[asterisk-bugs] [Asterisk 0019372]: Attended transfer - transfering phone left connected
Asterisk Bug Tracker
noreply at bugs.digium.com
Fri May 27 10:45:41 CDT 2011
The following issue requires your FEEDBACK.
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https://issues.asterisk.org/view.php?id=19372
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Reported By: jamicque
Assigned To:
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Project: Asterisk
Issue ID: 19372
Category: Channels/chan_sip/Transfers
Reproducibility: always
Severity: minor
Priority: normal
Status: feedback
Asterisk Version: 1.8.4.1
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: 2011-05-26 09:52 CDT
Last Modified: 2011-05-27 10:45 CDT
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Summary: Attended transfer - transfering phone left connected
Description:
The issue 0015833 still exists in newest version of Asterisk.
When doing a remote attended transfer in one of these 2 setups:
phones A,B,C --- proxy --- asterisks Z,X
when A->B call is on Z and B->C is on X, or:
phones A,B (with identity B1,B2), C --- asterisks Z,X
(A,B1 register on Z; B2,C on X)
when A->B1 call is on Z and B2->C is on X
In both scenarios Z and X are friends with no authentication needed.
The B phone doesn't get properly disconnected. asterisks invite/replace
each other properly and the audio channel is ok. B itself drops one of the
calls. But Z is not disconnecting B's call at all. You can replicate that
scenario with minimalistic dialplan - _X.,Dial(SIP/${EXTEN}) in default on
both sides.
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Relationships ID Summary
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related to 0007784 Attended SIP Transfer Call Teardown Issue
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(0135487) lmadsen (administrator) - 2011-05-27 10:45
https://issues.asterisk.org/view.php?id=19372#c135487
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Is this reproducible without the proxy in front?
Issue History
Date Modified Username Field Change
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2011-05-27 10:45 lmadsen Note Added: 0135487
2011-05-27 10:45 lmadsen Status new => feedback
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