[asterisk-bugs] [Asterisk 0011755]: Asterisk crash when make call from SIP endpoint to H323 using chan_h323

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=11755 
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Reported By:                balgaa
Assigned To:                
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Project:                    Asterisk
Issue ID:                   11755
Category:                   Channels/chan_h323
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
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:              
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Date Submitted:             01-12-2008 23:28 CST
Last Modified:              01-13-2008 00:21 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)

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---------------------------------------------------------------------- 
 balgaa - 01-13-08 00:21  
---------------------------------------------------------------------- 
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. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
01-13-08 00:21  balgaa         Note Added: 0076834                          
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