[asterisk-users] asterisk-1.6.0.9-x86_64: voicemail: Segmentation fault (core dumped)
Martin
asterisklist at callthem.info
Sat Apr 18 20:09:42 CDT 2009
Hi,
Your backtrace doesn't make sense to me.
Do you have in main/stdtime/localtime.c
this function that way ?
struct ast_tm *ast_localtime(const struct timeval *timep, struct
ast_tm *tmp, const char *zone)
{
const struct state *sp = ast_tzset(zone);
memset(tmp, 0, sizeof(*tmp));
return sp ? localsub(timep, 0L, tmp, sp) : NULL;
}
If so your backtrace states:
> #3 0x00000000004da92d in ast_tzset (zone=0x7f7f741e5bf9 "UTC")
> at stdtime/localtime.c:1029
> #4 0x00000000004db98c in ast_localtime (timep=0x7f7f8407c500,
> tmp=0x7f7f84076ba0, zone=0x0) at stdtime/localtime.c:1142
ast_localtime is called with zone=NULL
and yet ast_tzset is called with zone = "UTC"
you must have downloaded some version with hardcoded "UTC" timezone ...
or there's a major memory problem ...
Martin
On Sat, Apr 18, 2009 at 7:04 PM, Justin Piszcz <jpiszcz at lucidpixels.com> wrote:
> Hello,
>
> Information:
> gcc -v: gcc version 4.3.3 (Debian 4.3.3-3)
> os: Debian/Testing
>
> Pulled latest release from asterisk site, compiled, installed it.
>
> I have a barebones configuration:
> $ ls -l asterisk
> extensions.conf
> modules.conf
> sip.conf
> users.conf
> voicemail.conf
>
> You can see them here:
> http://home.comcast.net/~jpiszcz/20090418/extensions.conf
> http://home.comcast.net/~jpiszcz/20090418/modules.conf
> http://home.comcast.net/~jpiszcz/20090418/sip.conf
> http://home.comcast.net/~jpiszcz/20090418/users.conf
> http://home.comcast.net/~jpiszcz/20090418/voicemail.conf
>
> When I perform the following actions, asterisk segfaults:
>
> 1. Dial *61.
> 2. Enter password: XXXX
> 3. Enter 3 for advanced options.
> 4. Press 5 to leave a message, press * to return to the main menu.
> 5. Extension: 6000
> 6. Please leave your message after the tone, when done, please hangup or
> press the pound key (it segfaults right after it says pound key)
> 7. Segmentation fault
>
> == Using SIP RTP CoS mark 5
> -- Executing [*61 at line1:1] VoiceMailMain("SIP/line1-01d646b0", "6001") in new stack
> -- <SIP/line1-01d646b0> Playing 'vm-password.gsm' (language 'en')
> DTMF begin '1' received on SIP/line1-01d646b0
> DTMF begin ignored '1' on SIP/line1-01d646b0
> DTMF end '1' received on SIP/line1-01d646b0, duration 190 ms
> DTMF end passthrough '1' on SIP/line1-01d646b0
> /DTMF begin '2' received on SIP/line1-01d646b0
> DTMF begin ignored '2' on SIP/line1-01d646b0
> DTMF end '2' received on SIP/line1-01d646b0, duration 130 ms
> DTMF end passthrough '2' on SIP/line1-01d646b0
> DTMF begin '3' received on SIP/line1-01d646b0
> DTMF begin ignored '3' on SIP/line1-01d646b0
> DTMF end '3' received on SIP/line1-01d646b0, duration 130 ms
> DTMF end passthrough '3' on SIP/line1-01d646b0
> DTMF begin '4' received on SIP/line1-01d646b0
> DTMF begin ignored '4' on SIP/line1-01d646b0
> DTMF end '4' received on SIP/line1-01d646b0, duration 170 ms
> DTMF end passthrough '4' on SIP/line1-01d646b0
> DTMF begin '#' received on SIP/line1-01d646b0
> DTMF begin ignored '#' on SIP/line1-01d646b0
> DTMF end '#' received on SIP/line1-01d646b0, duration 170 ms
> DTMF end passthrough '#' on SIP/line1-01d646b0
> -- <SIP/line1-01d646b0> Playing 'vm-youhave.gsm' (language 'en')
> -- <SIP/line1-01d646b0> Playing 'vm-no.gsm' (language 'en')
> -- <SIP/line1-01d646b0> Playing 'vm-messages.gsm' (language 'en')
> -- <SIP/line1-01d646b0> Playing 'vm-opts.gsm' (language 'en')
> -- <SIP/line1-01d646b0> Playing 'vm-helpexit.gsm' (language 'en')
> DTMF begin '3' received on SIP/line1-01d646b0
> DTMF begin ignored '3' on SIP/line1-01d646b0
> DTMF end '3' received on SIP/line1-01d646b0, duration 130 ms
> DTMF end passthrough '3' on SIP/line1-01d646b0
> -- <SIP/line1-01d646b0> Playing 'vm-leavemsg.gsm' (language 'en')
> -- <SIP/line1-01d646b0> Playing 'vm-starmain.gsm' (language 'en')
> -- <SIP/line1-01d646b0> Playing 'vm-leavemsg.gsm' (language 'en')
> DTMF begin '5' received on SIP/line1-01d646b0
> DTMF begin ignored '5' on SIP/line1-01d646b0
> -- <SIP/line1-01d646b0> Playing 'vm-starmain.gsm' (language 'en')
> DTMF end '5' received on SIP/line1-01d646b0, duration 170 ms
> DTMF end passthrough '5' on SIP/line1-01d646b0
> -- <SIP/line1-01d646b0> Playing 'vm-extension.gsm' (language 'en')
> DTMF begin '6' received on SIP/line1-01d646b0
> DTMF begin ignored '6' on SIP/line1-01d646b0
> DTMF end '6' received on SIP/line1-01d646b0, duration 170 ms
> DTMF end passthrough '6' on SIP/line1-01d646b0
> DTMF begin '0' received on SIP/line1-01d646b0
> DTMF begin ignored '0' on SIP/line1-01d646b0
> DTMF end '0' received on SIP/line1-01d646b0, duration 130 ms
> DTMF end passthrough '0' on SIP/line1-01d646b0
> DTMF begin '0' received on SIP/line1-01d646b0
> DTMF begin ignored '0' on SIP/line1-01d646b0
> DTMF end '0' received on SIP/line1-01d646b0, duration 170 ms
> DTMF end passthrough '0' on SIP/line1-01d646b0
> DTMF begin '0' received on SIP/line1-01d646b0
> DTMF begin ignored '0' on SIP/line1-01d646b0
> DTMF end '0' received on SIP/line1-01d646b0, duration 170 ms
> DTMF end passthrough '0' on SIP/line1-01d646b0
> -- <SIP/line1-01d646b0> Playing 'digits/6.gsm' (language 'en')
> -- <SIP/line1-01d646b0> Playing 'digits/0.gsm' (language 'en')
> -- <SIP/line1-01d646b0> Playing 'digits/0.gsm' (language 'en')
> -- <SIP/line1-01d646b0> Playing 'digits/0.gsm' (language 'en')
> -- <SIP/line1-01d646b0> Playing 'vm-intro.gsm' (language 'en')
> -- <SIP/line1-01d646b0> Playing 'beep.gsm' (language 'en')
> Segmentation fault (core dumped)
>
> Here is a backtrace:
>
> Core was generated by `/vapp/sbin/asterisk -vvvvvvvvvvv'.
> Program terminated with signal 11, Segmentation fault.
> [New process 23729]
> [New process 23717]
> [New process 23721]
> [New process 23728]
> [New process 23718]
> [New process 23723]
> [New process 23719]
> [New process 23722]
> [New process 23720]
> [New process 23726]
> [New process 23727]
> #0 0x00000000004d8bb3 in tzload (name=0x536b26 "posixrules",
> sp=0x7f7f84076ba0, doextend=0) at stdtime/localtime.c:292
> 292 if ((strlen(p) + strlen(name) + 1) >= sizeof fullname)
> (gdb) bt
> #0 0x00000000004d8bb3 in tzload (name=0x536b26 "posixrules",
> sp=0x7f7f84076ba0, doextend=0) at stdtime/localtime.c:292
> #1 0x00000000004d94bf in tzparse (name=0x7f7f8406c7a5 "", sp=0x7f7f84076ba0,
> lastditch=<value optimized out>) at stdtime/localtime.c:811
> #2 0x00000000004d9152 in tzload (name=<value optimized out>, sp=0x881280,
> doextend=1) at stdtime/localtime.c:450
> #3 0x00000000004da92d in ast_tzset (zone=0x7f7f741e5bf9 "UTC")
> at stdtime/localtime.c:1029
> #4 0x00000000004db98c in ast_localtime (timep=0x7f7f8407c500,
> tmp=0x7f7f84076ba0, zone=0x0) at stdtime/localtime.c:1142
> #5 0x00007f7f741d332f in get_date (s=0x7f7f84086490 "0\005\210", len=256)
> at app_voicemail.c:3788
> #6 0x00007f7f741dfb1a in leave_voicemail (chan=0x87f5f0,
> ext=<value optimized out>, options=0x7f7f84090c00) at app_voicemail.c:4476
> #7 0x00007f7f741e1ab0 in forward_message (chan=0x87f5f0, context=0x0,
> vms=0x7f7f84090d40, sender=0x7f7f84096e10,
> fmt=0x7f7f743ee4e0 "wav49|gsm|wav", flag=1, record_gain=0 '\0')
> at app_voicemail.c:5608
> #8 0x00007f7f741e2ece in vm_execmain (chan=0x87f5f0,
> data=<value optimized out>) at app_voicemail.c:7999
> #9 0x00000000004aedf5 in pbx_extension_helper (c=0x87f5f0,
> con=<value optimized out>, context=0x87f848 "line1", exten=0x87f898 "*61",
> priority=1, label=0x0, callerid=0x8794b0 "anonymous", action=E_SPAWN,
> found=0x7f7f8409c03c, combined_find_spawn=1) at pbx.c:942
> #10 0x00000000004b039a in __ast_pbx_run (c=0x87f5f0, args=0x0) at pbx.c:3614
> #11 0x00000000004b160b in pbx_thread (data=0x536b26) at pbx.c:3974
> #12 0x00000000004e6c0c in dummy_start (data=<value optimized out>)
> at utils.c:861
> #13 0x00007f7f8b531faa in start_thread () from /lib/libpthread.so.0
> #14 0x00007f7f87bc62bd in clone () from /lib/libc.so.6
> #15 0x0000000000000000 in ?? ()
> (gdb)
>
> Any clues?
>
> Justin.
>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
More information about the asterisk-users
mailing list