[asterisk-bugs] [Asterisk 0013233]: Asterisk reports dialstatus as "CONGESTION" when received a 480 "Temporarily unavailable"

Asterisk Bug Tracker noreply at bugs.digium.com
Mon Aug 4 09:32:34 CDT 2008


A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=13233 
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Reported By:                rickead2000
Assigned To:                
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Project:                    Asterisk
Issue ID:                   13233
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.21.1 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             2008-08-04 08:03 CDT
Last Modified:              2008-08-04 09:32 CDT
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Summary:                    Asterisk reports dialstatus as "CONGESTION" when
received a 480 "Temporarily unavailable"
Description: 
Asterisk currently sets the dialstatus to be "CONGESTION" when a 403
"Temporarily unavailable" is received.  It then sends on a 503 "Service
unavailable" to the other party.  For compliance with the RFC I believe
asterisk should be sending back the 480 Temp. unavail to the 3rd party as
it received it and probably should be setting the dialstatus to "NOANSWER"
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---------------------------------------------------------------------- 
 (0091043) davidw (reporter) - 2008-08-04 09:32
 http://bugs.digium.com/view.php?id=13233#c91043 
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I believe the reason that it returns Congestion is that Dial can have
multiple destinations, with different failure reasons and congestions sums
up the idea that there is no usable onward route.

However, http://bugs.digium.com/view.php?id=12885 also seems relevant, and that
seems to take the position
that one can more accurately reflect the called line state in the, common,
degenerate case of only one destination for the Dial application.

PS Which RFC?  The main SIP RFC doesn't apply to back to back end systems,
but the RFC mentioned in http://bugs.digium.com/view.php?id=12885 probably is
relevant. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-08-04 09:32 davidw         Note Added: 0091043                          
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