[asterisk-bugs] [Asterisk 0011755]: Asterisk crash when make call from SIP endpoint to H323 using chan_h323
noreply at bugs.digium.com
noreply at bugs.digium.com
Mon Jan 14 08:42:42 CST 2008
The following issue has been RESOLVED.
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http://bugs.digium.com/view.php?id=11755
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Reported By: balgaa
Assigned To: file
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Project: Asterisk
Issue ID: 11755
Category: Channels/chan_h323
Reproducibility: always
Severity: major
Priority: normal
Status: resolved
Asterisk Version: 1.4.17
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
Resolution: suspended
Fixed in Version:
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Date Submitted: 01-12-2008 23:28 CST
Last Modified: 01-14-2008 08:42 CST
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Summary: Asterisk crash when make call from SIP endpoint to
H323 using chan_h323
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 at 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
http://bugs.digium.com/view.php?id=0 0xb7f10024 in pthread_mutex_lock () from
/lib/tls/i686/cmov/libpthread.so.0
http://bugs.digium.com/view.php?id=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
http://bugs.digium.com/view.php?id=2 0xb60e9498 in dial_exec_full
(chan=0x81a2bf0, data=<value optimized
out>, peerflags=0xb5e50f54, continue_exec=0x0)
at app_dial.c:1202
http://bugs.digium.com/view.php?id=3 0xb60ee3e2 in dial_exec (chan=0x81a2bf0,
data=0xb5e52fc8) at
app_dial.c:1755
http://bugs.digium.com/view.php?id=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
http://bugs.digium.com/view.php?id=5 0x080cc32a in __ast_pbx_run (c=0x81a2bf0)
at pbx.c:2306
http://bugs.digium.com/view.php?id=6 0x080cd3ee in pbx_thread (data=0x81a2bf0)
at pbx.c:2623
http://bugs.digium.com/view.php?id=7 0x080faee0 in dummy_start (data=0x81d37a0)
at utils.c:852
http://bugs.digium.com/view.php?id=8 0xb7f0e240 in start_thread () from
/lib/tls/i686/cmov/libpthread.so.0
http://bugs.digium.com/view.php?id=9 0xb72e249e in clone () from
/lib/tls/i686/cmov/libc.so.6
(gdb)
======================================================================
----------------------------------------------------------------------
file - 01-14-08 08:42
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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.
Issue History
Date Modified Username Field Change
======================================================================
01-14-08 08:42 file Status new => resolved
01-14-08 08:42 file Resolution open => suspended
01-14-08 08:42 file Assigned To => file
01-14-08 08:42 file Note Added: 0076888
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