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

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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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:                       SWP-2666 
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:              2011-01-14 09:02 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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 (0130496) schmidts (manager) - 2011-01-14 09:02
 https://issues.asterisk.org/view.php?id=18399#c130496 
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maybe i am complete wrong but the warning:
[Nov 29 16:33:38] WARNING[25528]: chan_sip.c:12909
__set_address_from_contact: Invalid contact uri (missing sip: or sips:),
attempting to use anyway
says there is no contact header in the 200 ok message, and there is really
no contact in there only in the 183 session progress.

sorry if i missunderstood the problem but it looks like this missing
contact is the issue or the wrong handling of asterisk if its ok to have no
contact in a 200 ok.

best regards

stefan 

Issue History 
Date Modified    Username       Field                    Change               
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2011-01-14 09:02 schmidts       Note Added: 0130496                          
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