[asterisk-bugs] [Asterisk 0016701]: Attended transfer broken in 1.6.1.13

Asterisk Bug Tracker noreply at bugs.digium.com
Tue Jan 26 01:40:05 CST 2010


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=16701 
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Reported By:                mtryfoss
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16701
Category:                   PBX/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.1.13 
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:             2010-01-26 01:08 CST
Last Modified:              2010-01-26 01:40 CST
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Summary:                    Attended transfer broken in 1.6.1.13
Description: 
Error description: 

1. A calls B
2. B picks up and talks to A
3. B does attended transfer to C
4. C picks up, but B still hears ringing
5. A and B are connected again (AT timeout exceeded on console) 

https://issues.asterisk.org/view.php?id=16513 is a similar problem, but I
was told to file a different ticket since this is version 1.6.1.
====================================================================== 

---------------------------------------------------------------------- 
 (0117169) wdoekes (reporter) - 2010-01-26 01:40
 https://issues.asterisk.org/view.php?id=16701#c117169 
---------------------------------------------------------------------- 
I'm running 1.6.1.13 and it works like a charm over here:

    -- Called +316xxx at trunk
    -- Called +316xxx at world_out
    -- SIP/trunk-00000071 is ringing
    -- Local/+316xxx at world_out-666c;1 is ringing
    -- SIP/trunk-00000071 answered Local/+316xxx at world_out-666c;2
    -- Local/+316xxx at world_out-666c;1 answered SIP/phone2-00000070
    -- Packet2Packet bridging SIP/phone2-00000070 and SIP/trunk-00000071
  == Spawn extension (world_out, +316xxx, 1) exited non-zero on
'Local/+316xxx at world_out-666c;2'
    -- Stopped music on hold on SIP/phone1-0000006e
    -- Packet2Packet bridging SIP/phone1-0000006e and SIP/trunk-00000071 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-01-26 01:40 wdoekes        Note Added: 0117169                          
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