[asterisk-bugs] [Asterisk 0016604]: Asterisk does not send "183 Session Progress" when dialing through a dahdi analog line

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=16604 
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Reported By:                frawd
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16604
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           1.6.2.1-rc1 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-01-14 09:28 CST
Last Modified:              2010-02-02 16:16 CST
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Summary:                    Asterisk does not send "183 Session Progress" when
dialing through a dahdi analog line
Description: 
Scenario is a simple bridge between a SIP phone and an FXO line with
polarityswitch options turned on (and working). While the PSTN destination
hasn't answered yet, the SIP phone keeps silent and does not generate a
ring tone.

In 1.4 versions of Asterisk, after dialing the digits in the FXO, Asterisk
was sending a "183 Session Progress" back to the phone so it could generate
a progress ring.

In 1.6.2.1-rc1, it does not send it, probably related to some strange
failure to read frames:
[Jan 14 16:13:00] DEBUG[10593]: audiohook.c:248 audiohook_read_frame_both:
Failed to get 160 samples from read factory 0xa71d70

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 (0117570) frawd (reporter) - 2010-02-02 16:16
 https://issues.asterisk.org/view.php?id=16604#c117570 
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Please change to asterisk version 1.6.2.1 as rc1 doesn't exist anymore! 

Issue History 
Date Modified    Username       Field                    Change               
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2010-02-02 16:16 frawd          Note Added: 0117570                          
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