 calls (max unlimited) in 'leastrecent' strategy (0s
holdtime), W:0, C:0, A:0, SL:0.0% within 0s
   Members:
      SIP/1203 (dynamic) (Not in use) has taken no calls yet
      SIP/1003 (dynamic) (Not in use) has taken no calls yet
   No Callers

And here's the output from the CLI when the calls come in:
    -- Executing [726 at default:1] NoOp("SIP/1303-092637d0", "Sales Queue") in
new stack
    -- Executing [726 at default:2] Queue("SIP/1303-092637d0",
"sales|t|||1800|") in new stack
    -- Started music on hold, class 'default', on SIP/1303-092637d0
    -- SIP/1003-0926ae50 is ringing
    -- Executing [726 at default:1] NoOp("SIP/1103-09279020", "Sales Queue") in
new stack
    -- Executing [726 at default:2] Queue("SIP/1103-09279020",
"sales|t|||1800|") in new stack
    -- Started music on hold, class 'default', on SIP/1103-09279020
    -- SIP/1003-0926ae50 answered SIP/1303-092637d0
    -- Stopped music on hold on SIP/1303-092637d0
[Aug 15 14:01:04] ERROR[20347]: chan_sip.c:3192 update_call_counter: Call to
peer '1003' rejected due to usage limit of 1
    -- Couldn't call SIP/1003
[Aug 15 14:01:04] NOTICE[20347]: cdr.c:434 ast_cdr_free: CDR on channel
'SIP/1003-09276ef8' not posted
    -- SIP/1203-09290100 is ringing
  == Spawn extension (default, 726, 2) exited non-zero on
'SIP/1303-092637d0'
    -- SIP/1203-09290100 answered SIP/1103-09279020
    -- Stopped music on hold on SIP/1103-09279020
  == Spawn extension (default, 726, 2) exited non-zero on
'SIP/1103-09279020'

Has anyone seen this problem before or have a solution on it? Is it possible
somehow to tell Asterisk to only send one queue'd call to the Agent at the
time?

Thanks,
Best regards,
Tobias
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