[asterisk-bugs] [DAHDI-linux 0017224]: FXO Outgoing dial not passing audio until channel answer

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17224 
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Reported By:                sum
Assigned To:                
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Project:                    DAHDI-linux
Issue ID:                   17224
Category:                   dahdi (the module)
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
JIRA:                        
Reviewboard Link:            
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Date Submitted:             2010-04-21 14:53 CDT
Last Modified:              2010-04-23 10:05 CDT
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Summary:                    FXO Outgoing dial not passing audio until channel
answer
Description: 
I have a FXO conected to the PSTN, previously I just Dial a number and I
have the feedback of the telco (ringback), with the las version of dahdi
(not sure if dahdy or chan_dahdi issue), I have to do an answer befor dial
to let the ringback pass to the calling party.
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 (0120818) davidw (reporter) - 2010-04-23 10:05
 https://issues.asterisk.org/view.php?id=17224#c120818 
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Call progress is exactly about passing audio before answer!  Although
https://issues.asterisk.org/view.php?id=16604 is written (wrongly I think) from
a SIP perspective, SIP 183
Progress corresponds to an internal Asterisk control frame
AST_CONTROL_PROGRESS, which signals that the audio (ring back tones, and
other call progress tones) should passed through, even though the call
hasn't been answered.

The reason that issue doesn't mention dahdi driver versions is that they
are thinking in terms of the SIP symptoms, when, in my view, chan_sip is
simply reacting to the lack of an AST_CONTROL_PROGRESS from chan_dahdi. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-04-23 10:05 davidw         Note Added: 0120818                          
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