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

Asterisk Bug Tracker noreply at bugs.digium.com
Thu Jul 9 18:38:00 CDT 2009


The following issue has been ASSIGNED. 
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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:                     assigned
Asterisk Version:           SVN 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.1 
SVN Revision (number only!): 202764 
Request Review:              
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Date Submitted:             2009-06-24 08:01 CDT
Last Modified:              2009-07-09 18:38 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. 
====================================================================== 

---------------------------------------------------------------------- 
 (0107539) svnbot (reporter) - 2009-07-09 18:38
 https://issues.asterisk.org/view.php?id=15389#c107539 
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Repository: asterisk
Revision: 205728

U   branches/1.4/channels/chan_dahdi.c

------------------------------------------------------------------------
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=205728 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-07-09 18:38 svnbot         Checkin                                      
2009-07-09 18:38 svnbot         Note Added: 0107539                          
2009-07-09 18:38 svnbot         Status                   new => assigned     
2009-07-09 18:38 svnbot         Assigned To               => rmudgett        
======================================================================




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