asterisk-1.8.20.1 + chan-sccp-b-4.1-r4242 - crash

[marcelloceschia]

Reported by: Nikola Ciprich
Date: 20.02.2013
Email: [Chan-sccp-b-discussion] asterisk-1.8.20.1 + chan-sccp-b-4.1-r4242 - crash

#0  0x00007fb94c5aea40 in ?? ()
#1  0x0000000000515c39 in ast_translate (path=0x7fb94c5ac170, f=0x7fb94c599ad0, consume=0) at translate.c:370
#2  0x00000000004609a2 in ast_write (chan=0x12e9bc8, fr=0x7fb94c599ad0) at channel.c:5017
#3  0x00000000004678d0 in ast_generic_bridge (c0=0x12e6688, c1=0x12e9bc8, config=0x4080da10, fo=0x4080d510, rc=0x4080d508) at channel.c:7300
#4  0x00000000004680b7 in ast_channel_bridge (c0=0x12e6688, c1=0x12e9bc8, config=0x4080da10, fo=0x4080d510, rc=0x4080d508) at channel.c:7658
#5  0x0000000000499101 in ast_bridge_call (chan=0x12e6688, peer=0x12e9bc8, config=0x4080da10) at features.c:4120
#6  0x00007fb942974386 in try_calling (qe=0x4080f710, options=<value optimized out>, announceoverride=<value optimized out>, url=0x4080f651 "", tries=<value optimized out>, noption=<value optimized out>, agi=0x0, macro=0x0, gosub=0x0, ringing=0) at app_queue.c:5322
#7  0x00007fb94297933a in queue_exec (chan=0x12e6688, data=<value optimized out>) at app_queue.c:6273
#8  0x00000000004d8b0f in pbx_exec (c=0x12e6688, app=0x121d230, data=0x40812de0 "dvernik_tlac1,tR,,,120") at pbx.c:1446
#9  0x00000000004e29c8 in pbx_extension_helper (c=0x12e6688, con=<value optimized out>, context=0x12e6be0 "internal_phones", exten=0x12e6c30 "sw_15_9", priority=26, label=0x0, callerid=0x12f1ca0 "50", action=E_SPAWN, found=0x40814f5c, combined_find_spawn=1) at pbx.c:4489
#10 0x00000000004e2f10 in ast_spawn_extension (c=0x7fb94c5ac170, context=0x7fb94c5ac170 "МYL\271\177", exten=0x0, priority=1281305168, callerid=<value optimized out>, found=<value optimized out>, combined_find_spawn=1) at pbx.c:5127
#11 0x00000000004e53c4 in __ast_pbx_run (c=0x12e6688, args=0x0) at pbx.c:5230
#12 0x00000000004e70eb in pbx_thread (data=0x7fb94c5ac170) at pbx.c:5571
#13 0x000000000051d51e in dummy_start (data=<value optimized out>) at utils.c:1010
#14 0x00007fb954fc273d in start_thread () from /lib64/libpthread.so.0
#15 0x00007fb955e874bd in clone () from /lib64/libc.so.6

[dkgroot]

#4  0x00000000004641a4 in __ast_channel_alloc_ap (needqueue=0, state=0, cid_num=0x7f4540054ec8 "11", cid_name=0x7f4540054e78 "Michaela Bilkova", acctcode=0x7f4540054ddc "", exten=0x145cdfd "", context=0x7f4540054d78 "internal", linkedid=0x7f455c10d974 "1361460453.169", amaflag=0, file=0x7f45598b04db "pbx_impl/ast/ast108.c", line=848, function=0x7f45598b0dc0 "sccp_wrapper_asterisk18_allocPBXChannel", name_fmt=0x7f45598b0de8 "SCCP/%s-%08X", ap1=0x7f45449010a0, ap2=0x7f4544901080) at channel.c:1150

#5  0x000000000046507e in __ast_channel_alloc (needqueue=2, state=1074073056, cid_num=0x0, cid_name=0x2 <Address 0x2 out of bounds>, acctcode=0x7f4540000488 "x\004", exten=0x145cdfd "", context=0x7f4540054d78 "internal", linkedid=0x7f455c10d974 "1361460453.169", amaflag=0, file=0x7f45598b04db "pbx_impl/ast/ast108.c", line=848, function=0x7f45598b0dc0 "sccp_wrapper_asterisk18_allocPBXChannel", name_fmt=0x7f45598b0de8 "SCCP/%s-%08X") at channel.c:1374

How does that work, cid_name going from out of bounds to being ok again ?

[dkgroot]

Closing issue. No extra information. Possible memory issue on reporters machine.

[anyremote]

seems i have the same issue (Asterisk 1.8.32.1/Chan_SCCP-4.2.0_RC3_r6042)

1. Asterisk gets an incoming call through the SIP trunk
2. By IVR menu call is transferred to SCCP extension
3. Asterisk crashed

[anyremote]

stacktrace

Program received signal SIGSEGV, Segmentation fault.
[Switching to Thread 0x7f861e757700 (LWP 20925)]
0x000000000051bab4 in ?? ()
(gdb) bt
#0 0x000000000051bab4 in ?? ()
#1 0x000000000051bd3f in ast_translator_free_path ()
#2 0x000000000045d346 in ?? ()
#3 0x000000000045d8be in ?? ()
#4 0x000000000045da09 in ast_channel_make_compatible ()
#5 0x00007f86231cd334 in ?? () from /usr/lib64/asterisk/modules/app_dial.so
#6 0x00007f86231cf431 in ?? () from /usr/lib64/asterisk/modules/app_dial.so
#7 0x00007f86231d30d6 in ?? () from /usr/lib64/asterisk/modules/app_dial.so
#8 0x00000000004e381b in pbx_exec ()
#9 0x00000000004e40a9 in ?? ()
#10 0x00000000004e4580 in ast_spawn_extension ()
#11 0x00007f862a5be8e7 in ?? () from /usr/lib64/asterisk/modules/app_macro.so
#12 0x00000000004e381b in pbx_exec ()
#13 0x00000000004e40a9 in ?? ()
#14 0x00000000004e4580 in ast_spawn_extension ()
#15 0x00000000004e5675 in ?? ()
#16 0x00000000004e6afb in ?? ()
#17 0x0000000000522638 in ?? ()
#18 0x00000032bb8077e1 in start_thread () from /lib64/libpthread.so.0
#19 0x00000032bb0e68ed in clone () from /lib64/libc.so.6

[dkgroot]

Hi Mikhail,

Can you describe the steps required to reproduce the issue in a little bit more detail please. Does the IVR really perform a transfer or do you mean the end result of the IVR is a DIAL to an sccp device ?

Would it be possible to test this with master branch (>= revision c1642d) ? If the issue does not occur in this newer version you should be able to update to V4.2 or even svn-trunk / git-master (Both are considered stable).

With this older V4.1 version you would have run ./configure with --enable-debug --disable-optimization to make the stacktrace contain enough information to be usefull. With newer versions (V4.2 and up) this configure setting is the default.

[anyremote]

i have the following in extensions.conf:

exten => s,1,Answer()
exten => s,n,BackGround(vm-enter-num-to-call) ;
exten => s,n,WaitExten

so on incoming call it asks to enter number and then dials it

BTW, i manages to get a workaround:

my peer was defined as (users.conf):
[mytrunk]
type=peer
host=bla.bla.com
username=87564564843
insecure=port,invite
fromdomain=bla.bla.com
fromuser=87564564843
disallow=all
allow=ulaw
allow=g729
allow=alaw <----------no crash if i comment out this line
allow=gsm
qualify=yes
transport=tcp,udp
nat=yes
dtmfmode=inband
context=trunk_inbound
canreinvite=no

if i comment line allow=alaw asterisk starts to work properly

[anyremote]

possible it connected with absence of the package (default language is ru)
rpm -qa|grep aster-sounds
asterisk-sounds-core-ru-ulaw-1.4.25-1.el6.noarch
asterisk-sounds-core-en-ulaw-1.4.25-1.el6.noarch
asterisk-sounds-core-en-alaw-1.4.25-1.el6.noarch
asterisk-curl-1.8.32.1-1.el6.x86_64
asterisk-sounds-core-ru-gsm-1.4.25-1.el6.noarch
asterisk-sounds-core-en-1.4.25-1.el6.noarch
asterisk-sounds-1.2.1-5.nodist.rf.noarch
asterisk-sounds-core-ru-1.4.25-1.el6.noarch

so there are no ru-alaw installed, only en-alaw

[dkgroot]

Hi Mikhail,

Pretty impressed you where able to find that one. Chan-sccp-b should not be concerned about transcoding in any way. So the issue is still a little mysterious.

What does allow/disallow set in sccp.conf ?

We might have an issue with earlyrtp in this case, when BackGround starts to playback in the 'wrong' codec. Somehow the asterisk app_dial is not able to find a correct transcoding path between ulaw/alaw which should be pretty straight forward to be honest.

BTW using asterisk -rx "file convert [filename].ulaw [filename].alaw" You can pre-convert your files, so alaw files are also available. With a little bit of scripting you should be able to convert all your sound files.

[dkgroot]

Without a backtrace there is not much i can do. This looks like a transcoding issue, which may wel point to an issue in asterisk / app_dial.

Why are you still using V4.2.0 RC3 instead of V4.2 Stable or (V4.3 trunk) ? The first thing to do when you run into an issue, is to check if there the software might have been updated. Also Asterisk-1.8 is getting a little bit old by now, might be time to update to asterisk-11 (your choice).