[asterisk-bugs] [Asterisk 0018585]: [patch] AMI redirect from meetme - calls fail
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu Jan 13 03:42:26 CST 2011
A NOTE has been added to this issue.
======================================================================
https://issues.asterisk.org/view.php?id=18585
======================================================================
Reported By: oej
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 18585
Category: Channels/General
Reproducibility: always
Severity: minor
Priority: normal
Status: confirmed
Asterisk Version: SVN
JIRA: SWP-2893
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): 1.4
SVN Revision (number only!): 300917
Request Review:
======================================================================
Date Submitted: 2011-01-07 08:48 CST
Last Modified: 2011-01-13 03:42 CST
======================================================================
Summary: [patch] AMI redirect from meetme - calls fail
Description:
Possibly related to bug https://issues.asterisk.org/view.php?id=18230 and review
https://reviewboard.asterisk.org/r/1013/
Two calls in a meetme.
Issue a redirect to get one call out to the dialplan - play a prompt
there.
The call hangs up with no prompt played. The PBX jumps to the dialplan and
executes entries, but playback and wait fails
If you have a normal call and a normal PBX bridge, it works.
Testing with Asterisk 1.4 rev 300917
This has been working in earlier releases. I strongly suspect the changes
in the above commits to have changed something.
======================================================================
Relationships ID Summary
----------------------------------------------------------------------
related to 0018230 [regression] Redirect function (over co...
======================================================================
----------------------------------------------------------------------
(0130443) wedhorn (developer) - 2011-01-13 03:42
https://issues.asterisk.org/view.php?id=18585#c130443
----------------------------------------------------------------------
I can't think of one, except that at face value it seems reasonable. Why
keep the queue going if someone is hanging up? That's why I'd suggest a
review.
Personally I think it should go because there is no way that app_meetme
coders would be aware of all the ins and outs of who's changing what to
what in the softhangup bitmask. Nor should they need to be. The issue does
however indicate issues with ast_chan_hangup. Maybe it should be renamed to
ast_chan_maybe_maybenot_hangup.
Actually maybe a better name would be ast_chan_softhangup_bitmask_in_use.
Issue History
Date Modified Username Field Change
======================================================================
2011-01-13 03:42 wedhorn Note Added: 0130443
======================================================================
More information about the asterisk-bugs
mailing list