[asterisk-bugs] [Asterisk 0010508]: Segfault when answering ringing mobile while monitoring call with mixmonitor
noreply at bugs.digium.com
noreply at bugs.digium.com
Wed Oct 17 08:07:45 CDT 2007
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=10508
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Reported By: phokz
Assigned To: dbowerman
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Project: Asterisk
Issue ID: 10508
Category: Addons/chan_mobile
Reproducibility: always
Severity: crash
Priority: normal
Status: feedback
Asterisk Version: SVN
SVN Branch (only for SVN checkouts, not tarball releases): trunk
SVN Revision (number only!): 426
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 08-21-2007 03:50 CDT
Last Modified: 10-17-2007 08:07 CDT
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Summary: Segfault when answering ringing mobile while
monitoring call with mixmonitor
Description:
When I try to answer call and playback welcome message before forwarding
to sip extension, asterisk crashes.
Note that call is monitored by mixmonitor.
Asterisk is trunk r. 80129
chan_mobile is addons trunk r. 426
System is openSUSE 10.2. Mobile is Nokia6230i.
gdb backtrace follows:
Core was generated by `asterisk -vvvc'.
Program terminated with signal 11, Segmentation fault.
http://bugs.digium.com/view.php?id=0 0x080fb4b9 in ast_slinfactory_read
(sf=0x8226178, buf=0xb62550c0,
samples=160) at slinfactory.c:134
134 memcpy(sf->hold, frame_data,
remain * sizeof(*offset));
(gdb) bt
http://bugs.digium.com/view.php?id=0 0x080fb4b9 in ast_slinfactory_read
(sf=0x8226178, buf=0xb62550c0,
samples=160) at slinfactory.c:134
http://bugs.digium.com/view.php?id=1 0x08073903 in audiohook_read_frame_both
(audiohook=0x8226120,
samples=160) at audiohook.c:192
http://bugs.digium.com/view.php?id=2 0x08073d08 in ast_audiohook_read_frame
(audiohook=0x8226120,
samples=160, direction=AST_AUDIOHOOK_DIRECTION_BOTH, format=64)
at audiohook.c:256
http://bugs.digium.com/view.php?id=3 0xb651f376 in mixmonitor_thread
(obj=0x8226120) at
app_mixmonitor.c:165
http://bugs.digium.com/view.php?id=4 0x08108635 in dummy_start (data=0x8225ee0)
at utils.c:789
http://bugs.digium.com/view.php?id=5 0xb7d86112 in start_thread () from
/lib/libpthread.so.0
http://bugs.digium.com/view.php?id=6 0xb7ba12ee in clone () from /lib/libc.so.6
Without call monitoring it seems to work. This might be an issue in
asterisk itself, not chan_mobile.
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phokz - 10-17-07 08:07
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Sorry for delay in reporting back, I finally retested this issue.
But unfortunatelly, it still crashes.
Current versions:
asterisk SVN-trunk-r82029
chan_mobile from addons-trunk-471
(gdb) bt
http://bugs.digium.com/view.php?id=0 0xb7c4746e in fread () from /lib/libc.so.6
http://bugs.digium.com/view.php?id=1 0xb662ea8d in gsm_read (s=0x8234388,
whennext=0xb6393c6c) at
format_gsm.c:72
http://bugs.digium.com/view.php?id=2 0x080a4c80 in ast_readaudio_callback
(s=0x8234388) at file.c:624
http://bugs.digium.com/view.php?id=3 0x080a7033 in ast_streamfile
(chan=0x82299f0, filename=0xb6393de0
"letters/c",
preflang=0x82297ef "en") at file.c:714
http://bugs.digium.com/view.php?id=4 0xb71935e6 in playback_exec
(chan=0x82299f0, data=0xb6397f38) at
app_playback.c:450
http://bugs.digium.com/view.php?id=5 0x080ca715 in pbx_exec (c=0x82299f0,
app=0x81bb488, data=0xb6397f38)
at pbx.c:596
http://bugs.digium.com/view.php?id=6 0x080d2c6d in pbx_extension_helper
(c=0x82299f0, con=<value optimized
out>,
context=0x8229b78 "default", exten=0x8229bc8 "s", priority=4,
label=0x0,
callerid=0x8229370 "+420776026526", action=E_SPAWN) at pbx.c:1907
http://bugs.digium.com/view.php?id=7 0x080d4a81 in __ast_pbx_run (c=0x82299f0)
at pbx.c:2398
http://bugs.digium.com/view.php?id=8 0x080d5e7e in pbx_thread (data=0x82299f0)
at pbx.c:2753
http://bugs.digium.com/view.php?id=9 0x0810c43b in dummy_start (data=0x8229160)
at utils.c:845
http://bugs.digium.com/view.php?id=10 0xb7da5112 in start_thread () from
/lib/libpthread.so.0
http://bugs.digium.com/view.php?id=11 0xb7cb02ee in clone () from /lib/libc.so.6
Issue History
Date Modified Username Field Change
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10-17-07 08:07 phokz Note Added: 0072144
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