[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:                
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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:              2009-06-25 07:25 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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---------------------------------------------------------------------- 
 (0106968) alecdavis (reporter) - 2009-06-25 07:25
 https://issues.asterisk.org/view.php?id=15389#c106968 
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please remove bug15389_noaudio.diff.txt
after cleanup I removed dialing=0, and didn't test before submitting.
Note to me: Always test after cleanup, before submitting patch!!!!

uploaded bug15389_noaudio.diff2.txt:
if ISDN Proceeding and Progress Indicators are not seen, and the call is
answered the 'dialing' flag is left set, preventing an audio path setup.

This essentially 1 line patch, sets 'dialing' flag to 0, when
PRI_EVENT_ANSWER is fired.

=======================================
Console output below:
 - 1st call to PABX without PROGRESS or PROCEEDING indicators
 - 2nd call to PSTN with both PROCEEDING and PROGRESS.

Both calls were sucessful, both with Audio path.
Previously only the 2nd call below would have had an audio path.
=======================================

  == Using SIP RTP CoS mark 5
    -- Executing [4888 at trusted:1] Dial("SIP/8630-08cbefe8",
"DAHDI/G0/4888") in new stack
    -- Requested transfer capability: 0x00 - SPEECH
    -- Called G0/4888
!! Unknown IE 50 (cs5, Unknown Information Element)
    -- DAHDI/31-1 is ringing
    -- DAHDI/31-1 answered SIP/8630-08cbefe8
    -- Hungup 'DAHDI/31-1'
  == Spawn extension (trusted, 4888, 1) exited non-zero on
'SIP/8630-08cbefe8'
  == Using SIP RTP CoS mark 5


    -- Executing [1083210 at trusted:1] Dial("SIP/8630-08cc5a58",
"DAHDI/G0/083210,,r") in new stack
    -- Requested transfer capability: 0x00 - SPEECH
    -- Called G0/083210
    -- DAHDI/31-1 is proceeding passing it to SIP/8630-08cc5a58
    -- DAHDI/31-1 is making progress passing it to SIP/8630-08cc5a58
    -- DAHDI/31-1 is making progress passing it to SIP/8630-08cc5a58
!! Unknown IE 50 (cs5, Unknown Information Element)
    -- DAHDI/31-1 answered SIP/8630-08cc5a58
    -- Hungup 'DAHDI/31-1' 

Issue History 
Date Modified    Username       Field                    Change               
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2009-06-25 07:25 alecdavis      Note Added: 0106968                          
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