asterisk users - Feb 2004 - Norstar Integration with Asterisk via FXO or BRI ISDN

Hi,

I have a legacy Norstar system that I'm looking into integrating with my
Asterisk setup. 

My first attempts have worked, which involves a Wildcard X100P FXO card in
the * box connected to the Internal ATA (FXS port) on the Norstar system.

Calling from SIP -> Norstar works fine, since the SIP caller initiated the
call and generally will be sane enough to hangup the phone when complete,
making asterisk hangup the X100P.

However in the opposite direction, Norstar -> SIP, the problem begins,
thanks to no disconnect supervision/provision on the FXS port from the
Norstar, so the X100P doesn't know the Norstar caller has hangup.

A thought that occurred was perhaps Asterisk could do some sort of
"soft"
disconnect supervision? So when the FXO card has seized the line, but there
is no transmit/receive audio (or no major variation from the standard
background radiation) then after a safe timeout of say a minute or two, it
could disconnect the line? I'd be interested to hear others views on this,
would it work (or does it already exist?).

Another thought is that the Norstar has both an 4-port FXO trunk card and a
4-port BRI-ISDN trunk card. The 4-port FXO card is completely free for use,
however it doesn't support disconnect supervision, so same problem there.

However, the BRI-ISDN card obviously does support disconnect supervision
_and_ Direct InDial, even better! Only two ports are in use, so I could
easily borrow one.

At 10:59 PM +1000 2/2/04, Christopher Lee wrote:
>Hi,
>
>I have a legacy Norstar system that I'm looking into integrating with my
>Asterisk setup.
>[snip]
>
>However in the opposite direction, Norstar -> SIP, the problem begins,
>thanks to no disconnect supervision/provision on the FXS port from the
>Norstar, so the X100P doesn't know the Norstar caller has hangup.
>
>A thought that occurred was perhaps Asterisk could do some sort of
"soft"
>disconnect supervision? So when the FXO card has seized the line, but there
>is no transmit/receive audio (or no major variation from the standard
>background radiation) then after a safe timeout of say a minute or two, it
>could disconnect the line? I'd be interested to hear others views on
this,
>would it work (or does it already exist?).
>[snip]
>Cheers,
>Chris Lee
No, though there has been some discussion about similar problems with 
disconnect supervision with SIP.  Of course, the SIP disconnect 
supervision is packet-based (where there has been no reinvite) and 
your problem is audio-based; it's the same method, though - check to 
see if there is activity on the session in the TX or RX direction, 
and if no activity crossing threshold Y for more than N seconds, hang 
up.

See http://bugs.digium.com/bug_view_page.php?bug_id=0000207  for more 
discussion.

I think this is pretty reasonable, actually.  We have such a system 
in voicemail - if there is silence at the end of a message, the 
system hangs up and rewinds the VM recording to the last spike in 
audio energy.  You will get people yelping at you about how this 
breaks conference calling, or maybe it breaks prank calling.  My 
answer to this is that it should be a selectable switch/values on the 
Dial.  app_dial sure is getting crowded with ideas, but is still 
pretty lean on features.

Anyone feel like a) programming this or b) funding it?

JT