[asterisk-bugs] [JIRA] (ASTERISK-25498) Asterisk crashes when negotiating g729 without that module installed
Ben Klang (JIRA)
noreply at issues.asterisk.org
Mon Nov 2 10:08:33 CST 2015
[ https://issues.asterisk.org/jira/browse/ASTERISK-25498?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=228099#comment-228099 ]
Ben Klang commented on ASTERISK-25498:
--------------------------------------
Unfortunately, it looks like the issue persists with 13.6.0.
Final few lines in the Asterisk log:
{code}
[Nov 2 10:56:47] VERBOSE[14913][C-0000dc1e] app_amd.c: AMD: SIP/lsi-00012dd7 8329407074 (N/A) (Fmt: ulaw)
[Nov 2 10:56:47] VERBOSE[14913][C-0000dc1e] app_amd.c: AMD: initialSilence [2500] greeting [1500] afterGreetingSilence [600] totalAnalysisTime [3500] minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [3] silenceThreshold [256] maximumWordLength [5000]
[Nov 2 10:56:47] VERBOSE[14913][C-0000dc1e] app_amd.c: AMD: Channel [SIP/lsi-00012dd7]. Changed state to STATE_IN_SILENCE
[Nov 2 10:56:47] WARNING[14545][C-0000dbe6] translate.c: No translator path: (starting codec is not valid)
[Nov 2 10:56:47] WARNING[14785][C-0000dbe6] channel.c: Unable to find a codec translation path: (g729) -> (ulaw)
[Nov 2 10:56:47] VERBOSE[14785][C-0000dbe6] bridge_channel.c: Channel SIP/331-00012d86 left 'simple_bridge' basic-bridge <f138706b-58df-4c1d-91db-28c070e9e352>
[Nov 2 10:56:47] VERBOSE[14545][C-0000dbe6] bridge_channel.c: Channel SIP/lsi-00012d92 left 'simple_bridge' basic-bridge <f138706b-58df-4c1d-91db-28c070e9e352>
{code}
{code}
(gdb) bt
#0 0x00007fca1601f181 in ast_translate (path=0x0, f=f at entry=0x7fc8e80a60a0, consume=consume at entry=0) at translate.c:582
#1 0x00007fca15ee3430 in audiohook_list_translate_to_slin (frame=0x7fc8e80a60a0, direction=AST_AUDIOHOOK_DIRECTION_READ, audiohook_list=0x7fc8e83e4db0) at audiohook.c:837
#2 audio_audiohook_write_list (frame=0x7fc8e80a60a0, direction=AST_AUDIOHOOK_DIRECTION_READ, audiohook_list=0x7fc8e83e4db0, chan=0x7fc8e82f8448) at audiohook.c:931
#3 ast_audiohook_write_list (chan=chan at entry=0x7fc8e82f8448, audiohook_list=0x7fc8e83e4db0, direction=direction at entry=AST_AUDIOHOOK_DIRECTION_READ,
frame=frame at entry=0x7fc8e80a60a0) at audiohook.c:1086
#4 0x00007fca15f2bfe1 in __ast_read (chan=chan at entry=0x7fc8e82f8448, dropaudio=dropaudio at entry=0) at channel.c:4170
#5 0x00007fca15f2d507 in ast_read (chan=chan at entry=0x7fc8e82f8448) at channel.c:4277
#6 0x00007fc95b9015e8 in async_agi_read_frame (chan=0x7fc8e82f8448) at res_agi.c:1379
#7 launch_asyncagi (efd=0x0, argv=<optimized out>, argc=<optimized out>, chan=0x7fc8e82f8448) at res_agi.c:1576
#8 launch_script (opid=<synthetic pointer>, efd=0x0, fds=0x7fc8da3e3570, argv=<optimized out>, argc=<optimized out>, script=<optimized out>, chan=0x7fc8e82f8448)
at res_agi.c:1841
#9 agi_exec_full (chan=0x7fc8e82f8448, data=<optimized out>, enhanced=<optimized out>, dead=0) at res_agi.c:4145
#10 0x00007fca15fbcf45 in pbx_exec (c=c at entry=0x7fc8e82f8448, app=app at entry=0x7fca17271960, data=data at entry=0x7fc8da3e4bd0 "agi:async") at pbx.c:1720
#11 0x00007fca15fc9de8 in pbx_extension_helper (c=c at entry=0x7fc8e82f8448, con=con at entry=0x0, context=0x7fc8e82f8e00 "adhearsion-redirect", exten=exten at entry=0x7fc8e82f8e50 "1",
priority=priority at entry=1, label=label at entry=0x0, callerid=callerid at entry=0x7fc8e853f990 "8479165363", action=action at entry=E_SPAWN, found=found at entry=0x7fc8da3e6c7c,
combined_find_spawn=combined_find_spawn at entry=1) at pbx.c:4992
#12 0x00007fca15fcbe8f in ast_spawn_extension (combined_find_spawn=1, found=0x7fc8da3e6c7c, callerid=0x7fc8e853f990 "8479165363", priority=1, exten=0x7fc8e82f8e50 "1",
context=<optimized out>, c=0x7fc8e82f8448) at pbx.c:6162
#13 __ast_pbx_run (c=c at entry=0x7fc8e82f8448, args=args at entry=0x0) at pbx.c:6579
#14 0x00007fca15fcd3d3 in ast_pbx_run_args (c=c at entry=0x7fc8e82f8448, args=args at entry=0x0) at pbx.c:6947
#15 0x00007fca15fcd89d in ast_pbx_run (c=0x7fc8e82f8448) at pbx.c:6956
#16 pbx_outgoing_exec (data=data at entry=0x7fc8e8020618) at pbx.c:10491
#17 0x00007fca16023bfe in dummy_start (data=<optimized out>) at utils.c:1237
#18 0x00007fca13f0f182 in start_thread (arg=0x7fc8da3e7700) at pthread_create.c:312
#19 0x00007fca134eb47d in clone () at ../sysdeps/unix/sysv/linux/x86_64/clone.S:111
{code}
> Asterisk crashes when negotiating g729 without that module installed
> --------------------------------------------------------------------
>
> Key: ASTERISK-25498
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-25498
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Codecs/General
> Affects Versions: 13.4.0
> Environment: $ lscpu
> Architecture: x86_64
> CPU op-mode(s): 32-bit, 64-bit
> Byte Order: Little Endian
> CPU(s): 8
> On-line CPU(s) list: 0-7
> Thread(s) per core: 1
> Core(s) per socket: 4
> Socket(s): 2
> NUMA node(s): 1
> Vendor ID: GenuineIntel
> CPU family: 6
> Model: 45
> Stepping: 2
> CPU MHz: 2300.000
> BogoMIPS: 4600.00
> Hypervisor vendor: VMware
> Virtualization type: full
> L1d cache: 32K
> L1i cache: 32K
> L2 cache: 256K
> L3 cache: 20480K
> NUMA node0 CPU(s): 0-7
> $ free -m
> total used free shared buffers cached
> Mem: 12010 11670 339 0 174 8262
> -/+ buffers/cache: 3233 8777
> Swap: 4092 26 4066
> $ uname -mrs
> Linux 3.13.0-62-generic x86_64
> $ lsb_release -a
> No LSB modules are available.
> Distributor ID: Ubuntu
> Description: Ubuntu 14.04.3 LTS
> Release: 14.04
> Codename: trusty
> Reporter: Ben Langfeld
> Assignee: Ben Langfeld
> Attachments: backtrace.txt
>
>
> Regularly the following log lines are followed by Asterisk crashing (backtrace attached):
> ```
> [Oct 19 11:17:43] WARNING[15022][C-0001f68d] translate.c: No translator path: (starting codec is not valid)
> [Oct 19 11:17:43] WARNING[15051][C-0001f68d] channel.c: Unable to find a codec translation path: (g729) -> (ulaw)
> ```
> I do not see any relevant changes between 13.4.0 and 13.6.0, but I am in the process of upgrading anyway to confirm this issue is still present.
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