[asterisk-bugs] [Asterisk 0010415]: [have fix] Transfers stopped working when migrating from 1.4.9 to 1.4.10
noreply at bugs.digium.com
noreply at bugs.digium.com
Fri Aug 10 08:43:15 CDT 2007
The following issue requires your FEEDBACK.
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http://bugs.digium.com/view.php?id=10415
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Reported By: atis
Assigned To: file
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Project: Asterisk
Issue ID: 10415
Category: Resources/res_features
Reproducibility: always
Severity: major
Priority: normal
Status: feedback
Asterisk Version: 1.4.10
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: 08-09-2007 03:47 CDT
Last Modified: 08-10-2007 08:43 CDT
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Summary: [have fix] Transfers stopped working when migrating
from 1.4.9 to 1.4.10
Description:
I use SIP and Zap channels, and when switching from version 1.4.9 to 1.4.10
transfers stopped working.
Solution was to unmerge changes in res_features.c
(http://bugs.digium.com/view.php?id=10327).
2007-07-30 17:11 +0000 [r77768-77778] Joshua Colp <jcolp at digium.com>
* res/res_features.c: (closes issue
http://bugs.digium.com/view.php?id=10327) Reported by: kkiely
Instead of directly mucking with the extension/context/priority
of the channel we are transferring when it has a PBX simply
call
ast_async_goto on it. This will ensure that the channel gets
handled properly and sent to the right place.
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file - 08-10-07 08:43
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What is the exact callflow that is causing it to fail? Dialplan involved?
Complete console output? Dial arguments?
Issue History
Date Modified Username Field Change
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08-10-07 08:43 file Note Added: 0068710
08-10-07 08:43 file Status assigned => feedback
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