[asterisk-bugs] [Asterisk 0011993]: asterisk crashes when trying to make a call from SIP endpoint to h323 endpoint registered to gnugk

noreply at bugs.digium.com noreply at bugs.digium.com
Thu Feb 14 20:46:43 CST 2008


A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=11993 
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Reported By:                tsgan
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   11993
Category:                   Channels/chan_h323
Reproducibility:            always
Severity:                   crash
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:              
====================================================================== 
Date Submitted:             02-14-2008 03:44 CST
Last Modified:              02-14-2008 20:46 CST
====================================================================== 
Summary:                    asterisk crashes when trying to make a call from SIP
endpoint to h323 endpoint registered to gnugk
Description: 
asterisk crashes and segmentation faults with core dump when trying to make
a call from SIP endpoint to h323 endpoint registered to gnugk.
====================================================================== 

---------------------------------------------------------------------- 
 tsgan - 02-14-08 20:46  
---------------------------------------------------------------------- 
calling from h323 to sip endpoint makes core dump again.


*CLI> [Feb 15 10:48:06] DEBUG[8324]: pbx.c:1831 pbx_extension_helper:
Launching 'Dial'
    -- Executing [1100051 at default:1]
Dial("H323/ip$192.168.0.18:20001/1636", "SIP/1100051|3600|t") in new stack
[Feb 15 10:48:06] DEBUG[8324]: chan_sip.c:4506 sip_alloc: Allocating new
SIP dialog for (No Call-ID) - INVITE (With RTP)
[Feb 15 10:48:06] DEBUG[8324]: chan_sip.c:2732 do_setnat: Setting NAT on
RTP to Off
[Feb 15 10:48:06] DEBUG[8324]: chan_sip.c:4025 sip_new: This channel will
not be able to handle video.
Segmentation fault (core dumped)
daemon1# gdb asterisk asterisk.core 


(gdb) bt
http://bugs.digium.com/view.php?id=0  0x29057fbe in pthread_mutex_lock () from
/lib/libthr.so.3
http://bugs.digium.com/view.php?id=1  0x080c6ee8 in ast_rtp_make_compatible ()
http://bugs.digium.com/view.php?id=2  0x29bfc23e in dial_exec_full
(chan=0x29e3a000, data=0xbed4ab38,
peerflags=0xbed48ae4, continue_exec=0x0) at app_dial.c:1255
http://bugs.digium.com/view.php?id=3  0x29c00999 in dial_exec (chan=0x29e3a000,
data=0xbed4ab38) at
app_dial.c:1808
http://bugs.digium.com/view.php?id=4  0x080bad86 in ast_add_extension2 ()
http://bugs.digium.com/view.php?id=5  0x080bc345 in ast_async_goto_by_name ()
http://bugs.digium.com/view.php?id=6  0x080bd012 in ast_pbx_run ()
http://bugs.digium.com/view.php?id=7  0x080e5c35 in ast_wait_for_input ()
http://bugs.digium.com/view.php?id=8  0x29053b1f in pthread_getprio () from
/lib/libthr.so.3
http://bugs.digium.com/view.php?id=9  0x00000000 in ?? ()
(gdb) 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
02-14-08 20:46  tsgan          Note Added: 0082296                          
======================================================================




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