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

[Asterisk Bug Tracker]

The issue 0018287 has been set as DUPLICATE OF the following issue. 
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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:                     closed
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:              
Resolution:                 duplicate
Duplicate:                  0
Fixed in Version:           
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Date Submitted:             2010-10-22 14:54 CDT
Last Modified:              2010-11-18 14:03 CST
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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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Relationships       ID      Summary
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duplicate of        0018185 Blind transfer failure, A calls B, B tr...
has duplicate       0018287 Unattended transfer failure
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Issue History 
Date Modified    Username       Field                    Change               
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2010-11-18 14:03 lmadsen        Relationship added       has duplicate 0018287
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