[asterisk-bugs] [Asterisk 0018781]: [patch] segfault caused by remote_bridge_loop after a SIP to SIP attended transfer with an IAX2 call

Asterisk Bug Tracker noreply at bugs.digium.com
Thu Feb 17 04:13:07 CST 2011


A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=18781 
====================================================================== 
Reported By:                alecdavis
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   18781
Category:                   Core/RTP
Reproducibility:            always
Severity:                   crash
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           1.8.2.3 
JIRA:                       SWP-3081 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2011-02-09 19:18 CST
Last Modified:              2011-02-17 04:13 CST
====================================================================== 
Summary:                    [patch] segfault caused by remote_bridge_loop after
a SIP to SIP attended transfer with an IAX2 call
Description: 
Bug reported on 1.8.2 tag release, after experiencing similar on 1.8.2.3
tagged release.

core dump attached for 1.8.2


====================================================================== 

---------------------------------------------------------------------- 
 (0132071) alecdavis (manager) - 2011-02-17 04:13
 https://issues.asterisk.org/view.php?id=18781#c132071 
---------------------------------------------------------------------- 
Also fixes BAD MAGIC number, when call is from DAHDI, and call is
transferred.

Purposely disabled part of patch to demonstrate.

[Feb 17 23:08:18] WARNING[7746]: rtp_engine.c:1022 remote_bridge_loop:
Channels c0->'SIP/GXP8923-00000001' c1->'SIP/GXP8922-00000002' in
remote_bridge_loop
    -- Started music on hold, class 'default', on SIP/GXP8922-00000002
    -- Stopped music on hold on DAHDI/i1/4XXXX855-1
    -- Stopped music on hold on SIP/GXP8922-00000002
<b>[Feb 17 23:08:22] WARNING[7746]: rtp_engine.c:1030 remote_bridge_loop:
Oooh, something is weird, backing out
[Feb 17 23:08:22] WARNING[7746]: rtp_engine.c:1208 remote_bridge_loop:
Channel c0->'DAHDI/i1/45637855-1' changed, in bridge with
c1->'SIP/GXP8922-00000002'
[Feb 17 23:08:22] ERROR[7746]: astobj2.c:116 INTERNAL_OBJ: bad magic
number 0x1965 for 0xafba43d8
[Feb 17 23:08:22] ERROR[7746]: astobj2.c:116 INTERNAL_OBJ: bad magic
number 0x1965 for 0xafba43d8
[Feb 17 23:08:22] WARNING[7746]: rtp_engine.c:1216 remote_bridge_loop:
Channel c1->'SIP/GXP8922-00000002' changed, in bridge with
c0->'DAHDI/i1/4XXXX855-1'</b>
  == Spawn extension (trusted, 8923, 1) exited non-zero on
'SIP/GXP8923-00000001<ZOMBIE>' 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-02-17 04:13 alecdavis      Note Added: 0132071                          
======================================================================




More information about the asterisk-bugs mailing list