[asterisk-bugs] [Asterisk 0013686]: [patch] Console/dsp not hanging up after playing sound file.

Asterisk Bug Tracker noreply at bugs.digium.com
Sun Jan 25 18:03:54 CST 2009


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=13686 
====================================================================== 
Reported By:                itiliti
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   13686
Category:                   Channels/chan_oss
Reproducibility:            have not tried
Severity:                   minor
Priority:                   normal
Status:                     closed
Asterisk Version:           1.4.21.2 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 fixed
Fixed in Version:           
====================================================================== 
Date Submitted:             2008-10-13 16:51 CDT
Last Modified:              2009-01-25 18:03 CST
====================================================================== 
Summary:                    [patch] Console/dsp not hanging up after playing
sound file.
Description: 
We recently upgraded to Asterisk 1.4 on our phone server. Everything is
working great except that we notice on chan_oss, it will call the
Soundcard, then play the sound file, then it will not hangup. We are using
it with our paging system to play different sound files for different
departments. They are all part of a ring group, and it goes on to another
ring group if no one answers the phone.

Our code is this: 

exten => 900,1,Dial(console/dsp,,A(ring-raining))
exten => 900,Hangup

The issue is that it does not hang up the call afetr playing the sound
file. it actually puts the incoming call on the paging system. This worked
perfectly in Asterisk 1.2, but for some reason not in 1.4. We thought that
maybe there are new options to do this, but I couldnt find any.

Is this a bug, or am I missing something?

====================================================================== 

---------------------------------------------------------------------- 
 (0098715) svnbot (reporter) - 2009-01-25 18:03
 http://bugs.digium.com/view.php?id=13686#c98715 
---------------------------------------------------------------------- 
Repository: asterisk
Revision: 171190

_U  branches/1.6.1/
U   branches/1.6.1/channels/chan_oss.c

------------------------------------------------------------------------
r171190 | tilghman | 2009-01-25 18:03:53 -0600 (Sun, 25 Jan 2009) | 20
lines

Merged revisions 171188 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

................
  r171188 | tilghman | 2009-01-25 17:58:00 -0600 (Sun, 25 Jan 2009) | 13
lines
  
  Merged revisions 171187 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r171187 | tilghman | 2009-01-25 17:44:01 -0600 (Sun, 25 Jan 2009) | 6
lines
    
    Correctly track the hookstate
    (closes issue http://bugs.digium.com/view.php?id=13686)
     Reported by: itiliti
     Patches: 
           20081013__bug13686.diff.txt uploaded by Corydon76 (license 14)
  ........
................

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=171190 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-01-25 18:03 svnbot         Note Added: 0098715                          
======================================================================




More information about the asterisk-bugs mailing list