[asterisk-bugs] [Asterisk 0016196]: Core dump in audio_audiohook_write_list

Asterisk Bug Tracker noreply at bugs.digium.com
Mon Apr 12 22:47:05 CDT 2010


A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=16196 
====================================================================== 
Reported By:                atis
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   16196
Category:                   Core/General
Reproducibility:            sometimes
Severity:                   crash
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           SVN 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.1 
SVN Revision (number only!): 228147 
Request Review:              
====================================================================== 
Date Submitted:             2009-11-06 08:47 CST
Last Modified:              2010-04-12 22:47 CDT
====================================================================== 
Summary:                    Core dump in audio_audiohook_write_list
Description: 
This crash was introduced somewhere between 1.6.1.6 and r228147. I can
reproduce it quite easily while running automated tests.

# 0  0x0000000000446613 in audio_audiohook_write_list
(chan=0x2aaac415ae68, audiohook_list=0x2aaaac2a0728,
direction=AST_AUDIOHOOK_DIRECTION_WRITE, frame=0x2aaaac455658) at
/usr/dist/asterisk-svn-1.6.1-latest-vanilla/include/asterisk/util
s.h:270
# 1  0x0000000000446a7e in ast_audiohook_write_list (chan=0x2aaac415ae68,
audiohook_list=0x2aaaac2a0728, direction=AST_AUDIOHOOK_DIRECTION_WRITE,
frame=0x2aaaac455658) at audiohook.c:704
# 2  0x00000000004608d5 in ast_write (chan=0x2aaac415ae68, fr=0x2411e40)
at channel.c:3472
# 3  0x0000000000466055 in ast_generic_bridge (c0=0x2aaac4359818,
c1=0x2aaac415ae68, config=0x429a07b0, fo=0x4299eb38, rc=0x4299eb30,
bridge_end={tv_sec = 0, tv_usec = 0}) at channel.c:4855
# 4  0x0000000000467e9e in ast_channel_bridge (c0=0x2aaac4359818,
c1=0x2aaac415ae68, config=0x429a07b0, fo=0x4299eb38, rc=0x4299eb30) at
channel.c:5194
# 5  0x000000000049cfeb in ast_bridge_call (chan=0x2aaac4359818,
peer=0x2aaac415ae68, config=0x429a07b0) at features.c:2544
# 6  0x00002aaabc84171f in try_calling (qe=0x429a0e60, options=0x429a0db7
"", announceoverride=0x429a0db9 "", url=0x429a0db8 "", tries=0x429a1084,
noption=0x429a1080, agi=0x0, macro=0x0, gosub=0x0, ringing=0) at
app_queue.c:4058
# 7  0x00002aaabc8459cd in queue_exec (chan=0x2aaac4359818,
data=0x429a12d0) at app_queue.c:4998

====================================================================== 

---------------------------------------------------------------------- 
 (0120311) samy (reporter) - 2010-04-12 22:47
 https://issues.asterisk.org/view.php?id=16196#c120311 
---------------------------------------------------------------------- 
I believe the same issue occurs on my system when using MixMonitor and
app_swift (cepstral) combined. If I remove MixMonitor, the call is fine.

Asterisk 1.6.0.26
Linux adv 2.6.32.1-rscloud https://issues.asterisk.org/view.php?id=15 SMP Mon
Feb 22 13:22:15 UTC 2010 x86_64
x86_64 x86_64 GNU/Linux
Cepstral Swift 5.1.0

Last 2 lines in asterisk messages:
[Apr 12 20:32:00] VERBOSE[31050] logger.c:     -- Executing
[s at macro-speech:1] Swift("SIP/flowroute-1-00000006", ""this is a pure
test"") in new stack
[Apr 12 20:32:00] NOTICE[31050] app_swift.c: Text to Speak : this is a
pure test

(gdb) bt
https://issues.asterisk.org/view.php?id=0  ast_audiohook_write_list
(chan=0x1dc9620, audiohook_list=0x18e4a30, 
    direction=AST_AUDIOHOOK_DIRECTION_WRITE, frame=0x8eacc00000000001)
    at audiohook.c:713
https://issues.asterisk.org/view.php?id=1  0x000000000044a81c in ast_write
(chan=0x1dc9620, fr=0x417b0e20) at
channel.c:3528
https://issues.asterisk.org/view.php?id=2  0x00007fd291c54c94 in engine
(chan=0x1dc9620, data=0x417b3410) at
app_swift.c:402
https://issues.asterisk.org/view.php?id=3  0x000000000049dd52 in pbx_exec
(c=0x1dc9620, app=0x7fd29c0267e0,
data=0x417b3410)
    at pbx.c:951
https://issues.asterisk.org/view.php?id=4  0x00000000004a86fe in
pbx_extension_helper (c=0x1dc9620, 
    con=<value optimized out>, context=0x1dc9878 "macro-speech",
exten=0x1dc98c8 "s", 
    priority=1, label=0x0, callerid=0x1d63390 "3109990409",
action=E_SPAWN, 
    found=0x417b678c, combined_find_spawn=1) at pbx.c:3141
https://issues.asterisk.org/view.php?id=5  0x00000000004a8c50 in
ast_spawn_extension (c=0x1dc9620, 
    context=0x1 <Address 0x1 out of bounds>, 
    exten=0x8eacc00000000001 <Address 0x8eacc00000000001 out of bounds>, 
    priority=30851448, callerid=<value optimized out>, found=<value
optimized out>, 
    combined_find_spawn=1) at pbx.c:3608
https://issues.asterisk.org/view.php?id=6  0x00007fd29b3ada05 in _macro_exec
(chan=0x1dc9620,
data=0x7fd29c0ad0c0, 
    exclusive=0) at app_macro.c:336
https://issues.asterisk.org/view.php?id=7  0x000000000049dd52 in pbx_exec
(c=0x1dc9620, app=0x7fd29c017250,
data=0x417b88a0)
    at pbx.c:951
https://issues.asterisk.org/view.php?id=8  0x00000000004a86fe in
pbx_extension_helper (c=0x1dc9620, 
    con=<value optimized out>, context=0x1dc9878 "macro-speech",
exten=0x1dc98c8 "s", 
    priority=1, label=0x0, callerid=0x1d63390 "3109990409",
action=E_SPAWN, 
    found=0x417baeec, combined_find_spawn=1) at pbx.c:3141
https://issues.asterisk.org/view.php?id=9  0x00000000004ab9d6 in __ast_pbx_run
(c=0x1dc9620, args=0x0) at
pbx.c:3608
https://issues.asterisk.org/view.php?id=10 0x00000000004ad2bf in
ast_pbx_outgoing_exten (type=0x1d73f5c "SIP",
format=64, 
    data=0x1d7405c, timeout=45000, context=0x1d743ac "call-confirm", 
    exten=0x1d7435c "s", priority=1, reason=0x417bb05c, sync=2, 
    cid_num=0x1d74400 "3109990409", cid_name=0x1d74500 "3109990409",
vars=0x1d51ce0, 
    account=0x1d74600 "2", channel=0x0) at pbx.c:4010
https://issues.asterisk.org/view.php?id=11 0x00007fd29513aa4a in attempt_thread
(data=<value optimized out>)
    at pbx_spool.c:344
https://issues.asterisk.org/view.php?id=12 0x00000000004e1bbc in dummy_start
(data=<value optimized out>) at
utils.c:861
https://issues.asterisk.org/view.php?id=13 0x0000003890606617 in start_thread ()
from /lib64/libpthread.so.0
https://issues.asterisk.org/view.php?id=14 0x000000388f6d3c2d in clone () from
/lib64/libc.so.6 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-04-12 22:47 samy           Note Added: 0120311                          
======================================================================




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