Robby Dermody
2005-Nov-21 17:27 UTC
[Asterisk-Users] inquiry into Asterisk scripting scenario (VXML, AGI, IVR, etc)
Hey guys,
I've written an VoIP monitoring and recording application that pulls
VoIP phone call data off of the network (in real time) and makes it
available for monitoring. Besides being able to listen into these
conversations in progress through a web interface, we can actually
monitor these conversations over the phone as well. This is done through
a VoiceXML script sitting on a Cisco router (which is acting as a
VoiceXML voice browser). In this case, the user dials the system (i.e.
the router), and the voiceXML script starts executing. The user logs in
to my application over the phone through this script, and can listen to
a specific agent talking or just sample the agent conversations going
through the underlying IPT environment. The playback of the
conversations makes use the ability of the Cisco VoiceXML browser to do
audio-streaming through the <audio> command.
Our goal is to move this functionality off of a cisco router, and onto
the same box as my app, using Asterisk (or whatever other open source
software may be better suited for this). Basically, Asterisk would be
set up as a SIP/H.323 endpoint. If any user wanted to dial into our
voice monitoring system, they could just be transferred from the outside
IPT phone environment (be it Cisco, Avaya, etc) to our box, and Asterisk
would service the call by interacting with my app. This way, there is no
extra Cisco router to support, interacting with my app over the phone
interface is as easy as setting up a route point in the outside IPT
phone environment, and everything fits nicely on one box.
I am an Asterisk newbie, and at this point am just asking the question,
"is this possible, and how might I go about it (what should I be
researching)?" I have done some research, and there doesn't seem to be
any real open source VoiceXML 2.x browser implementation that would work
with asterisk. From what I can tell, that leaves IVR scripting (such as
http://users.pandora.be/Asterisk-PBX/IVR.htm) and AGI
(http://home.cogeco.ca/~camstuff/agi.html). Note that this voice
application, while not horribly complex, is somewhat complex and
requires things such as the following:
*Ability to interface with the application via HTTP (sending URL
requests and reading back received data): Every interaction between the
IVR script and my application is done via HTTP. For example, the command
to login a user (once the IVR script has gotten a userID and PIN) may
be: http://localhost:1234/login?userID=444&pin=4321, which returns a
success/failure code.
*G.711 (ulaw) audio streaming via a URL: Monitoring of agents is
done by submitting a URL request to my application, which will start
streaming back G711 ulaw data (which is the conversation in progress) as
it arrives. I see that Asterisk has a MP3Player command that allows
streaming mp3 playback. If I can write my own commands, it probably
wouldn't be too hard to do the same for g711 ulaw OR linear PCM, if this
functionality isn't already available.
*Dynamic prompt construction: The IVR interface also allows a user
to list the agents active in the environment. With the current voice XML
implementation, this is done by querying my application for the list,
which my app returns as a voicexml subdialog which control transfers to.
This kind of thing is where AGI might win out as the solution to take
(as I can whip up and massage the list dynamically easily with a
language like Python)
I would appreciate opinions of the best way to go about converting this
functionality to run on Asterisk. I'm especially interested in AGI, but
the documentation I can find on it is rather out of date it appears. If
there are any gurus in this field who are interested in helping out with
something like this, let me know and I could very possibly compensate
financially, if desired. BTW, if I went with AGI, I would be working
with Python (the whole app is written in Python/Pyrex, believe it or not! :)
Thanks!!!
Robby