[asterisk-bugs] [Asterisk 0012127]: Using state_interface and Local channels allows several simultenous calls to be sent to agent
noreply at bugs.digium.com
noreply at bugs.digium.com
Tue Mar 11 16:16:02 CDT 2008
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=12127
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Reported By: atis
Assigned To: putnopvut
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Project: Asterisk
Issue ID: 12127
Category: Applications/app_queue
Reproducibility: sometimes
Severity: minor
Priority: normal
Status: assigned
Asterisk Version: SVN
SVN Branch (only for SVN checkouts, not tarball releases): trunk
SVN Revision (number only!): 103809
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 03-03-2008 12:26 CST
Last Modified: 03-11-2008 16:16 CDT
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Summary: Using state_interface and Local channels allows
several simultenous calls to be sent to agent
Description:
I've seen this situation on 8-CPU 3GHz Xeon, that agent gets 4 queue calls
simultaneously.
For now i've been able to avoid this by checking/setting group at
beginning of Local channel, setting call-limit=1 for SIP peers (only for
those who shouldn't receive more than 1 call simultaneously) and checking
group upon unsuccessful Dial from Local channel. Those 3 fixes together are
giving some good results for one most-problematic queue, however that's not
a permanent and stable solution. If queue is creating several channels at
the same time, on slower but multi-cored systems this could lead to race
condition between GROUP_COUNT and Set(GROUP()) in two parallel channels,
and call-limit=1 also can't always be set.
I'm not sure if this can be implemented in good way, but I think the most
appropriate solution for this would be that Queue is setting device state
of Local channel, and other queue(s) (if member is in multiple queues)
receives event from first queue's devicestate change.
My system is Asterisk 1.4.14 with backported state_interface from r103809
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atis - 03-11-08 16:16
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Currently this wouldn't generate QueueMemberStatus event if chan_sip is
unloaded.
I think it's worth keeping update_status (maybe throwing out the part with
queue/member iteration), and maybe even using it from handle_statechange.
That would allow to generate AMI event of member status change correctly at
all ocasions, plus it would go well with one of my future patches (realtime
member status).
aragon: regarding pbx-test you could try decreasing call volume going
trough (increase delay between them, increase hangup probability). i've
seen that on extremly high load Asterisk will crash nearly always because
of various locking problems. Make sure load is real, and corresponds to
your production usage :)
btw, my production is already working for 1 and half days with completely
commented out update_status line, and no status problems so far.
Issue History
Date Modified Username Field Change
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03-11-08 16:16 atis Note Added: 0083773
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