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

[noreply at bugs.digium.com]

The following issue has been RESOLVED. 
====================================================================== 
http://bugs.digium.com/view.php?id=11607 
====================================================================== 
Reported By:                sobomax
Assigned To:                qwell
====================================================================== 
Project:                    Asterisk
Issue ID:                   11607
Category:                   Channels/chan_zap
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     resolved
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:              
Resolution:                 not fixable
Fixed in Version:           
====================================================================== 
Date Submitted:             12-20-2007 04:40 CST
Last Modified:              01-15-2008 15:15 CST
====================================================================== 
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
====================================================================== 

---------------------------------------------------------------------- 
 qwell - 01-15-08 15:15  
---------------------------------------------------------------------- 
Unfortunately, this isn't something that can be fixed, and that's due to
issues with analog signaling itself, rather than something inside of
Asterisk.

The easiest way around this would be to answer the incoming call before
sending it on to the SIP device, but that would cause a CDR to be created,
ringing would be stopped, and your provider would likely charge you for the
call.  You could also try to do some dialplan magic, and keep the incoming
side of the call up after the Dial(), and check DIALSTATUS, then do an
answer and immediate hangup. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
01-15-08 15:15  qwell          Note Added: 0078052                          
======================================================================