Billion 810VGTX Router
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*67 Anonymous Call: Hide your phone number from being displayed at the remote terminal. It is only applied to the
next call when you enter this control character. The detailed operation procedure is “Off Hook -> *67 -> On Hook ->
Off Hook -> Dial”. This feature is disabled by default.
Phone #: This is the fast dial which you can dial out a phone number immediately without waiting.
Note: Refer to the Special Dial Code section in this Manual for more details.
Codec Preference
Codec is known as Coder-Decoder, it is used for data signal conversion. Setting the priority of voice compression with
Priority 1 represents the top priority.
G.729: It is used to encode and decode voice information into a single packet to reduce bandwidth consumption.
G.711
µ
-LAW: It is a basic non-compressed encoder and decoder technique.
µ
-LAW uses pulse code modulation
(PCM) encoder and decoder to convert a 14-bit linear sample.
G.711A-LAW: It is a basic non-compressed encoder and decoder technique. A-LAW uses pulse code modulation
(PCM) encoder and decoder to convert a 13-bit linear sample.
G.726-32: It is used to encode and decode voice information into a single packet to reduce bandwidth consumption.
Currently only supports bit rates of 32Kbps.
DTMF Method: The Inband, RFC 2833 and SIP INFO (RFC 2976) are supported.
Volume Control
Volume control enables you to adjust the voice quality of the telephones to a comfortable level. Press the “-” minus
sign to reduce either the microphone or speaker level of your telephone. Press the “+”, plus sign to increase either
the microphone or speaker level of your telephone.
PSTN Dial Plan (Router with LINE port only)
This section enables you to configure the “VoIP with PSTN switching” on your system. You can define a range of dial
plans that will make regular call switch from VoIP to the PSTN line. Prefix numbers are an essential key to make a
difference between VoIP and Regular phone calls. If the actual numbers dialed match with the prefix number defined
in this dial plan, the dialed number will be rerouted to the PSTN to make a regular call. Otherwise, the number will be
rerouted to the VoIP networks.
Note: In order to utilize this feature, you must have registered and connected to your SIP Server first.
