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VoIP/(802.11g) ADSL2+ (VPN) Firewall Router
Chapter 4: Configuration
110
Confirm Password:
Re-enter the password for confirmation.
Display Name:
This parameter will be appeared on the Caller ID.
General Settings
This section reflects and contains basic settings for the VoIP module from selected provider in the Wizard
section. Fail to provide correct information will halt making calls out to the Internet.
SIP Device Parameters
SIP:
To use VoIP SIP as VoIP call signaling protocol. Default is set to
Disable.
Silence Suppression (VAD):
Voice Activation Detection (VAD) prevents transmitting the nature silence
to consume the bandwidth. It is also known as Silence Suppression which is a software application that
ensures the bandwidth is reserved only when voice activity is activated.
Default is set to
Enable.
Echo Cancellation:
G.168 echo canceller is an ITU-T standard.
It is used for isolating the echo while
you are on the phone. This helps you not to hear much of your own voice reflecting on the phone while
you talk. Default is set to
Enable.
RTP Port:
Provide the based value from the media (RTP) ports that are assigned for various endpoints
and the different call sessions that may exist within an end-point. (Range from 5100 to 65535, default
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value is 5100)
Region:
This selection is a drop-down box, which allows user to select the country for which the VoIP
device must work. When a country is selected, the country parameters are automatically loaded.
Voice QoS,
: Differentiated Services Code Point (DSCP), it is the first 6 bits in the ToS byte. DSCP
Marking allows users to assign specific application traffic to be executed in priority by the next Router
based on the DSCP value.
See Table 4. The DSCP Mapping Table:
Note:
To be sure the router(s) in the backbones network have the capability in executing and checking the DSCP
through-out the QoS network.
Setting for Phone Port 1
Registrar Address(or Hostname):
Indicate the VoIP SIP registrar IP address.
Registrar Port:
Specify the port of the VoIP SIP registrar on which it will listen for register requests from
VoIP device.
Expire:
Expire time for the registration message sending.
User Domain/Realm:
Set different domain name for the VoIP SIP proxy server.
Outbound Proxy Address:
Indicate the VoIP SIP outbound proxy server IP address. This parameter is
very useful when VoIP device is behind a NAT.
Outbound Proxy Port:
Specify the port of the VoIP SIP outbound proxy on which it will listen for
messages.
Setting for Phone Port 2
Please refer to descriptions in “Setting for Phone Port 1”.
How to register to SIP Server
1)
In Wizard Section, select your VoIP Service Provider and input information in the filed of
Phone
Number, Authentication Username
and
Authentication Password.
2)
In Wizard Section, click Apply
button to apply the settings.
3)
In General Settings, make sure general VoIP SIP information are correctly inserted.
4)
In General Settings, click Apply
button to apply the settings.
5)
In General Settings, click Synch Now
button to register the account(s) with your VoIP server.
Advanced – Parameters
VoIP through IP Interface:
IP Interface decides where to send/receive the voip traffic; it includes: ipwan
and iplan.
Easy way to select the interface is to check the location of the SIP server.
If it locates some
where in the Internet then select
ipwan.
If the VoIP SIP server is on the local Network then select
iplan.
Voice Frame Size:
Frame size is available from 10ms to 60ms.
Frame size meaning how many
milliseconds the Voice packets will be queued and sent out.
It is ideal to have the same frame size in
both of Caller and Receiver.
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PSTN Auto-fallback:
Whenever VoIP SIP responses error and error code matching with the codes in the
Edit
section, the VoiP calls will automatically fallback to PSTN.
In the other word, the call will be called
via the PSTN when VoIP SIP returns an error code.
Click the
Edit
to add or remove the responses code.
To be sure the code is separated by a comma
(,).
For
more
information
about
SIP
responses
codes,
please
check
to
link
to
http://voip-info.org/wiki/view/sip+response+codes
where you can get to know the meaning of each error
code.
Advanced – PSTN Environment Adjustment
PSTN Environment Adjustment options will help you to adjust the onhook and offhook voltage detection
values for your environment.
You should use these if the default values are incorrect and result in PSTN
calls not being detected properly, e.g. calls being terminated within 5 seconds of being answered. The
actual levels are determined by your environment including the number and type of telephones used.
Note:
ONHOOK means hung up.
To take your phone OFFHOOK, lift the receiver then press Hook/Flash until you hear your normal PSTN
dialtone, not your VoIP dialtone. Wait several seconds and then press Check Level.
You should check the OFFHOOK value for each telephone you have connected to this device. Set the
OFFHOOK voltage to the lowest setting registered for all your telephones, e.g. if your telephones return
values of 4, 5 and 7 then you should set your OFFHOOK voltage to 4.
Note:
The detected values will not automatically be set by the Check Level function; you must enter the lowest level
detected after testing all your telephones.
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Phone Port
This section displays status and allows you to edit the account information of your Phones.
Click
Edit
to
update your phone information.
Login Account Configuration
Phone Number:
This parameter holds the registration ID of the user within the VoIP SIP registrar.
Authentication Username:
Same as Phone Number.
Authentication Password:
This parameter holds the password used for authentication within
VoIP SIP registrar.
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Confirm Password:
Re-enter the password for confirmation.
Display Name:
This parameter will be appeared on the Caller ID.
Codec Preference
Codec is known as Coder-Decoder used for data signal conversion.
Set the priority of voice
compression; Priority 1 owns the top priority.
G.729:
It is used to encoder and decoder voice information into a single packet which reduces the
bandwidth consumption.
8kbps bandwidth is needed.
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 14-bit linear sample.
64kbps bandwidth is needed.
G.711A-LAW:
It is a basic non-compressed encoder and decoder technique.
μ
-LAW uses pulse code
modulation (PCM) encoder and decoder to convert 13-bit linear sample.
64kbps bandwidth is needed.
Non
-
used:
Only available in Priority 2 and 3.
It is selected if codec is not to be used.
Note:
Codec priority is assigned in the order as G.729 > G.711
μ
-LAW > G.711A-LAW
.
Speed Dial
Speed Dial comes handy to store frequently used telephone numbers which you can press number from
0 to 9 and the pound sign (#) on the phone keypad to activate the function. For example, speed dial to
phone number lists on 9, just press keypad
9
then
#
.
Your router will automatically call out to number
listed on entry 9.
For examples:
If your friend Tim gives you a SIP URL as sip: [email protected] then you can fill in as
.
If your friend Felix gives you a SIP URL as sip: [email protected] then you can fill in as
.
If your friend Greg gives you an IP address "192.246.56.56" only, then you can fill in as “192.246.56.56”.
In case, some of users may use DDNS, you can fill in with domain name as well.
Volume Control
Volume control helps you to adjust the voice quality of telephone to the best comfortable listening level.
Press “
-
“, the minus sign, to reduce either microphone or/both speaker’s level of your telephone.
Press “
+
“, the plus sign, to increase either microphone or/both speaker’s level of your telephone.

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