[asterisk-bugs] [Asterisk 0018192]: Call dropped on SIP REFER in 1.8.0

[Asterisk Bug Tracker]

The following issue has been SUBMITTED. 
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https://issues.asterisk.org/view.php?id=18192 
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Reported By:                job
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18192
Category:                   Core/Channels
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.8.0 
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:             2010-10-22 14:54 CDT
Last Modified:              2010-10-22 14:54 CDT
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Summary:                    Call dropped on SIP REFER in 1.8.0
Description: 
Asterisk drops a call silently when SIP REFER is sent. The call is dropped
when ast_queue_control is called in main/channel.c to read the remaining
frames.

There are no clues left in debug log or sip debug when this happens which
makes for sad users. This behavior was introduced in rc4, and call
transfers using SIP has been broken since then.

See below for a patch. The correct way to do this is probably to fix
ast_check_hangup so it does not return true when the channel is clearly not
hung up yet.
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Issue History 
Date Modified    Username       Field                    Change               
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2010-10-22 14:54 job            New Issue                                    
2010-10-22 14:54 job            Asterisk Version          => 1.8.0           
2010-10-22 14:54 job            Regression                => No              
2010-10-22 14:54 job            SVN Branch (only for SVN checkouts, not tarball
releases) => N/A             
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