[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
Wed Jun 3 17:23:33 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: 2009-06-03 17:23 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)
======================================================================
----------------------------------------------------------------------
(0105956) slic (reporter) - 2009-06-03 17:23
https://issues.asterisk.org/view.php?id=14932#c105956
----------------------------------------------------------------------
Carel,
I'm having all sorts of weird problems with * 1.6.1.0 (straight from
download, + addons) and x86_64 (on Fedora 8): although the console never
freezes, I've seen, in the last 12 hours:
a) cdr_addon_mysql.c: Failed to insert into database: (1136) Column count
doesn't match value count at row 1. This cleared itself after restarting
Asterisk.
b) problems reloading/restarting (at a certain point, "restart now" did
nothing at all--I was still in the CLI, core show calls said one call was
active [but in fact it wasn't], had to logout and manually kill -9 the PID
of asterisk)
c) SIP stops responding after approx. 45 minutes (and less than three
processed calls), meaning that:
c1 - No SIP messages appear to be sent/received, even with SIP debug
enabled
c2 - sip show peers reports all peers are registered; however, I have a
SPA3102 that is logging to syslog and I see that it is actually complaining
that it can't register ([0]RegFail.Retry in 30 is the entry in syslog)
c3 - if I initiate a call while asterisk in "hanging", all I see is ==
Using SIP RTP CoS mark 5 (although I have verbose set to 6)--nothing else,
no progress in dialplan, and of course call doesn't go through
c4 - "reload" command does not clear this problem, once it shows up. I
have to "restart now" (and this hasn't worked every time either)
c5 - On another occasion, all peers appeared as not registered (they were
registered a few minutes before) except for one--even restarting the IP
phones didn't make them register again (this seems another instance of
problem c1)
I don't think SIP dumps will be of much help--when the problem shows up,
all sip messages simply stop! (not much to dump anymore...)
I haven't used voicemail, I have a very very simple dialplan.
All this is very strange. I was using 1.4. + addons on this machine until
last monday, and didn't have any problems. I also have 1.6.0.5 (without
addons) on another machine, running without any problems whatsoever since
January (but this is a 32-bit machine).
I know this was a bit OT, but I see a common link in the x86_64
architecture here...
Issue History
Date Modified Username Field Change
======================================================================
2009-06-03 17:23 slic Note Added: 0105956
======================================================================
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