[asterisk-bugs] [Asterisk 0011753]: app_channelredirect relies on ast_parseable_goto which fails to redirect channels

[noreply at bugs.digium.com]

The following issue has been ASSIGNED. 
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http://bugs.digium.com/view.php?id=11753 
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Reported By:                johan
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   11753
Category:                   Applications/app_channelredirect
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     assigned
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!): 98558 
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             01-12-2008 16:43 CST
Last Modified:              02-04-2008 09:13 CST
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Summary:                    app_channelredirect relies on ast_parseable_goto
which fails to redirect channels
Description: 
It seems like ChannelRedirect isn't working very well in trunk. I patched
ChannelRedirect to report the status in bug 0011553 for asterisk
1.4-trunk.
When I was porting this patch to trunk I stumbled on this issue.

The source av ast_parseable_goto that seems to fail even if there is a
vaild channel and destination. I have tested the following scenarious:
I place call http://bugs.digium.com/view.php?id=1 in either MusicOnHold(),
Meetme(), Playback() then I've a
call http://bugs.digium.com/view.php?id=2 that makes a
Channelredirect(call-numer-1-channelname,newcontext,newexten,1)
This always fails.

However if you do a core show channels after this unsucessful redirect you
will se:
Channel              Location             State   Application(Data)
Zap/pseudo-598993578 s at default:1          Rsrvd   (None)
SIP/callnumer1-08222 s at newcontext:0       Up      MeetMe(1,dm)

Note the newcontext:0...

And in the case you redirect channel http://bugs.digium.com/view.php?id=1 where
it does a Playback() the
redirect will occur after the Playback is finished. This will not happen
with the other applications thou.

Maybe I'm making a misstake, but this confuses me a lot...
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---------------------------------------------------------------------- 
 svnbot - 02-04-08 09:13  
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Repository: asterisk
Revision: 102272

U   trunk/main/pbx.c

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r102272 | file | 2008-02-04 09:13:23 -0600 (Mon, 04 Feb 2008) | 6 lines

Update handling of asyncgoto so it properly works on channels that are
currently executing a PBX.
(closes issue http://bugs.digium.com/view.php?id=11914)
Reported by: arnd
(closes issue http://bugs.digium.com/view.php?id=11753)
Reported by: johan

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http://svn.digium.com/view/asterisk?view=rev&revision=102272 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
02-04-08 09:13  svnbot         Checkin                                      
02-04-08 09:13  svnbot         Note Added: 0081660                          
02-04-08 09:13  svnbot         Status                   feedback => assigned
02-04-08 09:13  svnbot         Assigned To               => file            
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