[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:                     feedback
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:              2009-11-25 22:07 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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Relationships       ID      Summary
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related to          0016185 Crash of Outgoing Call
has duplicate       0016277 Outoing calls disconnected immediately ...
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 (0114305) ebroad (manager) - 2009-11-25 22:07
 https://issues.asterisk.org/view.php?id=16238#c114305 
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I have been noticing this as well on SVN trunk(recent, within 24 hours).
Asterisk sends an immediate bye after the callee picks up, nothing really
in the logs. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-11-25 22:07 ebroad         Note Added: 0114305                          
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