[asterisk-bugs] [Asterisk 0018399]: Call torn down upon connection when early media 183 used
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Mar 2 18:10:59 CST 2011
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: 2011-03-02 18:10 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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Relationships ID Summary
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related to 0018632 missing Contact header in 200 OK to INVITE
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(0132553) jacobli (reporter) - 2011-03-02 18:10
https://issues.asterisk.org/view.php?id=18399#c132553
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The problem still exist in vers. 1.8.3
Issue History
Date Modified Username Field Change
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2011-03-02 18:10 jacobli Note Added: 0132553
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