[asterisk-bugs] [Asterisk 0015817]: crash in local_attended_transfer, likely related to moh - 1.4.26.1

Asterisk Bug Tracker noreply at bugs.digium.com
Mon Sep 14 15:13:03 CDT 2009


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=15817 
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Reported By:                zerohalo
Assigned To:                
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Project:                    Asterisk
Issue ID:                   15817
Category:                   Channels/chan_sip/Transfers
Reproducibility:            sometimes
Severity:                   crash
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           1.4.26.1 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-09-02 09:42 CDT
Last Modified:              2009-09-14 15:13 CDT
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Summary:                    crash in local_attended_transfer, likely related to
moh - 1.4.26.1
Description: 
Reopening this as new. Attended SIP transfer from Polycom UA, no queue
involved.

backtrace attached.
======================================================================
Relationships       ID      Summary
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related to          0015845 Crash during attended transfer occurs
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 (0110631) zerohalo (reporter) - 2009-09-14 15:13
 https://issues.asterisk.org/view.php?id=15817#c110631 
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Here's a snap of what the CLI looked like at time of crash:

    -- Executing [s at macro-vm:2] GotoIf("SIP/XXXX1119-0848-b68b4018",
"0?7:3") in new stack
    -- Goto (macro-vm,s,3)
    -- Executing [s at macro-vm:3] GosubIf("SIP/XXXX1119-0848-b68b4018",
"0?strip") in new stack
    -- Executing [s at macro-vm:4] Answer("SIP/XXXX1119-0848-b68b4018", "")
in new stack
  == Spawn extension (macro-out, s, 50) exited non-zero on
'SIP/XXXX0163-b6949180' in macro 'out'
  == Spawn extension (xxxxxx, XXXXXXXXXX, 51) exited non-zero on
'SIP/XXXX0163-b6949180'
    -- Executing [s at xxxxxxxxxxxx:11] BackGround("SIP/gateway1-b685d768",
"xxxx/xxxxxx/xxxxxx-greeting") in new stack
    -- <SIP/gateway1-b685d768> Playing 'xxxx/xxxxxx/xxxxxx-greeting'
(language 'en')
    -- Started music on hold, class 'classical', on SIP/gateway2-b70b6bc8
    -- Executing [s at macro-vm:6] VoiceMail("SIP/gateway1-b6c0b640",
"XXXX at xxxxx|su") in new stack
    -- <SIP/gateway1-b6c0b640> Playing
'/var/spool/asterisk/voicemail/xxxx/XXXX/unavail' (language 'en')
    -- Executing [s at macro-vm:6] VoiceMail("SIP/XXXX1119-0848-b68b4018",
"XXXX at xxxx|su") in new stack
    -- <SIP/XXXX1119-0848-b68b4018> Playing
'/var/spool/asterisk/voicemail/xxxx/XXXX/unavail' (language 'en')
asterisk*CLI> *** glibc detected *** double free or corruption (out):
0xb64fcd00 ***

Disconnected from Asterisk server 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-09-14 15:13 zerohalo       Note Added: 0110631                          
======================================================================




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