I’m a totally * newbie, but want to move from Dialogic SDK to Asterisk.
Is the headline question possible?
My head-scenario with Asterisk is:
A. 5 SIP extensions (hardware, not softphones), with headset and auto-answer enabled.
B. 1 SIP trunk (multiplies channels and multiplies DID’s).
C. some kind of queue where all incoming trunk-calls are set on hold.
D. the 5 SIP extensions knows nothing about the queue, but a ‘booking-software’ should be able to ‘redirect’ (over socket AMI) one of the queue-calls (after own choice) to an SIP-extension. (The auto-answer on the SIP-phone will automatically pick up the call and the operator got the call/customer in the headset… without touching the phone or headset).
Furthermore socket-AMI should be able to voice-record the call, put it on hold (again) and finally drop the call.
Is this possible?