[asterisk-bugs] [Asterisk 0016701]: Attended transfer broken in 1.6.1.13
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Jan 26 01:40:05 CST 2010
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=16701
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Reported By: mtryfoss
Assigned To:
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Project: Asterisk
Issue ID: 16701
Category: PBX/General
Reproducibility: always
Severity: major
Priority: normal
Status: new
Asterisk Version: 1.6.1.13
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: 2010-01-26 01:08 CST
Last Modified: 2010-01-26 01:40 CST
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Summary: Attended transfer broken in 1.6.1.13
Description:
Error description:
1. A calls B
2. B picks up and talks to A
3. B does attended transfer to C
4. C picks up, but B still hears ringing
5. A and B are connected again (AT timeout exceeded on console)
https://issues.asterisk.org/view.php?id=16513 is a similar problem, but I
was told to file a different ticket since this is version 1.6.1.
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(0117169) wdoekes (reporter) - 2010-01-26 01:40
https://issues.asterisk.org/view.php?id=16701#c117169
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I'm running 1.6.1.13 and it works like a charm over here:
-- Called +316xxx at trunk
-- Called +316xxx at world_out
-- SIP/trunk-00000071 is ringing
-- Local/+316xxx at world_out-666c;1 is ringing
-- SIP/trunk-00000071 answered Local/+316xxx at world_out-666c;2
-- Local/+316xxx at world_out-666c;1 answered SIP/phone2-00000070
-- Packet2Packet bridging SIP/phone2-00000070 and SIP/trunk-00000071
== Spawn extension (world_out, +316xxx, 1) exited non-zero on
'Local/+316xxx at world_out-666c;2'
-- Stopped music on hold on SIP/phone1-0000006e
-- Packet2Packet bridging SIP/phone1-0000006e and SIP/trunk-00000071
Issue History
Date Modified Username Field Change
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2010-01-26 01:40 wdoekes Note Added: 0117169
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