[asterisk-bugs] [Asterisk 0015205]: Dropping frame since I'm still dialing on DAHDi/1-1...

Asterisk Bug Tracker noreply at bugs.digium.com
Thu Jul 9 18:51:51 CDT 2009


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=15205 
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Reported By:                vinsik
Assigned To:                
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Project:                    Asterisk
Issue ID:                   15205
Category:                   Channels/chan_dahdi
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.1.0 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-05-27 05:04 CDT
Last Modified:              2009-07-09 18:51 CDT
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Summary:                    Dropping frame since I'm still dialing on
DAHDi/1-1...
Description: 
chan_dahdi seems to think that call is still in dialing state.
All though it is not. Call is answered on the other side.

I have tried callprogress=yes, (both sip.conf and chan_dahdi.conf)

And weird thing is when i press a DTMF key, channel gets established and
audio is working fine.
====================================================================== 

---------------------------------------------------------------------- 
 (0107544) svnbot (reporter) - 2009-07-09 18:51
 https://issues.asterisk.org/view.php?id=15205#c107544 
---------------------------------------------------------------------- 
Repository: asterisk
Revision: 205730

U   branches/1.6.1/channels/chan_dahdi.c

------------------------------------------------------------------------
r205730 | rmudgett | 2009-07-09 18:51:50 -0500 (Thu, 09 Jul 2009) | 28
lines

Merged revisions 205728 via svn merge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r205728 | rmudgett | 2009-07-09 18:37:53 -0500 (Thu, 09 Jul 2009) | 21
lines
  
  No audio on calls from Asterisk to various ISDN devices until DTMF sent
by caller.
  
  Add missing clearing of the dialing flag when the ISDN call is
CONNECTED.
  (i.e. When libpri generates the event PRI_EVENT_ANSWER.)
  
  (closes issue https://issues.asterisk.org/view.php?id=15420)
  Reported by: scottbmilne
  Patches:
        bug15420-1.4.25.1-diff2.txt uploaded by alecdavis (license 585)
  Tested by: scottbmilne, alecdavis
  
  (closes issue https://issues.asterisk.org/view.php?id=15416)
  Reported by: avinoash
  
  (closes issue https://issues.asterisk.org/view.php?id=15389)
  Reported by: alecdavis
  
  This patch should also fix the following issue:
  (issue https://issues.asterisk.org/view.php?id=15205)
  Reported by: vinsik
........

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

http://svn.digium.com/view/asterisk?view=rev&revision=205730 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-07-09 18:51 svnbot         Checkin                                      
2009-07-09 18:51 svnbot         Note Added: 0107544                          
======================================================================




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