[asterisk-bugs] [Asterisk 0019337]: Call shows on hold after attended transfer with a Polycom phone
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon May 23 16:19:42 CDT 2011
A NOTE has been added to this issue.
======================================================================
https://issues.asterisk.org/view.php?id=19337
======================================================================
Reported By: remiq
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 19337
Category: Channels/chan_sip/Transfers
Reproducibility: always
Severity: minor
Priority: normal
Status: acknowledged
Asterisk Version: SVN
JIRA: SWP-3492
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): 1.8
SVN Revision (number only!): 319938
Request Review:
======================================================================
Date Submitted: 2011-05-20 10:33 CDT
Last Modified: 2011-05-23 16:19 CDT
======================================================================
Summary: Call shows on hold after attended transfer with a
Polycom phone
Description:
When I do an attended transfer from a Polycom IP650 the call is
transferring successfully, but the call is not releasing properly on the
phone that is initiating the transfer. Instead the call shows that it is
on hold.
======================================================================
----------------------------------------------------------------------
(0135313) mdavenport (administrator) - 2011-05-23 16:19
https://issues.asterisk.org/view.php?id=19337#c135313
----------------------------------------------------------------------
I still can't duplicate this. I can create an issue when defining my Cisco
7960 (7.2 SIP firmware) with nat=yes if I don't set pedantic=no in the
general section of sip.conf - because my Cisco phone doesn't properly
format its SIP requests. So, if I've got Cisco phones in the mix,
pedantic=no is a good idea.
I don't have the SIP devices you have to try and fully re-create your
setup.
That said, where Phone A = Cisco 7960 (SIP 7.2), Phone B = Polycom 670
(3.3.1.0933), Phone C = Polycom 450 (3.3.1.0933).
I can call from Phone A to Phone B.
Phone B answers.
Phone B initiates attended transfer to Phone C.
Phone C answers.
Phone B completes transfer.
Phones A & C converse.
Phone B hangs up.
Issue History
Date Modified Username Field Change
======================================================================
2011-05-23 16:19 mdavenport Note Added: 0135313
======================================================================
More information about the asterisk-bugs
mailing list