[Asterisk-Users] problems with analog interface to PBX

[Dan Fernandez]

Thanks for the response.

Have you try the new TDM FXO cards?  Does call progress work with those?


----- Original Message ----- 
From: "Vic Cross" <vicc at veejoe.com.au>
To: <asterisk-users at lists.digium.com>
Sent: Thursday, May 13, 2004 5:46 AM
Subject: Re: [Asterisk-Users] problems with analog interface to PBX


> 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
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>