[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:              2011-01-26 15:41 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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 (0131084) jakovas (reporter) - 2011-01-26 15:41
 https://issues.asterisk.org/view.php?id=18399#c131084 
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I have run into the same issue. Using Asterisk-1.8.2.2 and
libpri-1.4.11.5.
SIP-phone (Polycom,Grandstream) -> asterisk -> PRI -> some numbers
Getting "dead air" (if option "r" is not added to the DIAL command) and
disconnection after the dial timeout is expired when calling to the number
mentioned above and to several more numbers:
1-888-272-6565 (AT&T), 1-800-332-3226 (Safeco).

When calling to all of numbers which are not returning a ring tone a
"cause code 127" record appears in the log:
 
... app_dial.c:1288 in wait_for_answer: ...is proceeding passing it ...
... sig_pri.c:5080 in pri_dchannel ... PROGRESS with cause code 127
received

But most of such kind of numbers starts an IVR dialog after a short "dead
air" excluding mentioned numbers. 

Issue History 
Date Modified    Username       Field                    Change               
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2011-01-26 15:41 jakovas        Note Added: 0131084                          
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