asterisk users - May 2004 - problems with analog interface to PBX

Folks,

For the last few days I've been trying to experiment with a Panasonic  PBX
and an X100P but have run into quite a few problems which I am not sure if they
can be solved with this type of card (how about TDM01B?)

1) I wanted to use *'s IVR capabilities, so I routed the calls to the
extension where the x100p was connected to.

Asterisk should answer the call, playback a message, dial another PBX extension
and if no one answers dial another extension (via IAX).

The first problem I ran into was that the Flash application doesn't really
work. To get around this I added another x100p to dial the new extension. The
problem I ran here was that even though I specified in the Dial app to just dial
for 30 seconds, it rang forever as if * cannot recongnize that no one had picked
up.  Asterisk does seem to detect hangups and busy tones (I have busydetect=yes
and busycount=10)

Am I trying to do something that the x100p is not capable of?  Would making
changes to the indications.conf help at all?

2) I would also like to use * for voicemail. The user should be able to dial the
extension where the x100p is connected and asterisk recognized the extension the
user is dialing and request for the password? Is this possible?

Thanks
Dan


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On Wed, 12 May 2004, Dan Fernandez wrote:
> Asterisk should answer the call, playback a message, dial another PBX
> extension and if no one answers dial another extension (via IAX).
> 
> The first problem I ran into was that the Flash application doesn't
> really work. To get around this I added another x100p to dial the new
> extension.  The problem I ran here was that even though I specified in
> the Dial app to just dial for 30 seconds, it rang forever as if * cannot
> recongnize that no one had picked up.  Asterisk does seem to detect
> hangups and busy tones (I have busydetect=yes and busycount=10)
In the absence of call progress detection settings, Zap analog channels
tell Dial() that they are Connected more-or-less as soon as they have
completed dialling (I see this on the display of my 7960: I see Proceeding
for a second or two, then Connected, when I dial through an X100P).  So,
the timeout on your Dial() never gets triggered because the channel
reports a connected call almost straight away.

To do what you want, you would need callprogress=yes -- as long as your
Panasonic PBX generates authentic US tones.  busydetect will only detect
busy (!), not ringback or congestion or any of the other tones you would
need to make your application work the way you want -- call progress 
detection tries to do this for you.

The bad news is that even if your PBX generates US tones, reports are that
the detection is not too reliable.
> Am I trying to do something that the x100p is not capable of?  Would
> making changes to the indications.conf help at all?
It's not that the X100P can't do the job, it's more that analogue
lines
can't do the job :)  Seriously, if your PBX generates US tones then give 
callprogress=yes a try.  From my reading of the code, the tones specified 
in indications.conf are unrelated to the way the * DSP does call progress 
detection (have a look at functions like ast_dsp_call_progress() in dsp.c 
if you're really curious).
> 2) I would also like to use * for voicemail. The user should be able to
> dial the extension where the x100p is connected and asterisk recognized
> the extension the user is dialing and request for the password? Is this
> possible?
On an analogue channel via an X100P, there is no "called number"  
indication.  So you can't tell what number the caller dialled to reach
you.  If you wanted to use the * box as a voicemail-only machine, you
could drop the caller straight into VoiceMailMain, but if you wanted other
functions (conference rooms, VoIP gateway, etc) you would need to use an
IVR...

   "press 1 to access Voicemail...
    press 2 to reach a Voice-over-IP user...
    press 3 to join a conference...
    ..."

This doesn't really help your original need: to dial another number on the
PBX and get control back if needed.  If callprogress=yes doesn't work for
you, you could try something like this (off the top of my head):

exten => 4,1,Playback(trying-press-*-to-come-back)
exten => 4,2,Dial(Zap/1/1234,,Hg)
exten => 4,3,Goto(103)
exten => 4,103,Playback(sorry-cant-reach)
exten => 4,104,Goto(menu,s,1)

On the Dial(), the option H enables caller hangup using '*', and g says
go
on in context when the destination channel hangs up.  This would put your
caller in the driver seat and get them to do the tone detection for you ;)


Hope this helps,
Vic Cross

On Wed, 12 May 2004, Dan Fernandez wrote:
>Folks,
>
>For the last few days I've been trying to experiment with a Panasonic
PBX
>and an X100P but have run into quite a few problems which I am not sure
>if they can be solved with this type of card (how about TDM01B?)
>
>1) I wanted to use *'s IVR capabilities, so I routed the calls to the
>   extension where the x100p was connected to.
>
>Asterisk should answer the call, playback a message, dial another PBX
>extension and if no one answers dial another extension (via IAX).
>
>The first problem I ran into was that the Flash application doesn't
>really work. To get around this I added another x100p to dial the new
>extension. The problem I ran here was that even though I specified in the
>Dial app to just dial for 30 seconds, it rang forever as if * cannot
>recongnize that no one had picked up.  Asterisk does seem to detect
>hangups and busy tones (I have busydetect=yes and busycount=10)
For about 6 months, we were using the same logical setup (a channelbank of
FXO cards for a Merlin Legend switch, with asterisk doing incoming IVR /
autoattendant, then transferring the calls out to the Legend, and
handling voicemail).  The first problem I encountered that I hadn't
expected had to do with asterisk transferring the call back to the Legend.
I did a Flash(), a SendDTMF(), and another Flash() - the Legend saw this
as an attended transfer, and it caused some oddities.  Turns out I needed
to Flash(), SendDTMF(), Hangup().  Along the way, I found the Flash times
that the legend was expecting to see, and adjusted them in the source
code, so as to eliminate occasional flash detection problems.

I'd take time to plug an analog set into the extension you have the X100P
on, and make sure you can flash/transfer calls like you're expecting
asterisk to.  There's no reason (that I know of) that your flash can't
give you exactly the behavior you're looking for.

Good luck to you,

Steve