[asterisk-bugs] [Asterisk 0019247]: When a call is blind-transfered to another extension, the CLI wrongly display the transfered call as been hung up.

Asterisk Bug Tracker noreply at bugs.digium.com
Fri May 6 06:16:51 CDT 2011


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=19247 
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Reported By:                thomas_foerster
Assigned To:                
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Project:                    Asterisk
Issue ID:                   19247
Category:                   Channels/chan_sip/Transfers
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           SVN 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.2 
SVN Revision (number only!): 307624 
Request Review:              
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Date Submitted:             2011-05-05 15:41 CDT
Last Modified:              2011-05-06 06:16 CDT
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Summary:                    When a call is blind-transfered to another
extension, the CLI wrongly display the transfered call as been hung up.
Description: 
Phone A is on a call with PSTN caller (SIP/sbc2-00000063). 
Phone A originates a blind transfer to extension B (992101)
CLI displays "== Spawn extension (customer, 992101, 1) exited non-zero on
'SIP/sbc2-00000063'"
Phone B is called and successfully bridged to PSTN caller. 

There isn't any problem in the way the call is actually handled.
Nevertheless the CLI displays that the transfered call is being hang up and
then it bridges it to the destination SIP peer. This is happening on a
server that handles a heavy load of blind transfers and Asterisk is
crashing daily on this specific server. It seems that the server crashes
when an "already-closed-on-the-CLI" channel is really hung up.


All this does not happen when using attended transfer instead of blind
transfer.
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---------------------------------------------------------------------- 
 (0134574) davidw (reporter) - 2011-05-06 06:16
 https://issues.asterisk.org/view.php?id=19247#c134574 
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As I understand it, mainline support for 1.6.2 ceased on April 21st.

The way that Asterisk implements SIP transfers, the original channel IS
hung up.  The only reservation I might have is that I think the channel
should have been re-named to a zombie, at that point.

Asterisk cannot redirect a bridged channel.  Instead it clones it, a
process called masquerading, then allows the original channel to hang up,
with some flags set to limit the amount of cleaning up that is done.

If you are getting crashes, you need to address the actual crashes, with a
backtrace, using a 1.8.x version (built unoptimised). 

Issue History 
Date Modified    Username       Field                    Change               
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2011-05-06 06:16 davidw         Note Added: 0134574                          
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