[asterisk-bugs] [Asterisk 0015389]: [patch] no audio with SIP call to ISDN PRI, if neither Progress or Proceeding are received.

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=15389 
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Reported By:                alecdavis
Assigned To:                rmudgett
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Project:                    Asterisk
Issue ID:                   15389
Category:                   Channels/chan_dahdi
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     closed
Asterisk Version:           SVN 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.1 
SVN Revision (number only!): 202764 
Request Review:              
Resolution:                 fixed
Fixed in Version:           
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Date Submitted:             2009-06-24 08:01 CDT
Last Modified:              2009-07-09 18:56 CDT
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Summary:                    [patch] no audio with SIP call to ISDN PRI, if
neither Progress or Proceeding are received.
Description: 
After an upgrade from Asterisk SVN-r178446 to Asterisk 1.6.1-Branch.

Calling from SIP a phone to our ISDN PABX there is now no audio in either
direction.
Calling from the PABX into an Asterisk SIP phone audio is fine.

Audio on outbound call works fine on Asterisk 1.6.1.0-rc3 Tag,
but is broken on Asterisk 1.6.1.0-rc4 Tag.

The original patch (noted below) needs further work, to ensure audio path
is up when call is 'answered', may be earlier, when 'ringing' is received. 
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 (0107551) svnbot (reporter) - 2009-07-09 18:56
 https://issues.asterisk.org/view.php?id=15389#c107551 
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Repository: asterisk
Revision: 205731

U   branches/1.6.2/channels/chan_dahdi.c

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r205731 | rmudgett | 2009-07-09 18:56:21 -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
........

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http://svn.digium.com/view/asterisk?view=rev&revision=205731 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-07-09 18:56 svnbot         Note Added: 0107551                          
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