[Asterisk-Users] Still trying program -> phone call

[Andy Powell]

John,

Yes, asterisk can do that, and in fact it's very simple. The problem at the moment is your level of knowledge of asterisk, but this can be resolved... 

There are a number of things you need:

1 Access to the PSTN - this can be done via a single X100P card (plugs into a standard phone line) or one of the sinlge port T1 cards or 4 port TDM410 cards (if you need a shedload of lines). You can also use a VoIP -> PSTN gateway or gateway service (such as, but not limited to, NuFone)

If you use the X100P, then as I say, a standard analogue phone line is all you need (you can add upto 3 X100P's iirc without issues).
If you want to use the T1 cards then you need to get your local telco to deliver T1's to your location 

The process I think you are trying to create would be something like this:

Your script reads a database and generates call files for each person it needs to call. The call file contains the number to ring and the context and extention to deliver the called person to. This extension simply runs an AGI script that plays the menu and waits for user input ( various things can be done if the user does not respond eg flag a potential user in distress) - this information is then fed back to the database so that you can report on it... (or alternatively raise an immediate alarm). The AGI can be scripted in almost any language you like. 

Personally I think the medical service should just employ more people for home visits, since seeing a person is better than just hearing an automated voice on the phone.. (you may have differing opinions, but I come from a culture of free healthcare (however bad it is at the moment))

You are correct, the WiSIP is just a distraction, besides, from what I saw at CeBit, wait a few months and you'll have more choice. I'd suggest getting a copy of SJPhone (what we call a softphone - ie it's software not hardware) from www.sjlabs.com - it's a nice simple interface.

It looks to me like you put  800 with a context of callme in your .call file...

I suggest that you abide by the adage, "learn to walk before you run" ... You can take a look at my guide at http://www.automated.it/guidetoasterisk.htm (there are others) which may help clear up one or two points of understanding...

You could of course pay my air fare to Boston (and back) and hotel costs and I'd gladly help you out in person.. after my time at VON in Boston last year I wouldn't mind visiting again :D

HTH

Andy



*********** REPLY SEPARATOR  ***********

On 01/04/2004 at 17:35 John Chambers wrote:

>A while back, I asked about using Asterisk in a medical environment where
>the task
>is to write a program that connects to a phone and sends a message like:
>
>     Hello Mrs. Jones. How are you doing today? Press 1 if you're
>     OK.   Press  2 if you need help.  Or start talking, and your
>     message will be passed to a person.
>
>After connecting and sending the sound file, the program would obviously
>need to
>listen for keys and voice, and do something sensible with them.
>
>Since then, I've done a bunch of installing, testing, and especially
>experimenting
>with variants of the sample.call file.  So far I haven't been able to
>answer the
>question of whether what's wanted is possible.  Maybe a couple of
>questions that
>we've come up with will clarify things.
>
>One is whether we can make a call to a regular land-line or cell phone, or
>just
>to VoIP-type phones.  If it's possible, what do we need to know about
>routing?
>We're guessing that we need to somehow relay through some sort of
>IP-to-PSTN
>gateway, but information on this seems to be rather muddy.
>
>Asking our local telcos (and Boston has a bunch of them ;-) gets a lot of
>clueless
>responses. If we mutter the acronym VOIP, they perk up and start trying to
>sell us
>their promised VOIP phone service.  But this has nothing to do with what
>we want
>to do, which is to get a *program* to make the call.  This obviously
>implies that
>the connection escapes from the IP cloud and enters the PSTN cloud, but
>how? If
>we need to purchase service with some gateway provider, how do we ask for
>it?
>
>Actually, I've been really tempted to get a WiSIP phone, to get familiar
>with
>that.  But as far as I can tell, it would just take time away from the real
>project, so I haven't. OTOH, if using it would make VoIP clearer to us
>newbies,
>maybe it would be a good idea. Or maybe a softphone on my Powerbook would
>be
>a better way to go.  Or both?  In any case, talking to a SIP phone isn't
>very
>interesting to us yet, since few people have them.  A demo would have to be
>to the phones on people's desks or in their pockets.
>
>Meanwhile, another sort of question is how to find explanations of
>asterisk's
>many cryptic error messages.  For example, after cleaning stuff out,
>downloading
>from CVS, doing a make and make install (and ignoring errors ;-), then
>firing
>up "asterisk -vvvc" and copying one of my test*.call files to the outgoing
>directory, I got:
>
>*CLI>
>     -- Attempting call on Zap/1/12223334444 for 800 at callme:2 (Retry 1)
>Apr  1 16:29:08 NOTICE[17424]: channel.c:1563 __ast_request_and_dial:
>Unable to request channel Zap/1/12223334444
>Apr  1 16:29:08 NOTICE[17424]: pbx_spool.c:199 attempt_thread: Call failed
>to go through, reason 0
>
>(I replaced my home/cell number with 2223334444 for illustration's sake.)
>Anyway,
>I haven't yet succeeded in digging explanations out of www.voip-info.org. 
>And
>that 800 at callme:2 is a bit of a mystery; I see it all the time but have no
>idea
>why it's there.
>
>Further clues (or hyperlinks to clues) are welcome.
>
>I would like to report that "Yes, Asterisk can do that."  So far my report
>has
>to be "Well, maybe, but I can't really tell yet."
>
>(We're also well aware of the serious potential for phone spam in this
>project.
>My reaction tends to be "Not if I can block it."  But it's something that's
>obviously going to be a problem in a few years. ;-)
>
>
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