Need to setup Asterisk, the server is up and running, Sip Trunk is setup through [url removed, login to view] 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)