[asterisk-bugs] [Asterisk 0017068]: AGI->wait_for_digit or AGI->exec('Read' do not report digits back on an outgoing call

Asterisk Bug Tracker noreply at bugs.digium.com
Wed Apr 21 21:55:50 CDT 2010


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17068 
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Reported By:                xblurone
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17068
Category:                   Resources/res_agi
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           SVN 
JIRA:                       SWP-1320 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.1 
SVN Revision (number only!): 253347 
Request Review:              
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Date Submitted:             2010-03-21 05:35 CDT
Last Modified:              2010-04-21 21:55 CDT
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Summary:                    AGI->wait_for_digit or AGI->exec('Read'  do not
report digits back on an outgoing call
Description: 
Hi,

I am using a callback conference solution which makes and outgoing call
which points to a perl AGI script (which used to work some versions of
asterisk ago). Now I am running the latest SVN (Asterisk
SVN-branch-1.6.1-r253347) and neither Read nor wait_for_digit give any
digits back on DAHDI channels, or SIP channels.

The same script has worked for a few years from version 1.4 onwards. I
can't remember when it stopped working however (sorry).
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---------------------------------------------------------------------- 
 (0120734) xblurone (reporter) - 2010-04-21 21:55
 https://issues.asterisk.org/view.php?id=17068#c120734 
---------------------------------------------------------------------- 
lmadsen, how would I do that? (sorry if this is a very stupid question). I
was under the impression this is a bugtracker, and this clearly is a bug,
and it is a pressing one I can't work around... How would asterisk-biz help
me here? 

If I regress to asterisk-biz, I am having more than one critical bug
(mostly sip and stability related) to deal with.

To me this is still a major bug, and as you can see I am not alone in
this.

If anybody can tell me their version 1.6.1 or 1.6.2 asterisk where this
still works, I'd gladly try to reverse to that version and see if my other
issues don't come back.

~ Please help ~ 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-04-21 21:55 xblurone       Note Added: 0120734                          
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