asterisk users - Jan 2006 - FW: Re: hangup detection

Thanks for your suggestion Steve.
I have done as you advised and set  busypattern=300,200 to match the sample
I recorded.
This hasn't worked though, asterisk doesn't seem to detect the busy
signal.
Does asterisk require a the signal to be in a certain power range?  The
signal I get
is very quiet.
Thanks for your help
Regards
Jonathan



 
On Mon, 19 Dec 2005, [ISO-8859-1] Diego Andr?s Asenjo Gonz?lez wrote:


> Hi everybody!
> 
> Jonathan wrote:
> > 
> > Hi,
> >  
> > I'm using a td400p card with an FXO port and asterisk 1.2.1 in
South
> > Korea and asterisk isn't detecting when PSTN callers hangup.
> > I've gone through all the settings related to hangup detection and
none
> > work.  I've tried:
> >     hanguponpolarityswitch=yes
> >     callprogress=yes
> >     busydetect=yes
> >     busycount=6  
> I'm using asterisk/zaptel 1.0.10 and have the same situation. I'm
in
> Colombia and tried with a lof of loadzone
> >  
> > Debug doesn't show reverse polarity events so I'm pretty
stuck.
> >  
> > I've got zaptel configured with a loadzone of US and kewlstart
signialling.
> >  
> > Has anybody had success with these cards/asterisk in South Korea? 
> ?Or in the world?
> >  


We implemented a busypattern= option for the zapata.conf that might help 

you.



Test like so:  Dial into your Asterisk system via the FXO port to an 

extension on your box.  Now hang up from the outside.  Listen to the call 

on the internal extension.



If you hear a regular beep-beep tone of some sort, busypattern= might help 

you.



You need to time exactly the length of the beep and the length of the 

silence.  (To get it nice and accurate, record it, then load into 

Audacity and measure).



Say it comes out at 750 msec of beep, 500 msec of silence.  Then adjust 

your zapata.conf like so:



busydetect=yes

callprogress=no

busypattern=750,500

busycount=4



Regards,

Steve Davies



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A little background.  I'm integrating asterisk as the voicemail service
for an old Meridian/Norstar pbx which has an ATA-2 connected.  The ATA-2
is used to connect an analog device (such as a voice modem) to the pbx.
 In the past we've used vgetty and a voice modem with varying degrees of
success.

The problem is the ATA-2 does not provide disconnect supervision.  When
the outside caller hangs up, the line just goes silent.  No busy signal.
   If the inside analog device hangs up the call, the ATA-2 then
provides a dial tone.

Right now, we've been using silence detection in voicemail to determine
when the call ends.  This works most of the time.  On longer calls (say
more than 5 minutes), sometimes the silence detection does not function
and the call continues (with a bunch of recorded silence) until the
maximum voicemail length limit is reached.

Obviously, busy-detect is not going to work.  The current silence
threshold is set at the default 128.  The FXO module is configured with
Kewl Start signalling.  Would I get better results (or any different
results) if I switched to Loop Start?

Any other ideas?

Thanks,

Darrick
-- 
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com