[asterisk-bugs] [Asterisk 0018585]: AMI redirect from meetme - calls fail

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18585 
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Reported By:                oej
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18585
Category:                   Channels/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           SVN 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases):  1.4  
SVN Revision (number only!): 300917 
Request Review:              
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Date Submitted:             2011-01-07 08:48 CST
Last Modified:              2011-01-10 20:18 CST
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Summary:                    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.
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 (0130368) wedhorn (developer) - 2011-01-10 20:18
 https://issues.asterisk.org/view.php?id=18585#c130368 
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Had a play and found the issue is introduced in r291577.

Notably, tested with skinny devices, but r291392 plays demo-congrats when
AMI redirect from Meetme, whereas r291577 hangs up.

The additional hangup in 291577 is obvious (and arguably correct). Looks
like AST_SOFTHANGUP_ASYNCGOTO needs to be cleared before the next frame is
read.

Or.... Look at removing AST_SOFTHANGUP_ASYNCGOTO. It looks like it's set
in ast_async_goto, and unset in collect_digits and avoids an error state
(only for the first digit), and unset in __ast_pbx_run and avoids error
states. However, given that it is only set when there is a pbx, unsetting
when initiating a pbx would probably never occur. 

Issue History 
Date Modified    Username       Field                    Change               
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2011-01-10 20:18 wedhorn        Note Added: 0130368                          
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