[asterisk-bugs] [Asterisk 0011607]: POTS->VoIP calling defficiency

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A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=11607 
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Reported By:                sobomax
Assigned To:                
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Project:                    Asterisk
Issue ID:                   11607
Category:                   Channels/chan_zap
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:            SVN 
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:             12-20-2007 04:40 CST
Last Modified:              12-20-2007 16:18 CST
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Summary:                    POTS->VoIP calling defficiency
Description: 
There is a problem with call that comes via analogue Zaptel interface and
then is hanged up before it has been answered there is no way to
communicate rejection to the POTS side (i.e. you can't hang up before it
has been aswered). Therefore when the next ring pulse comes in shortly on
the same POTS call it's considered to be a "new" call and so that the new
call is initiated. This continues until the calling party on the POTS side
hangs up. I have attached very crude patch that we have had for years to
help with this situation by waiting until the calling party in turn hangs
up (by detecting ring timeout).  

-Maxim
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---------------------------------------------------------------------- 
 sobomax - 12-20-07 16:18  
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Any analogue PSTN hardware will have this problem. TDM400P is the one in
our case.

-Maxim 

Issue History 
Date Modified   Username       Field                    Change               
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12-20-07 16:18  sobomax        Note Added: 0075769                          
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