[asterisk-bugs] [Asterisk 0011389]: While you are dialing from ATA/listening to the ring , if you press touchtone, then hangup, the other line rings forever.

noreply at bugs.digium.com noreply at bugs.digium.com
Tue Nov 27 16:02:51 CST 2007


A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=11389 
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Reported By:                andrewgray
Assigned To:                
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Project:                    Asterisk
Issue ID:                   11389
Category:                   Applications/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:            1.2.24  
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:             11-27-2007 12:47 CST
Last Modified:              11-27-2007 16:02 CST
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Summary:                    While you are dialing from ATA/listening to the ring
, if you press touchtone, then hangup,  the other line rings forever.
Description: 
Verified on Asterisk 1.2.24, 1.2.14, 1.2.13, 1.2.9.

Using an ATA, if you dial another extension, and while you are listening
to it ring, if you press a touchtone key sending out a DTMF tone, then hang
up your phone, the other phone rings forever.
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---------------------------------------------------------------------- 
 andrewgray - 11-27-07 16:02  
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OK, we loaded 1.4.14 and the same thing happened. We dialed an extension
using an ATA, and while it was ringing, we pressed a DTMF key, and then
hung our phone up.  The other phone kept right on ringing.  Except on the
sip debug screen, it said that there was going to be timeout in 60 seconds,
and sure enough, it quit  ringing after a minute or so.  So we went back to
1.2.24 and it also times out and stops ringing after a minute.  

With sip debug ON, there is so much activity on the terminal that it is
impossible to cut and paste the sip debug output.  It scrolls by too fast. 
So we are not able to get a sip debug output.  But it does it every time. 
So it is not hard to reproduce.


Andrew Gray 

Issue History 
Date Modified   Username       Field                    Change               
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11-27-07 16:02  andrewgray     Note Added: 0074438                          
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