[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
Wed Jun 24 08:14:35 CDT 2009


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:                
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Project:                    Asterisk
Issue ID:                   15389
Category:                   Channels/chan_dahdi
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
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:              2009-06-24 08:14 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. 
====================================================================== 

---------------------------------------------------------------------- 
 (0106902) alecdavis (reporter) - 2009-06-24 08:14
 https://issues.asterisk.org/view.php?id=15389#c106902 
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uploaded:
1-6-1-nonworking-4888.txt: PRI debug capture of call to 4888 with no audio
present, with Asterisk SVN-branch-1.6.1-r202764.

working-4888.txt: PRI debug capture of call to 4888 with audio present,
this was with downgraded asterisk SVN-r178446

There's no diffence at the ISDN level.

What I forgot to add in the main description, was that if you pressed any
button on the phone that created DTMF while there should have been audio
present, that audio was then present for the remainder of the call. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-06-24 08:14 alecdavis      Note Added: 0106902                          
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