[asterisk-bugs] [Asterisk 0011999]: memory leak
noreply at bugs.digium.com
noreply at bugs.digium.com
Wed Mar 5 23:57:17 CST 2008
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=11999
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Reported By: destiny6628
Assigned To:
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Project: Asterisk
Issue ID: 11999
Category: Core-General
Reproducibility: always
Severity: major
Priority: normal
Status: feedback
Asterisk Version: 1.4.18
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 02-15-2008 01:20 CST
Last Modified: 03-05-2008 23:57 CST
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Summary: memory leak
Description:
hi we are using asterisk 1.4.18 , zaptel-1.4.8 and libpri 1.4.3 with 4 e1
card on ibm x3200 server and 90 active calls running on asterisk cli and
after some hours of dialing , swap memory starts getting used and asterisk
gives core dump .
Asterisk is running with safe_asterisk.
vmstat ouput is as follows
[root at localhost cron]# vmstat
procs -----------memory---------- ---swap-- -----io---- --system--
-----cpu------
r b swpd free buff cache si so bi bo in cs us sy id
wa st
1 0 568 57308 44708 1848488 0 0 10 119 1293 509 2 2
95 0 0
Free -m is as follows
[root at localhost cron]# free -m
total used free shared buffers cached
Mem: 2025 1974 51 0 43 1806
-/+ buffers/cache: 124 1901
Swap: 1983 0 1983
OPERATING SYSTEM is centos 5 and kernel version is [root at localhost cron]#
uname -a
Linux localhost.localdomain 2.6.18-8.el5 http://bugs.digium.com/view.php?id=1
SMP Thu Mar 15 19:57:35 EDT
2007 i686 i686 i386 GNU/Linux
Cannot do valgrind test as it put heavy load on the server and its a
production server .
RAM On the server is 2GB .
Only Asterisk application is running on that .
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destiny6628 - 03-05-08 23:57
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hi thanks for your prompt reply and appreciate you have taken a time out
and looked into our issue .
I ll explain you why we are using local channels and our dial plan as well
.
We are using a dialer application where we dial out the local channels and
when sip channels gets formed we transfer the call to the executives.
Local channel we are using because we send monitor and then change monitor
is required for recording the calls .
exten => _91.,1,AGI(call_log,10.0.0.5|10.0.0.190)
exten => _91.,2,GotoIf($["${CALLFILENAME}" != ""]?3:4)
exten =>
_91.,3,monitor(wav,/var/spool/asterisk/monitor/ORIG/${STRFTIME(${EPOCH},Asia/Calcutta,%Y%m%d)}/${CALLFILENAME})
exten => _91.,4,Dial(Zap/g1/${EXTEN:3},,o|)
exten => _91.,5,Hangup
Caller id is also required for transferring the call to the agents thats
why we are using ,,o as well .
At the starting local channel gets formed and after that SIP or ZAP
channel gets takes place and thats why we are using local channel .
We are using httpd and php for the web interface and other pages for
dialing and all .
------------------------------------
Today also in the morning we had one core dump but thats was not due to
hangup on local channel but due to
Core was generated by `/usr/sbin/asterisk -vvvg -c'.
Program terminated with signal 11, Segmentation fault.
http://bugs.digium.com/view.php?id=0 0x08099217 in do_devstate_changes
(data=0x0) at devicestate.c:365
365 cur = AST_LIST_REMOVE_HEAD(&state_changes, list);
(gdb) bt
http://bugs.digium.com/view.php?id=0 0x08099217 in do_devstate_changes
(data=0x0) at devicestate.c:365
http://bugs.digium.com/view.php?id=1 0x081014c2 in dummy_start (data=0x886adb8)
at utils.c:865
http://bugs.digium.com/view.php?id=2 0x007f545b in start_thread () from
/lib/libpthread.so.0
http://bugs.digium.com/view.php?id=3 0x0095624e in clone () from /lib/libc.so.6
Attaching the core dump as well by the name of core.txt
Will also upload the console output as well if again asterisk gets crashed
.
Thanks again for your response .
Issue History
Date Modified Username Field Change
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03-05-08 23:57 destiny6628 Note Added: 0083526
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