Are some Asterisk experts able to judge the feasability of the following project and guide me about the best approach?
I want to build a ‘web-driven phone chat’ system driven by our corporate
Intranet (J2EE based). The required functionality can be broken down into these steps:
- The system calls out to an external phone number (initiated by J2EE server, which also provides the phone number to call)
- This call initially would be connected to music.
- On a signal from J2EE the call would get connected to a specified MeetMe room.
- On a further signal from J2EE the call would get disconnected from MeetMe and reconnected to music. [There could be further iterations of 3) and 4).]
- Finally, on a signal from J2EE the call would get disconnected completely.
- The J2EE server should be able to get feedback about the current phone status of the call (caller picks up, hangs up …).
Can this be done with Asterisk alone (? I am especially not sure about the transitions between 2), 3), and 4). (Essentially, an ongoing phone conversation gets rerouted without any commands coming from the phone.)
If doable: which Asterisk commands would I use for the various steps?
If not doable with Asterisk alone: what’s the best way to approach this? A J2EE-driven softphone for each user that makes the outbound call to that user’s phone and conferences in the music, the meetme room, etc. as needed?
Have looked at documentation and various sites, but it’s not easy to get a feel for the right approach. Any guidance is greatly appreciated.