[asterisk-bugs] [Asterisk 0017358]: problem with ringinuse=no, queue members receive sometimes two calls
Asterisk Bug Tracker
noreply at bugs.digium.com
Fri May 21 06:51:59 CDT 2010
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=17358
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Reported By: nik600
Assigned To:
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Project: Asterisk
Issue ID: 17358
Category: Applications/app_queue
Reproducibility: sometimes
Severity: major
Priority: normal
Status: acknowledged
Asterisk Version: 1.4.31
JIRA: SWP-1512
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
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Date Submitted: 2010-05-19 02:52 CDT
Last Modified: 2010-05-21 06:51 CDT
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Summary: problem with ringinuse=no, queue members receive
sometimes two calls
Description:
Dear all
on a debian amd64 i've installed (from source) asterisk 1.4.31
On the system we have in average 50 concurrent calls in queue and 40
sip members.
I'm experiencing an apparently random problem:
sometimes some users receive 2 calls from asterisk, apparently
ignoring the ringinuse=no settings.
It appears on users that are members of many queues
As you can see from the log, the user goes in a status Ring+Inuse.
Any idea?
Why the call is still dispatched to the user if it is not in the "Not
in use" status?
i've added some customized log in the ring_entry function and this is the
result:
[May 18 14:13:04] DEBUG[24945] app_queue.c: KUMBELOG: queue=queue_1
count=1,membercount=13,ringinuse=0,device=SIP/PL1009,status=1
[May 18 14:13:04] DEBUG[24945] app_queue.c: Found matching member
SIP/PL1009 in queue 'queue_2'
[May 18 14:13:04] VERBOSE[24945] logger.c: -- Called SIP/PL1009
[May 18 14:13:05] VERBOSE[24945] logger.c: -- SIP/PL1009-00001807 is
ringing
[May 18 14:13:06] DEBUG[25098] app_queue.c: KUMBELOG: queue=queue_2
count=2,membercount=15,ringinuse=0,device=SIP/PL1009,status=1
[May 18 14:13:06] DEBUG[25098] app_queue.c: Found matching member
SIP/PL1009 in queue 'queue_1'
[May 18 14:13:06] DEBUG[25098] app_queue.c: Found matching member
SIP/PL1009 in queue 'queue_3'
[May 18 14:13:06] VERBOSE[25098] logger.c: -- Called SIP/PL1009
[May 18 14:13:07] VERBOSE[25098] logger.c: -- SIP/PL1009-00001808 is
ringing
[May 18 14:13:07] DEBUG[25312] app_queue.c: KUMBELOG: queue=queue_3
count=1,membercount=18,ringinuse=0,device=SIP/PL1009,status=6
[May 18 14:13:08] DEBUG[25382] app_queue.c: KUMBELOG: queue=queue_4
count=1,membercount=18,ringinuse=0,device=SIP/PL1009,status=6
[May 18 14:13:08] DEBUG[25224] app_queue.c: KUMBELOG: queue=queue_2
count=2,membercount=15,ringinuse=0,device=SIP/PL1009,status=6
[May 18 14:13:12] VERBOSE[25098] logger.c: -- SIP/PL1009-00001808
answered SIP/192.168.55.32-000017e6
[May 18 14:13:13] VERBOSE[25098] logger.c: -- Native bridging
SIP/192.168.55.32-000017e6 and SIP/PL1009-00001808
[May 18 14:13:14] DEBUG[25224] app_queue.c: KUMBELOG: queue=queue_2
count=1,membercount=15,ringinuse=0,device=SIP/PL1009,status=7
It seems that the system does not change the status of the user after
calling it, and then re-schedule a new call.
After that the status is updated and goes in a ring+inuse status (7)
Do you have any idea about what can cause that?
This is an example of my config
[PL1009]
context=mycontext
callerid=PhoneLine1009 <1009>
secret=pwd1009
type=peer
host=dynamic
call-limit=3
disallow=all
allow=ulaw
queues:
[queue_1]
weight=10
wrapuptime=0
strategy=leastrecent
joinempty=no
retry=0
autopause=yes
setinterfacevar=yes
eventwhencalled=yes
eventmemberstatus=yes
ringinuse=no
member => SIP/PL1009
[queue_2]
weight=10
wrapuptime=0
strategy=leastrecent
joinempty=no
retry=0
autopause=yes
setinterfacevar=yes
eventwhencalled=yes
eventmemberstatus=yes
ringinuse=no
member => SIP/PL1009
[queue_3]
weight=10
wrapuptime=0
strategy=leastrecent
joinempty=no
retry=0
autopause=yes
setinterfacevar=yes
eventwhencalled=yes
eventmemberstatus=yes
ringinuse=no
member => SIP/PL1009
======================================================================
----------------------------------------------------------------------
(0122239) nik600 (reporter) - 2010-05-21 06:51
https://issues.asterisk.org/view.php?id=17358#c122239
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cristiandimache, have you tried the
internal_timing = yes
option in asterisk.conf ?
Issue History
Date Modified Username Field Change
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2010-05-21 06:51 nik600 Note Added: 0122239
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