[asterisk-bugs] [Asterisk 0017956]: [patch] atxfer broken on 1.6.2.11
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Oct 12 12:48:03 CDT 2010
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=17956
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Reported By: ronald_verschaeren
Assigned To: mnicholson
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Project: Asterisk
Issue ID: 17956
Category: Features
Reproducibility: always
Severity: block
Priority: normal
Status: feedback
Target Version: 1.8.0
Asterisk Version: 1.6.2.11
JIRA: SWP-2167
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: 2010-09-06 03:14 CDT
Last Modified: 2010-10-12 12:48 CDT
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Summary: [patch] atxfer broken on 1.6.2.11
Description:
I'm using SIP phones.
- A calls B
- B transfers A to C by pressing ?2, then C's extension (atxfer =>
https://issues.asterisk.org/view.php?id=2 in
features.conf)
- C picks up, talks to B.
- B hangs up.
- both A and C hear silence
afaict, A is not bridged to C. This issue was not present in 1.6.1
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Relationships ID Summary
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related to 0017999 Issues with DTMF triggered attended tra...
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(0127892) nivek (reporter) - 2010-10-12 12:48
https://issues.asterisk.org/view.php?id=17956#c127892
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FWIW, We are getting the same type of results when transferring between
Polycom 650's using the transfer button (no DTMF transfers). Does not
happen on every transfer, but the results are always the same in that the
sound is inaudible or silent.
Most transfers are from an agent in a queue transferring the caller to
another extension.
Running Asterisk-1.6.2.11 under EL5 with latest Kernel with chan_sip and
chan_local.
Polycom phones are running sip 3.1.3.
Issue History
Date Modified Username Field Change
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2010-10-12 12:48 nivek Note Added: 0127892
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