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

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=16701 
====================================================================== 
Reported By:                mtryfoss
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   16701
Category:                   PBX/General
Reproducibility:            always
Severity:                   block
Priority:                   normal
Status:                     feedback
Target Version:             Potential Blocker
Asterisk Version:           1.6.1.13 
JIRA:                       SWP-802 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2010-01-26 01:08 CST
Last Modified:              2010-01-26 12:08 CST
====================================================================== 
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.
======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
related to          0016513 [patch] Transferee  can hear silence on...
related to          0016630 [regression] Transfer is broken
related to          0016563 r236981 Attended transfer fails to comp...
====================================================================== 

---------------------------------------------------------------------- 
 (0117206) mtryfoss (reporter) - 2010-01-26 12:08
 https://issues.asterisk.org/view.php?id=16701#c117206 
---------------------------------------------------------------------- 
1) It's built-in transfer

2)
    -- Started music on hold, class 'default', on IAX2/pstn2-1473
    -- <Local/48999303 at outgoing-85f0;1> Playing 'pbx-transfer.gsm'
(language 'en')
    -- Executing [111 at transfertest:1]
Goto("Local/111 at transfertest-c6d5;2", "pstn1,41410986,1") in new stack
    -- Goto (pstn1,41410986,1)
    -- Executing [41410986 at pstn1:1]
UserEvent("Local/111 at transfertest-c6d5;2", "dialednum,Dialednum: 41410986")
in new stack
    -- Executing [41410986 at pstn1:2] Dial("Local/111 at transfertest-c6d5;2",
"IAX2/pstn1/41410986") in new stack
    -- Called pstn1/41410986
    -- Call accepted by 85.19.69.8 (format alaw)
    -- Format for call is alaw
    -- IAX2/pstn1-7354 is proceeding passing it to
Local/111 at transfertest-c6d5;2
    -- IAX2/pstn1-7354 is ringing
    -- IAX2/pstn1-7354 stopped sounds
    -- IAX2/pstn1-7354 answered Local/111 at transfertest-c6d5;2
[Jan 26 19:08:14] NOTICE[11822]: features.c:2161
ast_feature_request_and_dial: We exceeded our AT-timeout
    -- Stopped music on hold on IAX2/pstn2-1473 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-01-26 12:08 mtryfoss       Note Added: 0117206                          
======================================================================