[Asterisk-Users] problems with analog interface to PBX

Jason Williams jason at redskycomputing.co.uk
Wed May 19 02:12:25 MST 2004


That is the way it works over one zap channel, to keep * in the call it 
would need to dial out on anoher line and that would then use an additional 
zap interface and tie it up for the duration of the call.


Jason

At 20:30 18/05/2004 -0300, you wrote:
>Yes, I've tried with SendDTMF, and it works, but if I do that, then * looses
>control of the call. That is, the call is transfered to the new extensions
>on the PBX but since * is not in the calll flow anymore, it doesn't know if
>on the other end they have ansered or not.
>
>
>----- Original Message -----
>From: "Steven Critchfield" <critch at basesys.com>
>To: <asterisk-users at lists.digium.com>
>Sent: Tuesday, May 18, 2004 5:56 PM
>Subject: Re: [Asterisk-Users] problems with analog interface to PBX
>
>
> > On Tue, 2004-05-18 at 15:45, Dan Fernandez wrote:
> > > Steve,
> > >
> > > Thanks for your respnose. The flash does seem to work. If I plug a phone
>on
> > > the x100p I can hear with the x100p flashes. I then get a dialtone. The
> > > problem is that when i try to dial again from that card, i get "cannot
> > > create zap channel". It seems that because the line is now off hook, the
> > > dial cannot proceed.
> >
> > Without having read the thread, flash returns you to the channel. From
> > that point use senddtmf to "dial" the numbers you want on the channel
> > you already have.
> >
> > > ----- Original Message -----
> > > From: "Steve Creel" <screel at turbs.com>
> > > To: <asterisk-users at lists.digium.com>
> > > Sent: Thursday, May 13, 2004 11:04 AM
> > > Subject: Re: [Asterisk-Users] problems with analog interface to PBX
> > >
> > >
> > > > 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
> > > > _______________________________________________
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> > > >
> > > _______________________________________________
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> > --
> > Steven Critchfield  <critch at basesys.com>
> >
> > _______________________________________________
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