[asterisk-bugs] [Asterisk 0018379]: attended transfer weird behaviour

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18379 
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Reported By:                gincantalupo
Assigned To:                rmudgett
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Project:                    Asterisk
Issue ID:                   18379
Category:                   Applications/app_dial
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     closed
Asterisk Version:           1.8.1-rc1 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 fixed
Fixed in Version:           
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Date Submitted:             2010-11-25 11:35 CST
Last Modified:              2011-01-19 15:29 CST
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Summary:                    attended transfer weird behaviour
Description: 
Just installed 1.8.1-rc1 and tried the attended transfer function with 3
snoms (firmware 8.x), A,B and C. When A calls B and B transfers to C but C
is busy or does not answer, 'pbx-invalid.gsm' sound is played...but the
called number is right!

Another test: when I try to transfer the call to a wrong number I get this
message:
WARNING[31448]: features.c:1861 builtin_atxfer: Did not read data
and after that the call is bounced back to the transferrer (shouldn't
Asterisk say invalid extension???)

My test extensions:
exten => 12,1,Dial(SIP/81,5,tT)
exten => 12,2,NoOp(${DIALSTATUS})
exten => 12,3,Hangup

exten => 14,1,Dial(SIP/8,5,tT)
exten => 14,2,NoOp(${DIALSTATUS})
exten => 14,3,Hangup

exten => 17,1,Dial(SIP/70,5,tT)
exten => 17,2,NoOp(${DIALSTATUS})
exten => 17,3,Hangup

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Relationships       ID      Summary
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related to          0018254 Attended transfer failure
related to          0017999 Issues with DTMF triggered attended tra...
related to          0018618 atxfer doesn't work
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 (0130740) svnbot (reporter) - 2011-01-19 15:29
 https://issues.asterisk.org/view.php?id=18379#c130740 
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Repository: asterisk
Revision: 302713

_U  branches/1.8/
U   branches/1.8/main/features.c

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r302713 | rmudgett | 2011-01-19 15:29:23 -0600 (Wed, 19 Jan 2011) | 29
lines

Merged revisions 302693 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.6.2

................
  r302693 | rmudgett | 2011-01-19 15:25:41 -0600 (Wed, 19 Jan 2011) | 22
lines
  
  Merged revisions 302671 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r302671 | rmudgett | 2011-01-19 15:21:56 -0600 (Wed, 19 Jan 2011) | 15
lines
    
    DTMF transfer plays the wrong sounds for wrong number or other call
failure.
    
    * Set the default for features.conf.sample xferfailsound option to
"beeperr"
    as documented instead of "pbx-invalid" and corrected the use of it in
DTMF
    blind transfer (https://issues.asterisk.org/view.php?id=1).
    
    * Improved DTMF blind transfer handling of wrong numbers.
    
    Most of the concerns in this issue were taken care of by the patch for
    issue 17999: Issues with DTMF triggered attended transfers.
    
    (closes issue https://issues.asterisk.org/view.php?id=18379)
    Reported by: gincantalupo
    Tested by: rmudgett
  ........
................

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http://svn.digium.com/view/asterisk?view=rev&revision=302713 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-01-19 15:29 svnbot         Checkin                                      
2011-01-19 15:29 svnbot         Note Added: 0130740                          
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