[asterisk-bugs] [Asterisk 0014932]: asterisk-1.6.0.9-x86_64 segfaults when leaving a voicemail internally to another extension
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Jun 2 03:16:29 CDT 2009
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=14932
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Reported By: jpiszcz
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 14932
Category: Applications/app_voicemail
Reproducibility: always
Severity: block
Priority: normal
Status: confirmed
Target Version: 1.6.0.10
Asterisk Version: 1.6.0.7
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
======================================================================
Date Submitted: 2009-04-19 07:59 CDT
Last Modified: 2009-06-02 03:16 CDT
======================================================================
Summary: asterisk-1.6.0.9-x86_64 segfaults when leaving a
voicemail internally to another extension
Description:
Taking the description from the e-mail I sent:
http://www.spinics.net/lists/asterisk/msg109496.html
Hello,
Information:
gcc -v: gcc version 4.3.3 (Debian 4.3.3-3)
os: Debian/Testing
Pulled latest release from asterisk site, compiled, installed it.
I have a barebones configuration:
$ ls -l asterisk
extensions.conf
modules.conf
sip.conf
users.conf
voicemail.conf
You can see them here:
http://home.comcast.net/~jpiszcz/20090418/extensions.conf
http://home.comcast.net/~jpiszcz/20090418/modules.conf
http://home.comcast.net/~jpiszcz/20090418/sip.conf
http://home.comcast.net/~jpiszcz/20090418/users.conf
http://home.comcast.net/~jpiszcz/20090418/voicemail.conf
When I perform the following actions, asterisk segfaults:
1. Dial *61.
2. Enter password: XXXX
3. Enter 3 for advanced options.
4. Press 5 to leave a message, press * to return to the main menu.
5. Extension: 6000
6. Please leave your message after the tone, when done, please hangup or
press the pound key (it segfaults right after it says pound key)
7. Segmentation fault
== Using SIP RTP CoS mark 5
-- Executing [*61 at line1:1] VoiceMailMain("SIP/line1-01d646b0",
"6001") in new stack
-- <SIP/line1-01d646b0> Playing 'vm-password.gsm' (language 'en')
DTMF begin '1' received on SIP/line1-01d646b0
DTMF begin ignored '1' on SIP/line1-01d646b0
DTMF end '1' received on SIP/line1-01d646b0, duration 190 ms
DTMF end passthrough '1' on SIP/line1-01d646b0
/DTMF begin '2' received on SIP/line1-01d646b0
DTMF begin ignored '2' on SIP/line1-01d646b0
DTMF end '2' received on SIP/line1-01d646b0, duration 130 ms
DTMF end passthrough '2' on SIP/line1-01d646b0
DTMF begin '3' received on SIP/line1-01d646b0
DTMF begin ignored '3' on SIP/line1-01d646b0
DTMF end '3' received on SIP/line1-01d646b0, duration 130 ms
DTMF end passthrough '3' on SIP/line1-01d646b0
DTMF begin '4' received on SIP/line1-01d646b0
DTMF begin ignored '4' on SIP/line1-01d646b0
DTMF end '4' received on SIP/line1-01d646b0, duration 170 ms
DTMF end passthrough '4' on SIP/line1-01d646b0
DTMF begin '#' received on SIP/line1-01d646b0
DTMF begin ignored '#' on SIP/line1-01d646b0
DTMF end '#' received on SIP/line1-01d646b0, duration 170 ms
DTMF end passthrough '#' on SIP/line1-01d646b0
-- <SIP/line1-01d646b0> Playing 'vm-youhave.gsm' (language 'en')
-- <SIP/line1-01d646b0> Playing 'vm-no.gsm' (language 'en')
-- <SIP/line1-01d646b0> Playing 'vm-messages.gsm' (language 'en')
-- <SIP/line1-01d646b0> Playing 'vm-opts.gsm' (language 'en')
-- <SIP/line1-01d646b0> Playing 'vm-helpexit.gsm' (language 'en')
DTMF begin '3' received on SIP/line1-01d646b0
DTMF begin ignored '3' on SIP/line1-01d646b0
DTMF end '3' received on SIP/line1-01d646b0, duration 130 ms
DTMF end passthrough '3' on SIP/line1-01d646b0
-- <SIP/line1-01d646b0> Playing 'vm-leavemsg.gsm' (language 'en')
-- <SIP/line1-01d646b0> Playing 'vm-starmain.gsm' (language 'en')
-- <SIP/line1-01d646b0> Playing 'vm-leavemsg.gsm' (language 'en')
DTMF begin '5' received on SIP/line1-01d646b0
DTMF begin ignored '5' on SIP/line1-01d646b0
-- <SIP/line1-01d646b0> Playing 'vm-starmain.gsm' (language 'en')
DTMF end '5' received on SIP/line1-01d646b0, duration 170 ms
DTMF end passthrough '5' on SIP/line1-01d646b0
-- <SIP/line1-01d646b0> Playing 'vm-extension.gsm' (language 'en')
DTMF begin '6' received on SIP/line1-01d646b0
DTMF begin ignored '6' on SIP/line1-01d646b0
DTMF end '6' received on SIP/line1-01d646b0, duration 170 ms
DTMF end passthrough '6' on SIP/line1-01d646b0
DTMF begin '0' received on SIP/line1-01d646b0
DTMF begin ignored '0' on SIP/line1-01d646b0
DTMF end '0' received on SIP/line1-01d646b0, duration 130 ms
DTMF end passthrough '0' on SIP/line1-01d646b0
DTMF begin '0' received on SIP/line1-01d646b0
DTMF begin ignored '0' on SIP/line1-01d646b0
DTMF end '0' received on SIP/line1-01d646b0, duration 170 ms
DTMF end passthrough '0' on SIP/line1-01d646b0
DTMF begin '0' received on SIP/line1-01d646b0
DTMF begin ignored '0' on SIP/line1-01d646b0
DTMF end '0' received on SIP/line1-01d646b0, duration 170 ms
DTMF end passthrough '0' on SIP/line1-01d646b0
-- <SIP/line1-01d646b0> Playing 'digits/6.gsm' (language 'en')
-- <SIP/line1-01d646b0> Playing 'digits/0.gsm' (language 'en')
-- <SIP/line1-01d646b0> Playing 'digits/0.gsm' (language 'en')
-- <SIP/line1-01d646b0> Playing 'digits/0.gsm' (language 'en')
-- <SIP/line1-01d646b0> Playing 'vm-intro.gsm' (language 'en')
-- <SIP/line1-01d646b0> Playing 'beep.gsm' (language 'en')
Segmentation fault (core dumped)
Here is a backtrace:
Core was generated by `/vapp/sbin/asterisk -vvvvvvvvvvv'.
Program terminated with signal 11, Segmentation fault.
[New process 23729]
[New process 23717]
[New process 23721]
[New process 23728]
[New process 23718]
[New process 23723]
[New process 23719]
[New process 23722]
[New process 23720]
[New process 23726]
[New process 23727]
https://issues.asterisk.org/view.php?id=0 0x00000000004d8bb3 in tzload
(name=0x536b26 "posixrules",
sp=0x7f7f84076ba0, doextend=0) at stdtime/localtime.c:292
292 if ((strlen(p) + strlen(name) + 1) >=
sizeof fullname)
(gdb) bt
https://issues.asterisk.org/view.php?id=0 0x00000000004d8bb3 in tzload
(name=0x536b26 "posixrules",
sp=0x7f7f84076ba0, doextend=0) at stdtime/localtime.c:292
https://issues.asterisk.org/view.php?id=1 0x00000000004d94bf in tzparse
(name=0x7f7f8406c7a5 "",
sp=0x7f7f84076ba0,
lastditch=<value optimized out>) at stdtime/localtime.c:811
https://issues.asterisk.org/view.php?id=2 0x00000000004d9152 in tzload
(name=<value optimized out>,
sp=0x881280,
doextend=1) at stdtime/localtime.c:450
https://issues.asterisk.org/view.php?id=3 0x00000000004da92d in ast_tzset
(zone=0x7f7f741e5bf9 "UTC")
at stdtime/localtime.c:1029
https://issues.asterisk.org/view.php?id=4 0x00000000004db98c in ast_localtime
(timep=0x7f7f8407c500,
tmp=0x7f7f84076ba0, zone=0x0) at stdtime/localtime.c:1142
https://issues.asterisk.org/view.php?id=5 0x00007f7f741d332f in get_date
(s=0x7f7f84086490 "0\005\210",
len=256)
at app_voicemail.c:3788
https://issues.asterisk.org/view.php?id=6 0x00007f7f741dfb1a in leave_voicemail
(chan=0x87f5f0,
ext=<value optimized out>, options=0x7f7f84090c00) at
app_voicemail.c:4476
https://issues.asterisk.org/view.php?id=7 0x00007f7f741e1ab0 in forward_message
(chan=0x87f5f0, context=0x0,
vms=0x7f7f84090d40, sender=0x7f7f84096e10,
fmt=0x7f7f743ee4e0 "wav49|gsm|wav", flag=1, record_gain=0 '\0')
at app_voicemail.c:5608
https://issues.asterisk.org/view.php?id=8 0x00007f7f741e2ece in vm_execmain
(chan=0x87f5f0,
data=<value optimized out>) at app_voicemail.c:7999
https://issues.asterisk.org/view.php?id=9 0x00000000004aedf5 in
pbx_extension_helper (c=0x87f5f0,
con=<value optimized out>, context=0x87f848 "line1", exten=0x87f898
"*61",
priority=1, label=0x0, callerid=0x8794b0 "anonymous",
action=E_SPAWN,
found=0x7f7f8409c03c, combined_find_spawn=1) at pbx.c:942
https://issues.asterisk.org/view.php?id=10 0x00000000004b039a in __ast_pbx_run
(c=0x87f5f0, args=0x0) at
pbx.c:3614
https://issues.asterisk.org/view.php?id=11 0x00000000004b160b in pbx_thread
(data=0x536b26) at pbx.c:3974
https://issues.asterisk.org/view.php?id=12 0x00000000004e6c0c in dummy_start
(data=<value optimized out>)
at utils.c:861
https://issues.asterisk.org/view.php?id=13 0x00007f7f8b531faa in start_thread ()
from /lib/libpthread.so.0
https://issues.asterisk.org/view.php?id=14 0x00007f7f87bc62bd in clone () from
/lib/libc.so.6
https://issues.asterisk.org/view.php?id=15 0x0000000000000000 in ?? ()
(gdb)
======================================================================
----------------------------------------------------------------------
(0105875) carel (reporter) - 2009-06-02 03:16
https://issues.asterisk.org/view.php?id=14932#c105875
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The workaround I had was quite fiddly. It would only work for a little
while and then start crashing again. I've also tried upgrading Ram from
256MB to 512MB, but still same issue. I've not been able to replicate this
on our development box - both has same version of Centos, the only
difference is the underlying hardware and the VPS are slightly different -
same kernel, both on XenServer. I've also tried a range of different
version of 1.6 including latest SVN but still had the same problem.
Also worth noting that occasionally asterisk would crash for no apparent
reason - we'd have no traffic at all on the box and then suddenly it
crashes.
Something else that was weird is that I could not reload asterisk -
there'd be about a 25% chance that SIP would stop responding. I'm not sure
if it is simply a case of the sip modules failing to reload (haven't
checked to be honest), but tcpdump would show sip packets arrive, but not
go out. Initially I thought this had something to do with the alias IP I
was using so I switched it off and stopped using reload. Not sure if it is
related in any way.
Eventually resorted to downgrading to 1.4 and have not had any issues
since.
Carel
Issue History
Date Modified Username Field Change
======================================================================
2009-06-02 03:16 carel Note Added: 0105875
======================================================================
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