[asterisk-bugs] [Asterisk 0013461]: H and h parameters do not hang up when * is dialed

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=13461 
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Reported By:                mboreback
Assigned To:                
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Project:                    Asterisk
Issue ID:                   13461
Category:                   Applications/app_dial
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.20 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             2008-09-11 10:34 CDT
Last Modified:              2008-10-14 11:56 CDT
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Summary:                    H and h parameters do not hang up when * is dialed
Description: 
I have a context used with a call file to connect two phones.
I use the h and H parameter, and are using one sip device and one iax
device in the connection.

I have the latest Trixbox version, Asterisk 1.4.20
When pressing the * nothing happens, I was hoping that the call should
end.

IS there a configuration issue here?

I notised that there was a closed issue case 0009905 last yeare but there
was not a clear solution.
Is the h and H parameter soposed to work in 1.4.20 and doues it require
any special configurations for SIP and IAX phone devices to work.

My Dial command includes sending the destination to an IVR via a macro
call, and this IVR includes an option to press *, and this is catched but
it looks like DTMF signaling catching is turned of after the call is
connected.
No indications in asterisk -rvvvvvvvvvvvvvvvvvv mode, it is only the sound
of the pressed key that is transmitted in the headset.
Both for Iax and Sip device.

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Relationships       ID      Summary
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related to          0013315 Dial h option not operating as defined ...
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 (0093615) blitzrage (administrator) - 2008-10-14 11:56
 http://bugs.digium.com/view.php?id=13461#c93615 
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You'll need to provide the relevant portions of the dialplan,
features.conf, and sip.conf in order to move this issue forward.

I see you're using trixbox -- please reproduce this issue on latest
Asterisk 1.4 with a vanilla Asterisk (i.e. not on trixbox) if at all
possible, as it very well could be a configuration issue.

Without the required information where we can reproduce the issue, it's
going to be hard to move this issue forward.

Thanks! 

Issue History 
Date Modified    Username       Field                    Change               
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2008-10-14 11:56 blitzrage      Note Added: 0093615                          
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