I am seeing the following seg fault when using a SIP connection
to Console/Dsp. It takes quite a long time to happen but it eventually
happens.
nothing else is on this box. just alsa and asterisk running sip and
console/dsp.
What should I do now?
Jerry
Program received signal SIGSEGV, Segmentation fault.
[Switching to Thread -1215857760 (LWP 29590)]
__ast_read (chan=0x8ed5e08, dropaudio=0) at channel.c:2052
2052 f = AST_LIST_REMOVE_HEAD(&chan->readq,
frame_list);
(gdb) where
#0 __ast_read (chan=0x8ed5e08, dropaudio=0) at channel.c:2052
#1 0x08087b69 in ast_channel_bridge (c0=0x8eca668, c1=0x8ed5e08,
config=0xb78720d0, fo=0xb7871ca0, rc=0xb7871ca4) at channel.c:2348
#2 0x002f4bad in ast_bridge_call (chan=0x8eca668, peer=0x8ed5e08,
config=0xb78720d0) at res_features.c:1422
#3 0x00cbf03a in dial_exec_full (chan=0x8eca668, data=) at app_dial.c:1699
#4 0x00cc1bd4 in dial_exec (chan=0xffffffff, data=0xffffffff)
at app_dial.c:1753
#5 0x080ca1d0 in pbx_extension_helper (c=0x8eca668, con=)
at /usr/src/digium/asterisk-1.4.21.1/include/asterisk/strings.h:35
#6 0x080ceb46 in __ast_pbx_run (c=0x8eca668) at pbx.c:2317
#7 0x080d097e in pbx_thread (data=0x8eca668) at pbx.c:2636
#8 0x080ff5e5 in dummy_start (data=0xffffffff) at utils.c:895
#9 0x005963cc in start_thread () from /lib/tls/libpthread.so.0
#10 0x004ef1ae in clone () from /lib/tls/libc.so.6
(gdb) q
The program is running. Quit anyway (and detach it)? (y or n) Detaching
from program: /usr/sbin/asterisk, process 28404
Jerry Geis wrote:> I am seeing the following seg fault when using a SIP connection > to Console/Dsp. It takes quite a long time to happen but it eventually > happens. > nothing else is on this box. just alsa and asterisk running sip and > console/dsp. > > What should I do now? > > Jerry > > Program received signal SIGSEGV, Segmentation fault. > [Switching to Thread -1215857760 (LWP 29590)] > __ast_read (chan=0x8ed5e08, dropaudio=0) at channel.c:2052 > 2052 f = AST_LIST_REMOVE_HEAD(&chan->readq, > frame_list); > (gdb) where > #0 __ast_read (chan=0x8ed5e08, dropaudio=0) at channel.c:2052 > #1 0x08087b69 in ast_channel_bridge (c0=0x8eca668, c1=0x8ed5e08, > config=0xb78720d0, fo=0xb7871ca0, rc=0xb7871ca4) at channel.c:2348 > #2 0x002f4bad in ast_bridge_call (chan=0x8eca668, peer=0x8ed5e08, > config=0xb78720d0) at res_features.c:1422 > #3 0x00cbf03a in dial_exec_full (chan=0x8eca668, data=) at > app_dial.c:1699 > #4 0x00cc1bd4 in dial_exec (chan=0xffffffff, data=0xffffffff) > at app_dial.c:1753 > #5 0x080ca1d0 in pbx_extension_helper (c=0x8eca668, con=) > at /usr/src/digium/asterisk-1.4.21.1/include/asterisk/strings.h:35 > #6 0x080ceb46 in __ast_pbx_run (c=0x8eca668) at pbx.c:2317 > #7 0x080d097e in pbx_thread (data=0x8eca668) at pbx.c:2636 > #8 0x080ff5e5 in dummy_start (data=0xffffffff) at utils.c:895 > #9 0x005963cc in start_thread () from /lib/tls/libpthread.so.0 > #10 0x004ef1ae in clone () from /lib/tls/libc.so.6 > (gdb) q > The program is running. Quit anyway (and detach it)? (y or n) > Detaching from program: /usr/sbin/asterisk, process 28404 >silly me it is 1.4.21.1 not 1.2.21.1 Jerry