Need to setup Asterisk, the server is up and running, Sip Trunk is setup through Sipstation.com in the Sipstation module. All IP phones on the internal network connect and function properly (Yealink T26 phones).
I need help with the following issues:
- Setting up the IVR (Digital Receptionist)
- Getting the lines to hunt correctly when one line is busy.
- Phones connecting outside the office can establish a connection with the Server, but when making a call (no voice is transferred, you cannot hear them, they cannot hear you, could be a port forwarding issue)