[asterisk-bugs] [Asterisk 0014173]: Agent shows "(In use)" and will not receive queue calls while agent is logged in waiting for queue calls (1.4.22)

Asterisk Bug Tracker noreply at bugs.digium.com
Wed Jan 14 13:54:00 CST 2009


A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=14173 
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Reported By:                nathan
Assigned To:                putnopvut
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Project:                    Asterisk
Issue ID:                   14173
Category:                   Applications/app_queue
Reproducibility:            random
Severity:                   major
Priority:                   normal
Status:                     assigned
Asterisk Version:           1.4.22 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-01-05 13:54 CST
Last Modified:              2009-01-14 13:53 CST
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Summary:                    Agent shows "(In use)" and will not receive queue
calls while agent is logged in waiting for queue calls (1.4.22)
Description: 
I have a queue with approximately 50 total agents, about 20 of which will
be logged in at any given time.  Seemingly randomly (if there's a pattern,
I haven't discovered it), sometimes after an agent completes an inbound
call the agent status (as displayed in "show queue x") changes to "(In
use)" and the agent never again receives a call until that agent logs off
and logs back into the queue.  

This will happen about 15-20 times every day with seemingly random agents
at seemingly random times.  It happens when there's only one agent logged
in or when we have a full 20 logged in.  I can't identify a pattern that
would cause this problem.

This causes a huge problem, because we are missing calls and breaking our
SLA agreements with clients even though we have enough capacity and people
sitting waiting for calls.  As a workaround we are trying to get each agent
to log off the queue and log back on the queue after each call, but they
often forget.
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---------------------------------------------------------------------- 
 (0097792) nathan (reporter) - 2009-01-14 13:53
 http://bugs.digium.com/view.php?id=14173#c97792 
---------------------------------------------------------------------- 
What's the purpose of the call-limit parameter on SIP configs?  Back when I
first started on Asterisk I inherited a 1.0 system that I cleaned up, then
migrated to 1.2, and now to 1.4, and I'm not sure what purpose the
call-limit serves in the first place, or if I need it at all.

Here's what one of my typical SIP configs looks like:

[5621]
mailbox=5621 at default
type=friend
secret=somepassword
nat=never
host=dynamic
port=5060
dtmfmode=rfc2833
callerid="John Doe" <5621>
qualify=2000
call-limit=500
progressinband=no

(If it's not really relevant to the problem, feel free to ignore this
comment or just address it later -- I'd rather not distract from the real
issue) 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-01-14 13:53 nathan         Note Added: 0097792                          
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