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Summary:ASTERISK-11224: Asterisk crash when make call from SIP endpoint to H323 using chan_h323
Reporter:Balgansuren Batsukh (balgaa)Labels:
Date Opened:2008-01-12 23:28:03.000-0600Date Closed:2011-06-07 14:02:42
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_h323
Versions:Frequency of
Occurrence
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Environment:Attachments:
Description:Recently I patched Asterisk 1.4.16/1.4.16.2/1.4.17 using www.b2bua.org codec negotiation patch. I tried on all versions.

After that when I make call from SIP endpoint to H323 network using chan_h323 driver Asterisk automatically crash.

I don't know what problem happen with Asterisk.

Below is backtrace:
----------------------
Asterisk Ready.
 == Parsing '/etc/asterisk/rpt.conf': Found
   -- Registered SIP '1100008' at 203.91.112.113 port 45900 expires 1800
   -- Saved useragent "X-PRO build 1082" for peer 1100008
[New Thread -1243259984 (LWP 6776)]
   -- Executing [00197699014447@default:1] Dial("SIP/1100008-081d00b0", "H323/00197699014447") in new stack

Program received signal SIGSEGV, Segmentation fault.
[Switching to Thread -1243259984 (LWP 6776)]
0xb7f10024 in pthread_mutex_lock () from /lib/tls/i686/cmov/libpthread.so.0
(gdb) bt
#0  0xb7f10024 in pthread_mutex_lock () from /lib/tls/i686/cmov/libpthread.so.0
#1  0x080d824d in ast_rtp_make_compatible (dest=0x823da18, src=0x81a2bf0, media=1)
   at /usr/src/asterisk-1.4.17/include/asterisk/lock.h:701
#2  0xb60e9498 in dial_exec_full (chan=0x81a2bf0, data=<value optimized out>, peerflags=0xb5e50f54, continue_exec=0x0)
   at app_dial.c:1202
#3  0xb60ee3e2 in dial_exec (chan=0x81a2bf0, data=0xb5e52fc8) at app_dial.c:1755
#4  0x080c984a in pbx_extension_helper (c=0x81a2bf0, con=0x0, context=0x81a2d70 "default", exten=0x81a2dc0 "00197699014447",
   priority=1, label=0x0, callerid=0x81d3780 "1100008", action=E_SPAWN) at pbx.c:532
ASTERISK-1  0x080cc32a in __ast_pbx_run (c=0x81a2bf0) at pbx.c:2306
ASTERISK-2  0x080cd3ee in pbx_thread (data=0x81a2bf0) at pbx.c:2623
ASTERISK-3  0x080faee0 in dummy_start (data=0x81d37a0) at utils.c:852
ASTERISK-4  0xb7f0e240 in start_thread () from /lib/tls/i686/cmov/libpthread.so.0
ASTERISK-5  0xb72e249e in clone () from /lib/tls/i686/cmov/libc.so.6
(gdb)
Comments:By: Balgansuren Batsukh (balgaa) 2008-01-13 00:21:40.000-0600

I found that when dial from analog phone connected to FXS of TDM400, Asterisk never crash.

But there also interesting thing happen, i can hear ringing tone on analog phone. If I call from my H323 phone to same number there voice response saying "Your called person mobile phone is off".

It show I can't hear real ringing tone from H323 equipment.

By: Balgansuren Batsukh (balgaa) 2008-01-13 00:22:46.000-0600

I found that when dial from analog phone connected to FXS of TDM400, Asterisk never crash.

But there also interesting thing happen, i can hear ringing tone on analog phone. If I call from my H323 phone to same number there voice response saying "Your called person mobile phone is off".

It show I can't hear real ringing tone from H323 equipment.

By: Balgansuren Batsukh (balgaa) 2008-01-13 19:57:46.000-0600

pbx:/home/balgaa# cat /proc/version
Linux version 2.6.18-5-686 (Debian 2.6.18.dfsg.1-17) (dannf@debian.org) (gcc version 4.1.2 20061115 (prerelease) (Debian 4.1.1-21)) #1 SMP Mon Dec 24 16:41:07 UTC 2007

By: Joshua C. Colp (jcolp) 2008-01-14 08:42:41.000-0600

Please try this with an unpatched version of Asterisk. If it is still an issue feel free to reopen. If not please notify the maintainer of the patch.

By: Balgansuren Batsukh (balgaa) 2008-01-14 10:20:55.000-0600

I reinstalled Asterisk-1.4.17 without codec negotiation patch and now from analog phone/sip phone Asterisk never crash.

But there also interesting thing happen, i can hear ringing tone on analog phone. If I call from my H323 phone to same number there voice response saying "Your called person mobile phone is off".

I saw when Asterisk get "Alerting" message from GnuGK it give sip/analog phone ringing tone. Same as before I can't hear real ringing tone from GnuGK.



By: Joshua C. Colp (jcolp) 2008-01-14 10:36:03.000-0600

Please open a new issue with complete logs and details.