Hi,
There is a library called Asterisk Java that uses the Asterisk Manager
interface to send Asterisk commands and get notified of new channels
etc.
Using that I built a GUI around it to move calls about. Its really
simple, I have two queues in Asterisk, the outside numbers go direct to
the incoming queue. Anything in that queue flashes blue/grey for
"ringing".
Once you answer a call it pre-populates the input box so you get the
caller history etc.
When you put the call on hold it goes into queue 2 and turns green to
signify ready for on air.
A limitation of the Asterisk Manager interface is its not really
suitable for lots of clients. You can use Asterisk Proxy programs but
it was a bit complicated for me so I wrote a Server component that
interfaces with the Manager Interface. It then uses XMPP to send simple
control messages for call incoming, transferring, answering etc.
All I'm doing is using a £50 Cisco IP Phone in the studio (so far Cisco
and Grandstream are the only ones I've tested with the Auto Answer
header, I've tried various software phones but none of them work with
Auto Answer). I'm then wiring the headset socket into the studio desk.
I haven't implemented conferencing yet so you can only do one on air at
a time (you could keep clicking different people to change who is on air
but they couldn't talk to each other at the same time).
There are privilege levels that allow some people to steal calls from
others if needed too.
I've got it all on a local git hub server at the moment. I'll look
into throwing it online once I've got some docs together to set it up.
On 2013-10-26 10:24, Hoggins! wrote:
Hello Wayne,
In our little student radio, we're using Asterisk, and I'm very
interested in your project.
How does it work ?
Our studio uses a "central" Asterisk server for routing calls in and
from the outside world, and between the different extensions in the
facilities. There is another Asterisk server that acts as a client :
it's the hybrid system, hosted on a computer, with an ADAT soundcard
connected with JACK. And we use the app_jack feature of this Asterisk
to
connect directly to the soundcard. There is a permanent open channel
between this Asterisk and the main one, on a MeetMe extension, so
that
it's possible to reroute new incoming calls to this MeetMe
conference.
This allows us to have an "infinite" number of callers on air,
talking
to each other.
The main difficulty until now is to remove a caller from the
conference.
We have to do this manually, so a technician must be present to enter
commands in the CLI. This is absolutely not good.
So waiting for the new Asterisk 12 with a REST API, your solution
looks
very promising, and I would like to test it if possible.
Hoggins!
Le 25/10/2013 19:56, Wayne Merricks a écrit :
Hi all,
I've been making a phone interface for Studio and phone answerer use
over the last 3months. I don't get a lot of free work time but its
been making steady progress.
If any of you rely on Asterisk as a backend and don't mind running
Java on machines, it might be useful for you.
Please note this is very early days as all the setup for databases
etc
is manual at the moment. It requires Asterisk, MySQL and an XMPP
server (I'm using Elastix in a VM).
Would be interested to know if this is useful to anyone and possibly
if anyone is brave enough to test it. It should be functionally
complete over the next couple of weeks, there are only minor bugs
that
I know of at the moment (stupid things like a caller being on a
Warning alert, the icon shows up correctly but the drop down box
defaults to Banned).
Screenshot would have got eaten by the mailing list so can be found
here:
http://thevoiceasia.com/phone.png
The plan is to put this into production over here sometime before
Windows XP dies next year. I've tested it on Windows and various
forms of Ubuntu so far. Any questions feel free to get in touch.
Regards,
Wayne Merricks
The Voice Asia
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