[asterisk-bugs] [Asterisk 0010915]: Problems when doing an attended and unattended transfer with Thomson Phones
noreply at bugs.digium.com
noreply at bugs.digium.com
Thu Oct 11 21:15:55 CDT 2007
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=10915
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Reported By: ramonpeek
Assigned To:
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Project: Asterisk
Issue ID: 10915
Category: Channels/General
Reproducibility: always
Severity: major
Priority: normal
Status: feedback
Asterisk Version: 1.4.12.1
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 10-08-2007 11:34 CDT
Last Modified: 10-11-2007 21:15 CDT
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Summary: Problems when doing an attended and unattended
transfer with Thomson Phones
Description:
When doing an attended transfer (party A -> B -> C) the phones A & B start
ringing after they finished the active call, but there is no audio
(ghostcalls)
When doing an unattended transfer the phone B (transferrer) start ringing
immediatly after transfering the party A to C.
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mavince - 10-11-07 21:15
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I loaded 1.4.13 last night. Using Polycom 600 phones, I can initiate an
attended transfer to a third phone (A->B->C) A goes on Hold and B<->C are
sending two way RTP. When I press the Transfer button again to connect
A<->C, the call disconnects. It appears that the attended transfer fails
right after a 491 between network and Asterisk server. Asterisk sends a BYE
to disconnect the call.
Blind transfer works. The difference appears to be a matter of timing
between SIP INVITES. Will report more tomorrow.
Conferencing work fine as well.
file - Should this problem continue under this issue or should another bug
report be created.
Issue History
Date Modified Username Field Change
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10-11-07 21:15 mavince Note Added: 0071858
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