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

Asterisk Bug Tracker noreply at bugs.digium.com
Fri Feb 12 10:39:29 CST 2010


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:              2010-02-12 10:39 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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---------------------------------------------------------------------- 
 (0118035) frawd (reporter) - 2010-02-12 10:39
 https://issues.asterisk.org/view.php?id=16604#c118035 
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Still there in 1.6.2.3, I'm beginning to think that it is normal and that
the behavior of 1.4 was wrong. Analog lines don't send progress Ringing so
it is quite logical for * not to forward anything to the phone.

Please confirm this and close the bug if necessary.

As a workaround, if I dial through an Analog line, I place the "r" option
to Dial. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-02-12 10:39 frawd          Note Added: 0118035                          
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