[asterisk-bugs] [Asterisk 0016238]: sip calls drop because of BYE's

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=16238 
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Reported By:                seandarcy
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16238
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.0.18-rc2 
JIRA:                        
Regression:                 Yes 
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:             2009-11-12 16:30 CST
Last Modified:              2009-11-12 16:49 CST
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Summary:                    sip calls drop because of BYE's
Description: 
On 1.6.0.16, 1.6.0.18-rc1 and 2, SIP calls die because * sends a BYE. We
have 12 internal SIP phones. The server connects them to sip, iax and dahdi
over libpri. iax and dahdi work fine. The internal SIP phones work fine.
But outgoing SIP calls are terminated with a BYE. We've tried sip over both
Teliax and Junction. Same result. Teliax support told us they could see the
BYE.
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 (0113753) seandarcy (reporter) - 2009-11-12 16:49
 https://issues.asterisk.org/view.php?id=16238#c113753 
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Attached sip.debug-clean. About line 514 you can see the BYE being sent. I
don't see anything that shows why the server decided to send a BYE.

Also attached is sip.conf and extensions.conf 

Issue History 
Date Modified    Username       Field                    Change               
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2009-11-12 16:49 seandarcy      Note Added: 0113753                          
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