[asterisk-bugs] [Asterisk 0017149]: Busy(xx) exits immediately on IAX channel

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17149 
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Reported By:                jlamanna
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17149
Category:                   Applications/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           Older 1.4 - please test a newer version 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-04-08 00:30 CDT
Last Modified:              2010-04-08 08:27 CDT
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Summary:                    Busy(xx) exits immediately on IAX channel
Description: 
I'm running Asterisk 1.4.26.3 and I've noticed an interesting problem
when trying to play a Busy tone over a IAX trunk from the PSTN.
It seems as though Busy(20) returns immediately (it does not
wait 20s), so the caller never hears the busy tone, but
the call just appears to hang up.
I don't believe this happens when trying to play a Busy on a SIP trunk.

The busy part of the dialplan looks like this,

exten => s-BUSY,1,Noop(Dial failed due to trunk reporting BUSY - giving
up)
exten => s-BUSY,n,Playtones(busy)
exten => s-BUSY,n,Busy(20)

The only way to remedy this is to put a Wait(20) between the
Playtones() and Busy().
Any ideas on why this only fails on IAX and not SIP?

I cannot test a later version of Asterisk because of the SIP relay bug.
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 (0120205) pabelanger (manager) - 2010-04-08 08:27
 https://issues.asterisk.org/view.php?id=17149#c120205 
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Thank you for taking the time to report this bug and helping to make
Asterisk better. Unfortunately, we cannot work on this bug because your
description didn't include enough information. You may find it helpful to
read "Asterisk Issue Guidelines"
http://www.asterisk.org/developers/bug-guidelines. [^] We'd be grateful if
you would then provide a more complete description of the problem.

At a minimum, we need:
1. the specific steps or actions you took that caused you to encounter the
problem,
2. the behavior you expected, and
3. the behavior you actually encountered (in as much detail as possible).

This likely includes output from the console with debug level logging, a
SIP trace (if this is SIP related), and configuration information such as
dialplan (e.g. extensions.conf) and channel configuration (e.g. sip.conf).

Thanks! 

Issue History 
Date Modified    Username       Field                    Change               
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2010-04-08 08:27 pabelanger     Note Added: 0120205                          
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