[asterisk-bugs] [Asterisk 0018399]: Call torn down upon connection when early media 183 used

Asterisk Bug Tracker noreply at bugs.digium.com
Tue Dec 7 11:50:10 CST 2010


The following issue has been UPDATED. 
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https://issues.asterisk.org/view.php?id=18399 
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Reported By:                eeman
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18399
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           1.8.1-rc1 
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:             2010-11-29 15:27 CST
Last Modified:              2010-12-07 11:50 CST
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Summary:                    Call torn down upon connection when early media 183
used
Description: 
Asterisk 1.8.1-rc1 & Asterisk 1.6.2.14
Centos 5.5

have scenario as such

Asterisk-1.8.1-rc1 -SIP-> Asterisk 1.6.2.14 -SIP-> Broadvox (Sonus
Softswitch)

When calling a TF number that uses early media for their IVR (example
1-800-626-2001); once the call gets connected and the 200 OK message is
received, my 1.8.1-rc1 box issues a BYE message with a HangupCauseCode of
0. I can reproduce this with several numbers that are using early media for
their IVR's. Just as soon as my call gets connected to a call-center's ACD
Queue I hear 1-2 seconds of the recording before the call is torn down. I
have tested this using a linksys SPA-2102 ATA, A Polycom IP501, as well as
a Digium FXS module and get identical results. 
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Issue History 
Date Modified    Username       Field                    Change               
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2010-12-07 11:50 lmadsen        Description Updated                          
2010-12-07 11:50 lmadsen        Additional Information Updated                  
 
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