[asterisk-bugs] [Asterisk 0018781]: segfault with remote_bridge_loop seems to be when transferring call

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18781 
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Reported By:                alecdavis
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18781
Category:                   Core/RTP
Reproducibility:            random
Severity:                   crash
Priority:                   normal
Status:                     new
Asterisk Version:           1.8.2.3 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2011-02-09 19:18 CST
Last Modified:              2011-02-09 22:38 CST
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Summary:                    segfault with remote_bridge_loop seems to be when
transferring 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


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 (0131776) alecdavis (manager) - 2011-02-09 22:38
 https://issues.asterisk.org/view.php?id=18781#c131776 
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Confirmed segfault is only when a call comes in over IAX2 TRUNK

>>>>> Call coming from IAX2 to GXP0001 <<<<<
    -- Executing [s at macro-procexten:32] Dial("IAX2/auckbdt-46",
"SIP/GXP0001,10,r") in new stack
  == Using SIP RTP CoS mark 5
    -- Called GXP0001
    -- SIP/GXP0001-00000008 is ringing
    -- SIP/GXP0001-00000008 answered IAX2/auckbdt-46
>>>>> GXP0001 now talking on line1 to IAX2 caller <<<<<

    -- Started music on hold, class 'default', on IAX2/auckbdt-46
>>>>> GXP0001 has select another line, IAX2 caller has MOH <<<<<

>>>>> GXP0001 dials GXP0004 <<<<<
    -- Executing [s at macro-procexten:32] Dial("SIP/GXP0001-00000009",
"SIP/GXP0004,20,r") in new stack
  == Using SIP RTP CoS mark 5
    -- Called GXP0004
    -- SIP/GXP0004-0000000a is ringing
    -- SIP/GXP0004-0000000a answered SIP/GXP0001-00000009
    -- Remotely bridging SIP/GXP0001-00000009 and SIP/GXP0004-0000000a
>>>>> GXP0001 now talking on line2 with GXP0004 <<<<<

>>>>> GXP0001 now transfers line1 to line 2 <<<<<
    -- Started music on hold, class 'default', on SIP/GXP0004-0000000a
    -- Stopped music on hold on IAX2/auckbdt-46
    -- Stopped music on hold on SIP/GXP0004-0000000a
astrid*CLI>
Disconnected from Asterisk server
>>>>> SEGFAULTED <<<<< 

Issue History 
Date Modified    Username       Field                    Change               
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2011-02-09 22:38 alecdavis      Note Added: 0131776                          
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