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

[Asterisk Bug Tracker]

The following issue has been SUBMITTED. 
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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 07:18 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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Issue History 
Date Modified    Username       Field                    Change               
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2010-04-28 07:18 gblades_skymarketNew Issue                                    
2010-04-28 07:18 gblades_skymarketAsterisk Version          => 1.4.30          
2010-04-28 07:18 gblades_skymarketRegression                => No              
2010-04-28 07:18 gblades_skymarketSVN Branch (only for SVN checkouts, not
tarball releases) => N/A             
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