CHAPTER 2 |
34
INSTALLATION AND CONFIGURATION
The SIP
EXTENSION
is like an extension number in a PBX system, with the difference that it is generally not required to be a
number, though most often it is. It maps incoming calls to the line positions on the controllers. Together with
SERVER
it makes
the SIP URI (address) identifying the line. The server is also used to form SIP addresses from numbers for outgoing calls.
You can assign the same extension to multiple buttons as we have done here. Calls will ring in on the first available button,
acting like a hunt group.
Address Book
There is yet another Address Book menu for each show, just in case. This one works exactly the same as the address book in
the studio profile. The VSets will display the sum of all the entries across the studio and show it’s logged into. This is so that
you can store commonly used station business numbers in the studio profile and show dependent “expert callers” in the show
profile. We told you the VX was flexible!
Using the VX to replace ‘Couplers’
If your station uses “Pre-delay IFB (Interruptible Fold-Back) Couplers or “listen lines”, the VX can
replace an entire wall of hardware, and by using DID (Direct Inward Dial) numbers instead of POTS
lines, you can also save a lot of money each month by replacing those expensive analog lines.
Create dummy studios for each coupler bank and shows with the phone number or extension you
want to use and tick the auto-answer box next to it. Choose the audio feed from among the Livewire
sources.
Full details are in the “Using VX for couplers” and “Creating ‘Dummy’ Studios” for call screening and
couplers App-note toward the back of this manual.
Call Audio Processing Page
The VX has dynamics processing on both the send (from studio, to caller) and receive (caller) audio directions.
The send (to the caller) audio processing is fixed and consists of a protection limiter and some EQ. The purpose of the limiter
is to protect the Telco line from the distortion that would result from clipping due to an audio overload. Livewire studio audio
has a much larger dynamic range than telephone lines. We use a limiter rather than an AGC because we don’t want the system
to increase low-level signals, which could cause feedback and sound unnatural to the caller. These features do not have any
controls to show in the menu set, we’d just thought you’d like to know what’s happening under the hood.
The receive processing includes ducking level, an AGC,
noise gate, and dynamic EQ.
The term “hybrid” is used only for convenience here,
since VOIP is inherently four-wire (separate transmit
and receive). The only echo possible is each from the far
end which is either acoustic or reflected at a four-wire to
two-wire (POTS) hybrid, found in Telco line equipment
connected to bad (usually very long) subscriber loops.
Receive AGC.
Automatic Gain Correction (AGC) control
is available for the caller’s audio. The purpose of the AGC
is to ensure that each caller sounds similar to the last,
no matter what kind of call it is (Cellular, VoIP, Legacy Connections, etc.) Another plus to having AGC present is it adds
to the production value since no “fader riding” is necessary. The AGC does it for you. We recommend setting this to “16”
(maximum)
