[asterisk-bugs] [Asterisk 0014164]: Dial() option d is not working

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=14164 
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Reported By:                DennisD
Assigned To:                putnopvut
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Project:                    Asterisk
Issue ID:                   14164
Category:                   Applications/app_dial
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     closed
Target Version:             1.6.0.5
Asterisk Version:           1.6.0.3-rc1 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.0 
SVN Revision (number only!): 167059 
Request Review:              
Resolution:                 fixed
Fixed in Version:           
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Date Submitted:             2009-01-02 15:08 CST
Last Modified:              2009-02-11 16:48 CST
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Summary:                    Dial() option d is not working
Description: 
"-- User hit 1 to disconnect call." doesn't show up and nothing happens
when using just the option d and pressing 1 when the call is ringing.

If I change the options to md, then it works (after a delay).  Also, if I
use retrydial, after it retries, then it works perfectly.

I have tried this with asterisk-1.6.0.3-rc1 and asterisk-1.6.0 from
SVN-branch-1.6.0-r167059M and they both do the same thing.


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---------------------------------------------------------------------- 
 (0099950) svnbot (reporter) - 2009-02-11 16:48
 http://bugs.digium.com/view.php?id=14164#c99950 
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Repository: asterisk
Revision: 174946

_U  branches/1.6.0/
U   branches/1.6.0/apps/app_dial.c
U   branches/1.6.0/apps/app_dictate.c
U   branches/1.6.0/apps/app_waitforsilence.c
U   branches/1.6.0/include/asterisk/channel.h
U   branches/1.6.0/main/channel.c
U   branches/1.6.0/main/pbx.c

------------------------------------------------------------------------
r174946 | mmichelson | 2009-02-11 16:48:11 -0600 (Wed, 11 Feb 2009) | 35
lines

Merged revisions 174945 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

........
  r174945 | mmichelson | 2009-02-11 16:41:01 -0600 (Wed, 11 Feb 2009) | 29
lines
  
  Fix 'd' option for app_dial and add new option to Answer application
  
  The 'd' option would not work for channel types which use RTP to
transport
  DTMF digits. The only way to allow for this to work was to answer the
channel
  if we saw that this option was enabled.
  
  I realized that this may cause issues with CDRs, specifically with
giving false
  dispositions and answer times. I therefore modified ast_answer to take
another
  parameter which would tell if the CDR should be marked answered.
  
  I also extended this to the Answer application so that the channel may
be answered
  but not CDRified if desired.
  
  I also modified app_dictate and app_waitforsilence to only answer the
channel if it
  is not already up, to help not allow for faulty CDR answer times.
  
  All of these changes are going into Asterisk trunk. For 1.6.0 and 1.6.1,
however, all
  the changes except for the change to the Answer application will go in
since we do
  not introduce new features into stable branches
  
  (closes issue http://bugs.digium.com/view.php?id=14164)
  Reported by: DennisD
  Patches:
        14164.patch uploaded by putnopvut (license 60)
  Tested by: putnopvut
  
  Review: http://reviewboard.digium.com/r/145
........

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http://svn.digium.com/view/asterisk?view=rev&revision=174946 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-02-11 16:48 svnbot         Checkin                                      
2009-02-11 16:48 svnbot         Note Added: 0099950                          
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