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[jsa at svep.se]

Hi there dear users and dear developers of Asterisk!
I've got a maybe strange idea, let's see if you laugh or think it's reasonable J

I'm using Asterisk with Digium TDM800P cards with 24 lines (as an answering machine).
Each analog line can be reached through a phonenumber, since they are each connected to my telephone provider. Yes expensive I know J


The challenge:

I'd like to somehow verify that my 1) TDM800P cards and 2) the analog lines, and 3) my operator is alive and working, and I have an Idea which I wonder if will/could work.

My first idea was to ask the zap-driver if it could detect if the line was ok, but no function existed to do that, what I could find. Anyone knows about one?

My second idea, was to try calling simply, to know if things were ok. But, I couldn't just call any number, I had to know the number was in use, and not disturbing anyone.
So, I called myself, or I called another of my phonelines.

So,
I'd like to use the asterisk manager interface in java to originate a call from one ZAP-channel, calling out to my telephone provider,
And then they will direct the call back to my, but into another ZAP-channel (since I'm calling that channel's number).

So: I'm making ZAP/1 calling out to no 323121321 -> telephone company, Ok: 323121321 belongs to this guy -> redirecting me to my ZAP/2 channel, which answers the call.

Then I have a connection, and ZAP/2 will answer and do some DTMF.
My first ZAP/1 is run through my java program, and I'd like to listen for certain DTMF-tones, to know I have a working circuit.

The goal for all of this, is to verify things are working, so my provider is not down, or one of my ZAP-lines are dead.

So far, I've tried calling and got some half-success, but not sure what is going on doing all the right way.
For ex: why am I calling with Zap1, to Zap 3, and then Zap 7 is answering? 3 channels used for one outgoing and one incoming call? Something must be very wrong J
Please educate me, dear experts.

Input?


Sincerely,
Johan Sandgren
www.svep.se<http://www.svep.se>, jsa at svep.se
Frosty Sweden but with some sunshine today !! J

My code and settings below, for information.

=============JAVA CODE (extract)

OriginateAction originateAction = new OriginateAction();
ManagerResponse originateResponse = null;
originateAction.setChannel("ZAP/1");
originateAction.setContext("Outgoing");
originateAction.setExten("201"); // maps to ZAP/7 through external phonecompany PBX
originateAction.setPriority(new Integer(1));
originateAction.setTimeout(new Long(15*1000)); // xml-milliseconds
originateAction.setAsync(false);

============== extensions.conf (extract)

[Incoming]
exten => s,1,Set(DYNAMIC_FEATURES=hangup)
exten => s,2,Agi(agi://localhost/answer.agi)

[Outgoing]
exten => _X.,1,Set(DYNAMIC_FEATURES=hangupfeature)
exten => _X.,n,Dial(Zap/3/${EXTEN}

============== Asterisk response:

AGI Debugging Enabled
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'stt' logged on from 127.0.0.1

[tzafrir.cohen at xorco...]

On Thu, Feb 14, 2008 at 01:17:45PM +0100, Johan Sandgren wrote: