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NAT Keep Alive:
Disabled by default. User can enable it if 7800VDP(O)X is placed behind a NAT
router to ping SIP server every 60seconds (can be changed base on need) to verify the SIP server is
working.
Registration Expire Timeout:
This sets time interval before timeout.
Registration Retry Interval:
The interval set to retry sending registration message.
SIP Transport Protocol:
The protocol adopted to transport SIP, UDP commonly used.
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SIP Account
SIP account is an independent section for SIP account settings, including Extension number, etc.
Click
Add
or
Edit
to add new account or modify the existing sip account.
Account Name:
User-defined account name.
Account Enabled:
Enable to activate the sip account.
Default Dial Plan Chosen (Port 1):
Enable to allow user to set the account as the default VoIP rules
for port 1.
Default Dial Plan Chosen (Port 2):
Enable to allow user to set the account as the default VoIP rules
for port 2.
Incoming Phone Port:
Select which phone port you are setting.
Extension:
The Phone number.
Display Name:
Enter a display name to identify the phone, like indicating the phone usage.
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User Name:
The user name user registers in the sip server.
Password:
The password user registers in the sip server.
Authentication ID:
It is an authentication code required for some ISP, and can be left empty if not
required.
Answering Machine:
Enable to activate the answering machine feature so that user can record and
listen to the messages of this phone.
Send Message Via E-mail:
Enable to send message left by callers via e-mail to the user.
DTMF Method:
DTMF stands for "Dual-Tone Multi-Frequency", and is a telecommunication signaling
over analog telephone lines widely used between telephone handsets and other communication
devices and the switching center. “DTMG method” provides ways to transmit DTMF for VoIP, such as
RFC 2833, SIP Info, SIP Info (short), Inband and Auto, and RFC2833 is the widely used one.
Preferred codec#1,2,3,4,5:
Codec is known as Coder-Decoder used for data signal conversion. Set
the priority of voice
compression; Priority 1 owns the top priority.
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 13-bit linear sample into 8-bit
value.
G.
729a: It is used to encoder and decoder voice information into a single packet which
reduces the bandwidth consumption.
G.726_32:
It is an ITU-T ADPCM speech codec standard covering the transmission of voice
at rates of 32kbit/s.
G.722:
G.722 is an ITU standard codec that provides 7 kHz wideband audio at data rates from
48, 56 and 64 kbit/s. G.722 sample audio data at a rate of 16 kHz (using 14 bits), double that
of traditional telephony interfaces, which results in superior audio quality and clarity.
G.711Mu-Law:
It is a basic non-compressed encoder and decoder technique.
μ
-LAW uses
pulse code modulation (PCM) encoder and decoder to convert 14-bit linear sample.
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204
Call Forward
Call features is designed to allow all calls or specific calls redirected to some specified call number
when under some special situation or contingency occurs.
Click
Edit
to change the Call Features.
Call Forwarding:
All calls forward to:
Set the call number to receive all the incoming calls unconditionally.
Busy calls forward to:
Set the call number to receive only busy calls.
No Answer calls forward to:
Set the call number to receive calls that are not answered.
Click
Apply
to save your settings.
(Busy calls forwarded to 0203333)
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Call Through
With the “Call Through” function, you can configure your 7800VDP(O)X so that certain calls are
forwarded to any destination number using a cheaper telephone connection, for example via landline
or mobile network, this can save costs.
Example: You are on the road and would like to use your mobile telephone to call somebody abroad.
You can either call that person directly, or you can call your 7800VDP(O)X at home and let the
7800VDP(O)X forward to the extension abroad via the Internet or less-expensive landline connection
at a much less expensive rate.
Note:
Two call logs can be generated when call through feature is activated in
Missed Call Log,
and
Outgoing Call Log
.
Parameters
Incoming Number:
Select a number (SIP account or PSTN) for which you wish to enable the Call
Through feature.
Number for Outgoing Calls:
Select a number (SIP account or PSTN) to be used to forward the call
to the destination.
PIN:
A four-digit password for forwarding calls. Default is 1234. It has to be input before user input
the destination call number.
Greeting Delay:
Greeting delay specifies how long the call through feature will be activated after
caller calls the incoming number.

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