[asterisk-bugs] [Asterisk 0017233]: Call transfer from Voicemail or Queue Application result Asterisk to crash. (SIP REFER)

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17233 
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Reported By:                anazaruk
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17233
Category:                   Channels/chan_sip/Transfers
Reproducibility:            always
Severity:                   crash
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.0.26 
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:             2010-04-22 22:23 CDT
Last Modified:              2010-04-23 02:57 CDT
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Summary:                    Call transfer from Voicemail or Queue Application
result Asterisk to crash. (SIP REFER)
Description: 
Polycom and Linksys SIP phones result same crash. 

1. Dial voicemail or queue.
2. While checking voicemail or being in queue, put call on hold and dial
another extension from the same IP phone.
3. Pick up the phone on another extension you dial.
4. Push transfer button. 

IP phone will send SIP REFER message to asterisk witch will result
asterisk to CORE.
..

It only crashes when you trying to transfer Voicemail or Queue to
Extension.
While trying to transfer extension to Queue or Voicemail it will not
crash.

Other call transfers work fine. (ext to ext or ext to PSTN number out
SIP)

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 (0120801) anazaruk (reporter) - 2010-04-23 02:57
 https://issues.asterisk.org/view.php?id=17233#c120801 
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please close this bug. I have figured it out.

Thank you 

Issue History 
Date Modified    Username       Field                    Change               
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2010-04-23 02:57 anazaruk       Note Added: 0120801                          
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