[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 Aug 22 15:22:25 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:
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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: 08-22-2007 15:22 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 - 08-22-07 15:22
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AFAIK no.
I had the same problem on another server running 1.4.10.1. When I disabled
chan_mobile, it worked great (about 300 calls IAX, SIP, ISDN CAPI) in one
day
of testing without problem.
When I deleted mixmonitor from all macros and re-enabled chan_mobile, it
works there.
When I tried to trace this issue I found following:
- It seems that only incoming calls are affected. But I need to
double-check it.
- When I used Nokia6230i, it crashed just after answer, but with Nokia6234
it crashed on next priority in dialplan. Sometimes I was able to reach
priority 7 (dialing SIP extension).
My dialplan:
[incoming-mobile]
exten => sms,1,Verbose(Incoming SMS from ${SMSSRC} ${SMSTXT})
exten => sms,n,Hangup()
exten => s,1,Wait(1)
exten => s,2,Answer
exten => s,3,MixMonitor(rec${TIMESTAMP}-${UNIQUEID}.wav)
exten => s,4,Playback(letters/c)
exten => s,5,Playback(letters/h)
exten => s,6,Playback(letters/l)
exten => s,7,Dial(SIP/666)
exten => s,8,Hangup()
gdb backtrace seems to be identical in both cases (N6230i, N6234 which
crashes later).
There might be also some problem with audio format conversion, since
playback letters in example above are in gsm format, and mixmonitor is
requested to use wav format. However, when I change .wav to .gsm nothing
changes.
Issue History
Date Modified Username Field Change
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08-22-07 15:22 phokz Note Added: 0069272
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