[asterisk-bugs] [Asterisk 0017256]: Call gets disconnected after approx 20 seconds of ringing

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17256 
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Reported By:                gblades_skymarket
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17256
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.30 
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:             2010-04-28 07:18 CDT
Last Modified:              2010-04-28 08:29 CDT
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Summary:                    Call gets disconnected after approx 20 seconds of
ringing
Description: 
We have a system where calls come in over ISDN and then get routed out over
SIP.
The SIP carrier gives a SIP/183 response to indicate call progress but
after approx 20 seconds of ringing asterisk disconnects the call. I am not
specifying a duration to the dial command so it should not timeout.

We were running an older version of asterisk (1.4.22) and at that time we
manually modified sip.conf to make it consider a 183 to be the same as a
180 response which fixed this problem but it can cause other issues itself)
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 (0121066) pabelanger (manager) - 2010-04-28 08:29
 https://issues.asterisk.org/view.php?id=17256#c121066 
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I agree with davidw, this is not a bug, but a support issue.

ISDN Cause Code 19:
No answer from user (user alerted)

The destination responds to the connection request but fails to complete
the connection within the prescribed time. The problem is at the remote end
of the connection. 

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Thanks for your comments. This does not appear to be a bug report and we
are closing it. We appreciate the difficulties you are facing, but it would
make more sense to raise your question in the support tracker,
http://www.asterisk.org/support 

Issue History 
Date Modified    Username       Field                    Change               
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2010-04-28 08:29 pabelanger     Note Added: 0121066                          
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