ReadyGate
V300
Web Administor Interface
User’s Manual
Version: 2.03
Date Issued by 2013/03/22
1
目錄
1.
Web .............................................................................................................................................4
1.1 Login Web UI...................................................................................................................4
1.2 Web UI function list....................................................................................................5
2. VoIP Function ..........................................................................................................................8
2.1 Portall Status ................................................................................................................8
2.2 Provision .......................................................................................................................10
2.3 Line Configure ............................................................................................................ 11
2.3.1 Line Setting ..................................................................................................... 11
2.3.2 Tone Setting....................................................................................................13
2.3.3 Line Feature ....................................................................................................15
2.3.4 Line Diagnostics ............................................................................................18
2.3.5 Line Impedance .............................................................................................19
2.3.6 Message Indicator ........................................................................................20
2.4 Routing Setup.............................................................................................................21
2.4.1 VoIP Call Out...................................................................................................21
2.4.2 VoIP Call In......................................................................................................31
2.4.3 VoIP Call In IVR .............................................................................................38
2.4.4 Routing Profile................................................................................................41
2.4.5 Forwarding.......................................................................................................44
2.4.6 Authorization ..................................................................................................47
2.5
Register Server ..........................................................................................................48
2.5.1 Register Status ..............................................................................................48
2.5.2 Register Server—SIP Protocol..................................................................50
2.5.3 Register Server—H.323 Protocol ............................................................52
2.6 Advance Setup ...........................................................................................................54
2.6.1 NAT Traversal..................................................................................................54
2.6.2 Listen Port........................................................................................................55
2.6.3 VoIP Package ..................................................................................................56
2.6.4 RTP Packet Summary ..................................................................................58
2.6.5 Flash & Call waiting......................................................................................59
2.6.6 Gain ....................................................................................................................60
2.6.7 QoS.....................................................................................................................62
2.6.8 CDR ....................................................................................................................63
2.6.9 FoIP ....................................................................................................................64
2.6.10 Prompt Voice & Beep ................................................................................65
2.6.11 Call log............................................................................................................67
2.7 Application ...................................................................................................................68
2.7.1 Ping Test ...........................................................................................................68
2
2.7.2 Centrex .............................................................................................................69
2.7.3 Telnet & SNMP................................................................................................70
2.7.4 Call Timer.........................................................................................................72
3. System Setup........................................................................................................................73
3.1 System ..........................................................................................................................73
3.1.1 System Status................................................................................................73
3.1.2 System Settings ............................................................................................78
3.1.3 Date&Time.......................................................................................................79
3.1.4 Administrator Settings ................................................................................80
3.2 VDSL ..............................................................................................................................81
3.2.1 VDSL Settings ................................................................................................81
3.2.2 VDSL Status ....................................................................................................82
3.3 WAN ...............................................................................................................................83
3.3.1 WAN Settings..................................................................................................83
3.3.2 DNS ....................................................................................................................87
3.4 LAN .................................................................................................................................88
3.4.1 LAN Settings ...................................................................................................88
3.4.2 DHCP Client List.............................................................................................90
3.5 Wireless ........................................................................................................................91
3.5.1 Basic Settings.................................................................................................91
3.5.2 Advance Settings ..........................................................................................92
3.5.3 Security.............................................................................................................93
3.5.4 Access Control................................................................................................94
3.5.5 Site Survey......................................................................................................95
3.6 NAT .................................................................................................................................96
3.6.1 Port Mapping...................................................................................................96
3.6.2 ALG .....................................................................................................................97
3.7 Firewall..........................................................................................................................98
3.7.1 Denial-of-Service ..........................................................................................98
3.8 Routing..........................................................................................................................99
3.8.1 Routing Table ..................................................................................................99
3.8.2 Static Routing...............................................................................................100
3.9 Bandwidth&VLAN ....................................................................................................101
3.9.1 Bandwidth Control ......................................................................................101
3.9.2 VLAN ................................................................................................................102
3.10 Backup / Restore ..................................................................................................103
3.10.1 Configurations ...........................................................................................103
3.10.2 VoIP module ...............................................................................................104
3.11 Reboot.......................................................................................................................105
3.12 Save Modification .................................................................................................106
3
1. Web
1.1 Login Web UI
Welcome to buy and use ALLWIN Tech's VoIP Router, this manual will help you
understand the operation using this device (hereinafter referred to as AwG) of
the WEB management interface.
You can manage your VoIP Router by AwG built-in Web management interface.
Please prepare a computer connected to the LAN side of AwG. As AwG the default
DHCP server service is ON, so keep your computer's TCP / IP settings to "Obtain
an IP address automatically" in order to obtain the right from AwG DHCP IP.
AwG default on the network which will become the Gateway, the default IP is
192.168.22.1, at the same time, it will be assigned to computers connected to
the LAN side IP address of a 192.168.22.x.
To set your computer's TCP / IP, by following the path set (in Windows XP for
example):
Start → Control Panel → Network Connections → Local Area Connection
content
→ Internet Protorcol (TCP / IP) → content → click Obtain an IP address
automatically
To access the management interface, on the computer, open IE browser in the
address bar enter: http://192.168.22.1/, as shown below:
Then the screen will first ask you to enter an administrator account password, the
default account is voip, the password is 1234. Enter the correct password to
enter the account management interface.
4
1.2
Web UI function list
AwG provide user-friendly management interface allows you to manage and
configure your router and VoIP functionality. Web UI, there are two major key
management project: VoIP functions, System Setup details of the items listed
below:
z
VoIP Function
„
„
„
„
„
„
Portall Status
Provision
Line Configure
‹ Line Setting
‹ Tone Setting
‹ Line Feature
‹ Line Diagnostics
‹ Line Impedance
‹ Message Indicator
Routing Setup
‹ VoIP Call Out
‹ VoIP Call In
‹ VoIP Call In IVR
‹ Routing Profile
‹ Forwarding
‹ Authorization
Register Server
‹ Register Status
‹ Server #1
‹ Server #2
‹ Server #3
‹ Server #4
Advance Setup
‹ NAT Traversal
‹ Listen Port
‹ VoIP Package
‹ RTP Summary
‹ Flash & Call waiting
‹ Gain
‹ QoS
‹ CDR
‹ FoIP
5
„
z
‹ Prompt Voice & Beep
‹ Call Log
Application
‹ Ping Test
‹ Centrex
‹ Telnet & SNMP
‹ Call Timer
System Setup
„
„
„
„
„
„
„
„
„
System
‹ System Status
‹ System Setting
‹ Date & Time
‹ Administrator Settings
VDSL
‹ VDSL Setting
‹ VDSL Status
WAN
‹ WAN Settings
‹ DNS
LAN
‹ LAN Setting
‹ DHCP Client List
Wireless
‹ Base Settings
‹ Advance Settings
‹ Security
‹ Access Control
‹ Site Survey
NAT
‹ Port Mapping
‹ ALG
Firewall
‹ Denial-of-Service
Routing
‹ Routing Table
‹ Static Routing
Bandwidth & VLAN
6
„
„
‹ Bandwidth Control
‹ VLAN
Backup/Restore
‹ Configurations
‹ VoIP module
Reboot
Language/語言
„
Save Modification
„
7
2. VoIP Function
This section describes the VoIP Setup functions of device, the location of the
menu items will be list to represent the range slash.
For example, / Line Configure / Line Setting / that is located in the Line
setting below Line Configure menu item..
2.1
Portall Status
2.1.1 / Portall Status /
Port Status: Displays the current call status or a last call result.
Description:
a. PC Time: displays the date and time of connected computer.
b. Gateway Time: displays the current date and time in the device
through the NTP Server on the network or manual set.
You can set time on / System / Date & Time / menu item.
A. Port Message
c.
d.
e.
f.
Port: display line number.
Type: Line interface type. FXS: connected to a telephone set
PBX Co. line.
Display Name: VoIP call user display name.
Status: current line status display.
Idle: non-use.
Signal: Waiting for dial-up or VoIP call connection in progress.
In: VoIP In-bound call in progress.
Out: VoIP Out-bound call in progres.
8
or
g.
z
z
z
z
h.
i.
j.
k.
l.
z
z
Connected IP: display connedted remote side type for this call.
PstnOut: Outbound call to analog line interface.
PstnIn: Inbound call from analog line interface.
rs: call via register server.
IP: direct VoIP call by IP.
Caller ID: Caller ID.
Start Time: start time of VoIP call.
End Time: End time of VoIP call..
Talking Sec: Total VoIP talking time in seconds.
Dialed number:
dial out numbers for out bound call.
received dialed numbers from In bound call.
B. Error Message
Display the last error message of failure call.
9
2.2
Provision
2.2.1 /Provision/
To use the auto provision function, the system have to install a dedicate Auto
Provision Server for keep all parameters for installed gateways. When Enable the
Auto Provision function, the System administer can modify all the Parameters of
each gateway on the local Provision Server, and remote gateway will automatic
download all the parameters from Provision Server.
The Gateways can link up to five provision servers simultaneously for Redundancy
backup the system.
10
2.3
2.3.1
Line Configure
Line Setting
2.3.1 /Line Configure/Line Setting/
In this configuration page, you can set the name of each line, line number,
voice volume, and physical port-related functions.
a.
Port: display line numbers, such as the first line or second line of the
state, and so on.
b.
Interface: Line interface type. FXS : connected to analog telephone
set or PBX co. lines.
c.
Name: definable the line name, this name will display on the other
side device during VoIP call.
Line Number: Define line extension number, can be given to each line
as the extension number.
d.
e.
TxGain: Transmitter gain, adjustable playback on Local phone
(handset) volume adjustment, increase the dB value of the local-end
phones will increase the playback volume.
f.
RxGain: receiver gain, adjustable Local phone microphone volume.
Increase the receiver volume will amplifie microphone volume to
transmit to the other end of the call.
g.
Inbound: Enable or Disable Inbound (VoIP Call Out) function. default
is Enable.
h.
Outbound: Enable or Disable Outbound (VoIP Call In) function. default
is Enable.
11
i.
Hotline: Enable/Disable Hotline function. When Enable Hotline
function, user do not need to dial number to make a VoIP call after
seize the line (FXS: off hook phone).
For example, If we want the first line to hotline, each time user seizes the
first line (FXS: off hook phone), without dialing any number, will
automatically make a VoIP call to gateway local on 168.56.109.22, and
dial 600 automatically as extension. Then we should enable first line’s
hotline function, and on / Routing Setup / VoIP Call Out / added a dial rule.
In the Area Code field to specify the first line input hl1 as Hot Line1, and
remember to Strip field, enter 3 to mask out "hl1". In the Prefix field, enter
the phone number you want to dial "600" See below:
Index Remark
1
Hot_Line
Area
Min
Max
Code
Digits
Digits
Destination Strip Prefix Profile Delete
hl1
10.1.1.1
12
3
600
Delete
2.3.2
Tone Setting
2.3.2/Line Configure/ Tone Setting
A. Call Progress Tone
Here to define the call progress tone for generate/detection. After modify, please
click / Save Modification /to re-start to take it effective.
„
Detect Voice Busy Cycle: When detects a match of a busy tone, and the number of
cycles up to the value of this setting, the device will be determined to confirm the
receipt of a busy tone, that will pass this dropped calls.
B. define Call progress tone
Here you can set up 15 items of audio specification for tones generate and detection,
such as dial tone, busy tone, ring tones, etc. Generally call progress tones are between
300 Hz to 2000 Hz.User can set up multiple groups of tone for detection, but only one
group will be used as generation.
a.
b.
c.
d.
Tone: Tone item index, Maximum 15 items.
Type:
„
Dial: dial tone, tone generated to wait user dial.
„
Busy: busy tone, generate/detection for line busy.
„
Ring: ringback tone, Generate when waiting for answer.
Low freq: low frequency setting, set the lower frequency of tone.
High freq: high-frequency setting,set the higher frequency of tone..
Each tone can include two frequencis if only one frequency need, set the
13
e.
High Freq. to zero.
T_ON_1, T_OFF_1, T_ON_2, T_OFF_2:
„
tone cadence interval time : includes dual-band beat interval to four
intervals (see below), the lowest is 30 milliseconds. (Unit in mS)
14
2.3.3
Line Feature
2.3.3.1
/Line Configure/ Line Feature
On this page, set the telephone line interface related parameters
A. Dial Pause signal length [100 ~ 3000] ms:
Defines the pause time (milliseconds, ms) defined for “,” comma symbol
used on Prefix field of / Routing Setting / VoIP Call Out / or / VoIP Call
In / . By default, enter a comma "," will cause a pause time 1000 ms
between DTMF digit, time can be set at least 100 ms, the maximum is 3000
ms. Users can use multiple consecutive “,” to exend the dial pause time.
B. Loop Current Drop & Polarity Reversal Generate
If the remote party disconnect the VoIP call, Local FXS interface can
enable/disable the following options:
„ Disable: Disable FXS interface to generate both line polarity reversal
signal and current interrupt signal function, just send busy tone.
Polaryti Reversal-> Enable FXS interface to generate line polarity
reversal signal.
„ Current Drop-> 1 S: Enable FXS interface to generate one second
length current interrupt signal.
„ Current Drop-> 2 S: Enable FXS interface to generate two seconds
length current interrupt signal.
„ Current Drop-> 3 S: Enable FXS interface to generate three seconds
length current interrupt signal.
„
15
C. When using FXS to answer, decide to bring out the phone number by
setting the following options:
„ Drop out: Do not send, in order to avoid the phone to hear the
unnecessary DTMF tones after answer the call..
„ Resned: send the DTMF number for PBX to transfer the call.
D. Method of FXS interface to generate CID (Caller ID ),the following
options:
„ Disable: turn off, do not send CID to phone
„ DTMF: send DTMF CID signal to phone
„ FSK Bell: send FSK Bell singal to phone
„ FSK ETSI: send FSK ETSI to phone
E. Call waiting ID Generate type:
„
Disable: turn off, do not enable the Call Waiting Caller ID
FSK Bell: generate Caller ID signals use FSK Bell
„ FSK ETSI: generate Caller ID signals use FSK ETSI
„
F.
When VoIP call out,Send ANI by
Register Number: send the number of registered
„ Line Number: send the setting Line number.
„ PSTN CID: Only on the FXO interface, send the received number from
the CID.
„
G. Define how the FXS interface to ring the phone line when VoIP call
in:
„ Free Random: Any unused available line.
st
„ Line number Priority: The 1 line has high priority; it will always ring
the 1st line if it is available. When 1st line is busy, it will try to ring 2nd
line if it is free.
st
nd
„ Rotation: 1 line ring first, then 2 line ring next time, when the latest
line ring this time, it will come back to ring 1st line next time.
All: Ring all phone lines if it is available.
„ Sequence: Ring all the available phone line one by one, the ring period
for ring each phone is definable.
„ Period (sec.): define the ring period (seconds) when select
“Sequence” ring.
„
16
2.3.3.2
H.
/Line Configure/ Line Feature
FXO Wait dial Tone before dial? FXO (FXO only):
When enable, FXO line will wait for dial tone before dail the number.
I.
FXO Line Hunting Method,FXO /VoIP Call In/ (FXO only)
Define how the FXO interface to ring the phone line when VoIP call in:
„ Free Random: Any unused available line.
st
„ Line number Priority: The 1 line has high priority; it will always ring
the 1st line if it is available. When 1st line is busy, it will try to ring 2nd
line if it is free.
st
nd
„ Rotation: 1 line ring first, then 2 line ring next time, when the latest
line ring this time, it will come back to ring 1st line next time.
J.
Ring No Rehook:
When enable, FXO interface will not terminal the incoming call from PSTN by
OFF-HOOK then ON-HOOK.
17
2.3.4
Line Diagnostics
V300 Series device supports line diagnostics as GR-909 specification.
2.3.4.1/Line Configure/ Line Diagnostics
Provide a 5 test items: Hazardous Potential, Foreign Voltage, Resistive Faults,
Offhook, REN
It will drop the call during the line testing.
Click [Detail] to view the detail test result.
18
2.3.5
Line Impedance
Configure the line impedance of FXS/FXO interface.
2.3.5 /Line Configure/ Line Impedance
19
2.3.6
Message Indicator
Define the method to notify the phone on FXS interface when receive a Voice
mail message notify from VoIP server.
2.3.6 /Line Configure/ Message Indicator
a.
Resend Data Time:Time interval between resend indicator message.
.
b.
voicemail notification message mode:
„
„
„
„
Disable :do not send the notification message.
LAMP: increase FXS interface voltage lighting message
notification lamp on phone.
DTMF: send specific DTMF code to on/off the message
notification on the phone, for the number specific to this function,
please contact the phone manufacturer.
FSK: send specific FSK singnal to on/off the message notification
on the phone, for the number specific to this function, please
contact the phone manufacturer.
20
2.4
Routing Setup
2.4.1 VoIP Call Out
2.4.1 /Routing Setup/VoIP Call Out
This page let you define the routing rule for Call out to VoIP. (User press the phone
number through phone set dial pad, then VoIP Router translate the phone number
by the routing table setting here to destination IP & dial out number then Call out
via network protocol).Here can define some special keyword like IPIVR, PSTN as
destination for some special function also.
Each time when you off hook the phone connected to this VoIP Router, you will hear
a dial tone or prompt voice to remind you to press the phone number, after you
input the number you called, if digits of the number of you called is not exceed the
Max Digits, please remember to press the # key for ending the input, if you do not
press # key for enter, gateway will automatically call out the number after timeout
of define on OtherDigitTime.
A. Time & Digits wait for dial out
The VoIP Router wait user input the number digits & time parameters as below:
Time & Digits wait for user Press.
a. MaxDigits: Define the maximum digits wait for user press for all VoIP Call Out,
if user press digits match the number defined here. It will go to translate for call
out rule without needed to press # key.
b. FirstDigitTime: Define the waiting time (seconds) for user press phone
number first digit. User need to press first digits before the setting time
21
(seconds) defined here, if VoIP Router wait for the defined seconds and there is
no any digits press, the VoIP Router will stop to wait and feedback the user busy
tone.
c.
OtherDigitTime: Define the waiting time (seconds) for user press phone
number secondary & the rest digits. User need to press the rest digits before the
seconds defined here, if VoIP Router wait for the defined seconds and there is no
any digits press, it will go to translate for call out rule without needed to press
# key.
d. Timeout for Re-entry route: When one of the rules on the VoIP call out rules
is matched and be execute, the device will wait the time( seconds) defined here
for successful connection, but if time out defined there still failure connection, it
will trying to reroute by another call rule setting by the “v”+ the number prefix.
For example as below, when the user try to call the destination number
12345678, it will try to call the gateway location at 168.11.22.33, but if wait 10
seconds and still can not successful connection, the gateway will abort the call
and try call out 10.10.10.10.
Timeout for Re-entry route:
Index Remark
1
2
Area Min
Max
Destination
Code Digits Digits
10
second.
Strip Prefix Profile Delete
Normal
8
8
8
168.11.22.33
Delete
rule
Backup
v8
10.10.10.10
Delete
rule
<The example that use “v” prefixes for reroute the call out>
When
user
enable
the
hot
line
function
on
/VoIP
Setup/Line
Configure/Line Setting/ menu, it will over ride the above parameters and
direct call out by hot line call out rule.
B.VoIP call out Routing Table
a.
Remark:
Remark for this routing rule. Please use UNDERLINE to replace the
SPACE due to HTTP protocol limitation.
b.
Area Code: Define the Prefix number fit this rule, any phone number prefix
digits matched with the rule will call out by this rule define. Please Notify there
is a compare order rule on this routing table. That mean the VoIP Router will
22
check the rule list from top to bottom one by one, any rule item matched with
the prefix digits that user press will go to call out directly no regard to the rest
rules below. For Example, if a rule item for area code 8862 is on Index 5,
another rule item for area code 886 on Index 6 below that will be ignored.
By setting the hln (hl1 for hot line one, hl2 for hot line two) on the area code
field and enable hot line function (/VoIP Setup/Line Configure/Line
Setting/), the VoIP Router can service the hot line direct call.
i.
Min Digits: define the minimum digits wait for user press for number
fit this rule, if user press digits less the number defined here. It will
keep waiting for input until exceed the FirstDigitTime defined time. If
user press digits more then Min Digits here, the VoIP Router will wait
time defined on OtherDigitTime then go to translate for call out rule
without needed to press # key.
ii.
Max Digits: define the maximum digits wait for user press for number
fit this rule, if user press digits match the number defined here. It will
go to translate for call out rule without needed to press # key.
iii.
Destination: Define the destination IP for call out number fit this rule,
user can input below format:
„
IP address, for example: 168.56.9.22
1.
for sip Æ please add sip: before ip address, for example
2.
sip:168.56.9.22
for h323 Æ please add h323: before ip address , for example
h323:168.56.9.22
„
URL, route via URL. For example: sip.fwd.com .This VoIP Router
can
setup
to
register
to
DDNS
service
(/System
Setup/Advanced/Dynamic DNS/) to let user call out to another
VoIP Router with dynamic IP by URL.
„
gkn : route via gatekeeper, it will get the destination IP by
gatekeeper setting (/VoIP Setup/Gatekeeper/) in advance. For
example: gk1 for gatekeeper 1. gk2 for gatekeeper 2. gk for all the
gatekeepers available ( search sequence: gk1 > gk2 > gk3 > gk4).
Gk3_2_1 will try gk3 first, then gk2, then gk1.
All the setting above can be added by port number, for examples:
168.56.9.22:8495 will call to 8495 port.
23
„
rsn: same as gkn
server.
, basically, it is used for SIP register
„
ipivr: Enter the Network parameter voice interactive setting
mode. User can use this function to enter all the WAN network
parameters without PC. ( Please refer the application note
“ IP IVR produce “ for more detail procedure ).
„
ldcfg: Restore all parameters to the default values. User can
assign a password to use this function to restore all the
parameters to the default values.
„
rect: Enter to voice record procedure . User can assign a
function code for enter the voice record procedure, when
press this code to enter the voice record procedure, the
device will record 30 seconds voice file and keep on sound
wave file ( G.711, uLaw), User can download the recorded
wave file on /VoIP Setup/Advance setting/Prompt Voice/
and used this file to upload for customization voice file or
used for busy tone analysis.
f.
„
agent: agent code setting. When a VoIP call in made by this
device, it will ring the assigned phone set. If the user want to
use the different phone set (connected to same device, but
did not ring) to answer the call, just off hook and enter this
agent code to redirect the call to this phone you used for talk.
„
lo: assign the route to local loop back. The destination IP of
this call will be the local host, i.e.:127.0.0.1
Strip: the number of digits will be ignored by user input. For example,
if user press the number is 886212345678 and the STRIP field is
setting to 4, the first 4 digits 8862 will be truncated and actually call
out number will be 12345678.
g.
Prefix: The numbers will be added on the prefix of the user press
number. For examples, if user press the number is 12345678 and the
PREFIX field is setting to 0028862, the actually call out number will be
002886212345678. Another example, if user press the number is 90,
STRIP field is setting to 2, and the PREFIX field is setting to
24
0,12345678, the actually call out number will be 0,12345678 ( “,”
mean delay 1 second). This example is especially useful for speed dial
function.
h.
Profile: Define the optional special call out parameters on this
destination. Please input the name you defined on the profile (/VoIP
Setup/Routing Setup/Routing Profile/) list.
i.
Delete: Delete this rule item on routing table.
To add new rule item on routing table, please assign the item number you
want to insert before, input AREA CODE and IP address then press ADD
button to add it on the list. Then modify the necessary information on the
routing table list.
Please remember to press the modify button to take it effect. For store back
to
flash
memory,
please
press
/Syetem
Maintenance/Save
Modification/ .
C.Setting Examples
Here is some VoIP call out routing table setting examples below:
a.
Define wait time and digits for destination phone number
MaxDigits:
10
FirstDigitTime(Sec):
30
OtherDigitTime(Sec):
5
In this case, when user picks up the phone, the VoIP router will generate 30
seconds (defined on FirstDigitTime) dial tone for user press DTMF for
destination phone number, After user press first digit DTMF from phone set
(for example, 0, the VoIP router will wait 5 seconds (defined on
OtherDigitTime) to press the rest phone number digits, if user did not press
any key within first 30 seconds, the VoIP Router will generate the busy tone
to terminate the call. After user press first digit and did not key any key
within 5 seconds, for example, like 601 it will call out 601 after 5 seconds,
but if user press 601#, it will direct call out 601 immediately without waiting
rest key.
25
In this case, the Max Digits is setting to 10, so if user dial 0212345678, 10
digits phone number, it will call out immediately without wait 5 seconds or #
key, that mean it will not accept phone number more than 10 digits like
02123456781, if user press that phone number, it still call out the number
to 0212345678 because maximum digits for phone number is 10.
b.
VoIP call out by IP:
Index
Remark
1
NY_office
Area Code Min Digits
Max Digits
6
IP Address
Strip Prefix
Profile
Delete
172.16.7.1
Delete
In this case, we assume that we have another VoIP router locate at New
York office and the IP is 172.16.7.1 , when we press any phone number
prefix is 6 will call to that VoIP router, for example, if we dial 601, the VoIP
will Call out 601 to another VoIP router locate at IP 172.16.7.1, you can
check the real call out IP and phone number at the VoIP Setup/ Port Status:
Port Message
Display
Port Type
Status Connected IP
Caller ID
Start Time
Talking
Dialed
Sec
number
End Time
Release by
name
1
FXS
Idle
2
FXS
Idle
c.
Index
172.16.7.1
2004/02/19
2004/02/19
13:55:10
13:55:43
28
601
(146)onHangup
Call by Domain name:
Remark Area Code Min Digits
Max
Digits
IP Address
Strip
Prefix
4
013902440272
2
abc
8621
4
4
Voiprouter.dyndns.org
3
xyz
86
2
5
www.voipgateway.com
Profile
Delete
Delete
In this case, by route rule 2, we set up a short cut number 8021 for dial out
number 013902440272 to another VoIP router, user just press 8021 will
cause cut 4 digits (8621) define on Strip, and add the number defined on
Strip (013902440272), then call to that gateway(voiprouter.dyndns.org)
and number(013902440272).
26
In this case, by route rule 3, we assume we have another VoIP Router locate
at china.ezvon.com URL, and we use prefix 86 to call out for this gateway,
the minimum digits for phone is 2 digits and the maximum phone number
digits is 5, any phone number contain over 5 digits will be truncated to 5
digits like 862013 will be truncated to 86201 for call out.
z
Caution:
There is order rule on this routing table; the VoIP router will check the
route table items by index order one by one. That mean, in above case, if
user put the area code item 86(index 1) above 8621(index 2), then the
route item 8621 will never been used.
Area
Remark
1
Take_All
86
10.1.1.1
2
Never_Used
8621
20.1.1.1
Code
Min Digits
Max
Index
IP Address
Digits
Strip
Prefix
Profile
Delete
never used
d.
Strip and Prefix
User is easy to combine using Strip and Prefix define to modify the phone
number from phone to real call out phone number, for example, if the VoIP
router is installed on Taipei and use another Gatekeeper to service global
service. When user just dial 10 digits Taipei phone number like
02-12345678(do not need to press # key because Max Digits setting is 10),
and the VoIP router will stripe the 02 ( 2 digits defined on Strip) ,add the
country code 8862 (defined on Prefix) then send 8862-12345678 out for
VoIP call, see below example:
Index
1
Remark Area Code Min Digits Max Digits
Taipei
02
IP Address
Strip
Prefix
gk
2
8862
10
Profile
Delete
Delete
By above setting, When you dial 0212345678, you can check the real call
out IP and phone number will change to 886212345678 at the VoIP Setup/
Port Status:
27
Port Message
Display
Port Type
Talking
Status Connected IP Caller ID
Start Time
End Time
name
1
FXS
Idle
2
FXS
Idle
e.
Dialed number
Release by
Sec
172.16.7.1
2004/02/19 2004/02/19
13:55:10
13:55:43
28
886285123390 (146)onHangup
Call via Server /SIP Register Server
This VoIP router can register up to 4 servers, for example:
Area
Min
Max
Code
Digits
Digits
Index
Remark
IP Address
Strip
Prefix
Profile
Delete
1
Via_RS2
1
rs2
Delete
2
RS1_3
2
rs1_3
Delete
3
RS_ALL
3
rs
Delete
By Index 1, if user input any phone number with prefix code is 1, The VoIP
Router will call out via Gatekeeper 2.
By Index 2, if user input any phone number with prefix code is 2, The VoIP
Router will try to call out by Gatekeeper 2 ( if register to Gatekeeper 2 is
successful), if Gatekeeper 2 is not available, it will check Gatekeeper 3, then
check Gatekeeper 1.That mean if register to gk2 is failure and register to
gk3 & gk1 is successful, the VoIP router will call out via gk3.
You can check the Gatekeeper register status on /VoIP Setup/Register
Server/Register Status.
f.
Call to different IP port
The default IP port used by VoIP router is 1720 for H.323 and 5060 for SIP,
if work with remote side of VoIP Router or gateway is change another port
number for VoIP, please assign another port number after destination IP or
URL. Please make sure both side use same port number for VoIP call,
otherwise it can not make call. You can change the VoIP router default listen
port on .
28
Index
Remark
1
Port_1719
1
sip:10.1.1.1:1719
Delete
2
Port_8495
2
www.voipgateway.com:8495
Delete
g.
Area Code Min Digits Max Digits
IP Address
Strip
Prefix
Profile
Delete
Profile:
Define the optional special VoIP parameters when calling to the destination.
Please input the name you defined on the profile (/VoIP Setup/Routing
Setup/Routing Profile/) list.
Example: if user set the VoIP Call Out & Routing Profile like below:
Area
Index
Remark
1
UsePF1
1
2
UsePF2
3
UseDefaultPF
Index
Name
ID1
VAD
AS
Max
Profile
Delete
rs1
pf1
Delete
2
10.1.1.2
pf2
Delete
3
www.voipgateway.com
Code
Min Digits
CODEC
ID2
IP Address
Digits
Strip
Prefix
Delete
H.245
DTMF
T.38
Package
Tunneling
Relay
FAX Relay
Frame
AS
ID3
AS
ID4
Q.931
Fast
Start
AS
Delet
e
1
2
pf1
ON
G.723.1
ON
00001
H.323
1001
E.164
pf2
ON
G.723.1
ON
00002
H.323
1002
E.164
Out band
OFF
3
ON
Delete
In band
OFF
3
OFF
Delete
When VoIP call out number with prefix 1 will use the Profile named PF1
(H.323 ID1 = 0001, E.164 ID=1001, DTMF Relay=Out band, Q.931 Fast
Start=ON) to Call out VoIP.
When VoIP call out number with prefix 2 will use the Profile named PF2
(H.323 ID1 = 0002, E.164 ID=1002, DTMF Relay=In band, Q.931 Fast
Start=OFF) to Call out VoIP.
When VoIP call out number with prefix 3, because there is no Profile
assigned, it will use the default value for VoIP out.
29
h.
Delete:Delete this rule item on routing table.
To add new rule item on routing table, please assign the item number you
want to insert before, input AREA CODE then press ADD button to add it on
the list. Then modify the necessary information on the routing table list.
Please remember to press the modify button to take it effect. For store
back to flash memory, please press /Syetem Maintenance/Save
Modification/.
30
2.4.2 VoIP Call In
2.4.2 /Routing Setup/VoIP Call In/
This page let you define the routing rule for Call in from VoIP. (VoIP Router got a
VoIP call required form Network, and then translates the phone number passed
from remote side VoIP Router to the real dial out number & line base on this VoIP
Call In routing table). Each time when the VoIP Router received a VoIP call from
Network, it will check with Area Code to see which rule matched to service, if no rule
matched, it will refuse to call out and will bound back the call.
When the VoIP Router received a VoIP called from network, it will check below rules
fields then decide line and number to dial out.
a.
Area Code: Define the Prefix number this rule service, any VoIP called from
network dialed number prefix digits matched with the rule will call out to phone
by this rule define. Please Notify there is a compare order rule on this routing
table. That mean the VoIP Router will check the rule list from top to bottom one
by one, any rule item matched with the prefix digits that user press will go to
call out directly no regard to the rest rules below.
For Example, if a rule item
for area code 8862 is on Index 1, another rule below that like index 2 for area
code 886 will be ignored.
Area Auth
Index
CallWaiting
Strip Prefix Maximum Minimum From To LineNo DisplayName
Alert
Profile
Forward
Delete
Code
1
886
□
□
Delete
2
8862
□
□
Delete
b.
Auth:Authorization IP check enable. Enable IP range authorization function.
When Enable, the gateway will check the remote caller IP range setting on
/VoIP
Setup/Routing
Setup/Authorization/,
if
it
is
within
the
authorization, the gateway will allow the call out, but if the remote caller’s IP
31
is not in the range, it will refuse to call out and terminate this call.
Area
Auth
Index
CallWaiting
Strip Prefix Maximum Minimum From To LineNo DisplayName
Alert
Profile
Forward
Delete
Code
1
8862
c.
4
5
□
Delete
Strip: Number of digits will be ignored by user input. For example, if received
VoIP call number is 886212345678 and the STRIP field is setting to 4, the first
4 digits 8862 will be truncated and actually call out number will be 12345678.
Area
Auth
Index
Display
Strip Prefix
Max
Min
Code
1
CallWaiting
From To LineNo
Alert
Profile
Forward
Delete
name
8862
□
4
□
Delete
Ex: VoIP Call in number is 886212345678 and real dial out number is
12345678 by strip 4 digits.
d.
Prefix: The numbers will be added on the prefix of received VoIP call number.
For examples, if received VoIP call number is 12345678 and the PREFIX field
is setting to 0028862, the actually call out number will be 002886212345678.
Area
Auth
Index
Display
Strip
Prefix
Max
Min
CallWaiting
From To LineNo
Code
Alert
Profile
Forward
Delete
name
1
□
0028862
□
Delete
Ex: VoIP Call in number is 12345678 and real dial out number is
0028862-12345678 by add 0028862 prefix.
Another example, if user VoIP router received a call number 90, STRIP field is
setting to 2, and the PREFIX field is setting to 0,12345678, the actually call
out number will be 0,12345678 ( , mean wait 1 second for PBX get line for dial
out to PSTN, the wait time for one , can be set on /VoIP Setup/Line
Configure/Line Feature/). This example is especially for speed dial
function.
Area
Auth
Index
CallWaiting
Strip
Prefix
Maximum Minimum
From
To LineNo DisplayName
Alert Profile Forward Delete
Code
1
90
□
2 0,12345678
□
32
Delete
Ex: VoIP Call in number is 90 and real dial out number is 0,12345678 by stripe
2 digits and add 0,12345678 prefix, so the real dial out number is
0,12345678.
e.
Maximum: Define the maximum digits of call number allow to dial. If the
length of dial number after pervious STRIP & PREFIX process is more than the
setting, it will deny dialing out. For example, you can set the Maximum dial out
digits is 8, for call to local area phone only, any VoIP call in attempt to dial
0712345678 out of 8 digits for call out long distance will been deny to call out.
Area
Auth
Index
CallWaiting
Strip
Prefix Maximum Minimum From To
LineNo DisplayName
Alert Profile
Forward Delete
Code
1
f.
□
8
□
Delete
Minimum: Define the minimum digits of call number allow to dial. If the length
of dial number after pervious STRIP & PREFIX process is less than the setting,
it will deny dialing out. For example, if set Minimum to 4, any VoIP call in
attempt to dial number less than 4 digits like 110, 911 will been deny to call
out.
Area
Auth
Index
CallWaiting
Strip
Prefix Maximum Minimum From To
LineNo
DisplayName
Alert Profile
Forward Delete
Code
1
□
4
□
Delete
Ex: VoIP Call in number is 911 and Minimum setting to 4, the VoIP router will
deny to call out.
g.
From: Define the beginning line number for service this area code VoIP
call. For example, if user assigned FROM 1 TO 1 for AREA CODE 601 in
this routing table, then any VoIP call for call in number 601 will ring the
line 1 only.
h.
To: Define the ending line number for service this area code VoIP call.
Area
Auth
Index
CallWaiting
Strip Prefix Maximum Minimum From To
LineNo DisplayName
Alert
Profile
Forward Delete
Code
1
601
□
1
1
□
Delete
2
602
□
2
2
□
Delete
□
Delete
3
□
33
Ex. Any VoIP Call in number with prefix 601 will ring the line 1, and Any VoIP Call in
number with prefix Call in number 602 will ring the line 2. any other numbers will
ring any available (not busy) lines.
i.
Line No: Click to enable if you want to force compare with the line number
setting on LINE CONFIGURE menu (/VoIP Setup/Line Configure/Line
Setting/). If the dial number after pervious STRIP & PREFIX process is
matched with the line number setting, the VoIP call will ring the dedicate phone
line that assigned with matched number.
j.
DisplayName:Enable Display Name function ,the device can display
the simple code.
For example :
From: 12736 <sip:+886987282721@ims.fchtet.net>
After enabled Display Name, you can display 12736。
k.
Call Waiting:Enable or Disable the call waiting function。
Enable:
„
and hold the original conversation. When some one call in
when you are busy on another phone call , you will hear a
du-du call waiting signal, please use flash key on your phone
to hold the original call and answer the incoming call, press
again flask key will switch back the original call party.
Disable: Disalbe the Call waiting function. The gateway will
reply the busy to remote side when the line is on used.
„
Area
During Talk, you can answer another phone call
Auth
CallWaiting
Index
Strip
Prefix Maximum Minimum From To
LineNo DisplayName
Alert Profile Forward Delete
Code
1
1
□
□
2
2
□
□
Delete
3
3
□
□
Delete
l.
Alert:
„
Enable
Delete
Control the Ring Back tone generate timing:
Mode 0: When this VoIP Router get ring back tone from phone line, it will
send the ring Alert signal to remote VoIP Router for generate ring back
tone.
„
Mode 1: Before this VoIP Router dial to phone line, it will send the
ring Alert signal to remote VoIP Router for generate ring back tone.
„
Mode 2: After this VoIP Router finish dial out number to phone line,
it will send Connect OK signal to remote VoIP Router.
34
„
Mode 3: Before this VoIP Router dial to phone line, it will send the
ring Alert signal to remote VoIP Router for generate ring back tone,
after this VoIP Router finish dial out number to phone line, it will
send Connect OK signal to remote VoIP Router.
m. Profile: Define the optional special VoIP parameters when received on this
destination. Please input the name you defined on the profile list (/VoIP
Setup/Routing Setup/Routing Profile/).
Example: if user set the VoIP Call in & Routing Profile like below:
Area
Auth
Index
CallWaiting
Strip
Prefix Maximum Minimum From To
LineNo DisplayName
Alert Profile Forward Delete
Code
1
1
□
□
2
2
□
□
3
3
□
□
Index
Name
ID1
VAD
AS
CODEC
ID2
Enable
pf1
Delete
pf2
Delete
Delete
H.245
DTMF
T.38
Package
Tunneling
Relay
FAX Relay
Frame
AS
ID3
AS
ID4
Q.931
Fast
Start
AS
Delet
e
1
2
pf1
ON
G.723.1
ON
00001
H.323
1001
E.164
pf2
ON
G.723.1
ON
00002
H.323
1002
E.164
Out band
OFF
3
ON
Delete
In band
OFF
3
OFF
Delete
When VoIP call in number with prefix 1 will use the Profile named PF1
( H.323 ID1 = 0001, E.164 ID=1001, DTMF Relay=Out band, Q.931
Fast Start=ON) to answer the VoIP Call in.
When VoIP call in number with prefix 2 will use the Profile named PF2
( H.323 ID1 = 0002, E.164 ID=1002, DTMF Relay=In band, Q.931 Fast
Start=OFF) to answer the VoIP Call in.
When VoIP call in number with prefix 3, because there is no Profile
assigned, it will use the default value for VoIP Call Out.
35
n.
Forward: Define the profile name for forward the unanswerable VoIP call on
this Call In rule. Please input the name you defined on the /Voip
Setup/Routing Setup/ Forwarding/ .
Example: if user set the VoIP Call in & Forward Profile like below:
Area
Auth
Index
CallWaiting
Strip
Prefix Maximum Minimum From To
LineNo DisplayName
Alert Profile Forward Delete
Code
1
601
□
1
1
□
2
602
□
2
2
□
3
□
Enable
PF1
CF1 Delete
PF2
CF2 Delete
□
Delete
Forwarding
Other: 10.1.1.1/104
No.
Name
Always
1
CF1
hk.big.com/301
2
CF2
OnBusy
No Answer
No Answer Sec Delete
Delete
assist.big.com
assist.big. om/610
30
Delete
In this case, when the VoIP router received a VoIP call in number with prefix
501 ( not 601 or 602 prefix defined on Call In Routing table), it will forward this
call to the IP & number defined on Other filed(in this case, 10.1.1.1/104, it
mean it will forward this call to IP 10.1.1.1 and calling number will change to
104).
When the line 1 user is going to have a tour to another location with same VoIP
router equipment, user setup the called prefix number 601 forward to profile
name CF1, and in CF1 profile, the Always field is set to hk.big.com/301, that
mean any call number with prefix 601 will always be forward to another VoIP
Router locate at hk.big.com and dial out number is 301.
When the line 2 is busy and another VoIP with prefix 602 Call in, it will forward
it to the assist.big.com with same number (defined on OnBusy).
When VoIP Call In number with prefix 602, it will ring the line 2 for 30 seconds
(defined on No Answer Sec.), if no one answer line 2 within 30 seconds, it
will forward the call to another VoIP Router located at assist.big.com and dial
out number is 610 (defined on No Answer)
36
o.
Delete: Delete this rule item on routing table.
To add new rule item on routing table, please assign the item number you want
to insert before, input AREA CODE then press ADD button to add it on the list.
Then modify the necessary information on the routing table list.
Please remember to press the modify button to take it effect. For store back to
flash memory, please press Save Modification (/Syetem Maintenance/Save
Modification/).
37
2.4.3 VoIP Call In IVR
2.4.3 /Routing Setup/VoIP Call In IVR/
When Enable the [Prompt Voice for VoIP Call In function on /VoIP Setup/Advance
Setup/Prompt Voice/ , all the remote party of VoIP caller will hear the
customization upload voice file and need press the destination number. All the
input number will be checked the number length and be strip/add prefix defined
on this page. When a matched area code be processed, it will use this number to
check on the /VoIP Setup/Routing Setup/VoIP Call In/ to decide the final route path.
If no match area code rule defined on the table, the gateway will response busy
tone and connect failure.
User can use this function as the password authorization on the outbound
gateway. For example, if we upload the voice file content on Prompt voice for
VoIP call in of /VoIP Setup/Advance Setup/Prompt Voice/ is “Please input the
password and destination number” and we set a compare rule as below:
Index
Remark
1
Password
Area
Code
8495
Min
Digits
7
Max Digits
Strip
Prefix
Delete
12
4
02
Delete
When a remote VoIP call in and want this gateway to outbound call, the remote
side user will hear voice prompt like
“Please input the password and
destination number”, because there is only a compare authorization rule, all the
none 8495 prefix phone number will not accept to dial out and will be disconnect.
( that mean we use 8495 as the outbound call authorization password), and the
digits of user input phone number should between 7 to 12 (include 4 digits come
from 8495),the number user input will strip the first 4 digits (8495) and add 02
prefix. Then this number will be checked by /VoIP Setup/Routing
Setup/VoIP Call In/ . For Example, if user input the number is849512345678,
it will strip 4 digits and add 02 prefix code, the use 0212345678 to find a call out
rule.
38
A. Time & Digits wait for user
The VoIP Router wait user input the number digits & time parameters as below:
Time & Digits wait for user Press.
a.
MaxDigits: Define the maximum digits wait for user press for all VoIP Call Out, if
user press digits match the number defined here. It will go to translate for call out
rule without needed to press # key.
b.
FirstDigitTime: Define the waiting time (seconds) for user press phone number
first digit. User need to press first digits before the setting time (seconds) defined
here, if VoIP Router wait for the defined seconds and there is no any digits press,
the VoIP Router will stop to wait and feedback the user busy tone.
c.
OtherDigitTime: Define the waiting time (seconds) for user press phone number
secondary & the rest digits. User need to press the rest digits before the
seconds
defined here, if VoIP Router wait for the defined seconds and there is no any digits
press, it will go to translate for call out rule without needed to press # key.
B.VoIP Call In IVR Routing Table
a.
Remark:
Remark for this routing rule. Please use UNDERLINE to replace the
SPACE due to HTTP protocol limitation.
b.
Area Code: Define the Prefix number fit this rule, any phone number prefix digits
matched with the rule will call out by this rule define. Please Notify there is a
compare order rule on this routing table. That mean the VoIP Router will check the
rule list from top to bottom one by one, any rule item matched with the prefix digits
that user press will go to call out directly no regard to the rest rules below. For
Example, if a number 84951xxxxxx is fit the rule item 1&2 , it will be processed by
rule 1 and never be processed by rule 2 , that mean that rule 2 is never been used.
Area
Code
8495
Index
Remark
1
password
2
No used rule 84951
Min
Digits
7
Max Digits
Strip
Prefix
Delete
12
4
02
Delete
39
c.
Min Digits: define the minimum digits wait for user press for number fit this rule,
if user press digits less the number defined here. It will keep waiting for input until
exceed the FirstDigitTime defined time. If user press digits more then Min Digits
here, the VoIP Router will wait time defined on OtherDigitTime then go to
translate for call out rule without needed to press # key.
d.
Max Digits: define the maximum digits wait for user press for number fit this rule,
if user press digits match the number defined here. It will go to translate for call out
rule without needed to press # key.
e.
Strip: the number of digits will be ignored by user input. For example, if user press
the number is 886212345678 and the STRIP field is setting to 4, the first 4 digits
8862 will be truncated and actually call out number will be 12345678.
f.
Prefix: The numbers will be added on the prefix of the user press number. For
examples, if user press the number is 12345678 and the PREFIX field is setting to
0028862, the actually call out number will be 002886212345678.
g.
Delete: Delete this rule item on routing table.
To add new rule item on routing table, please assign the item number you want to insert
before, input AREA CODE and IP address then press ADD button to add it on the list.
Then modify the necessary information on the routing table list.
Please remember to press the modify button to take it effect. For store back to flash
memory, please press /Syetem Maintenance/Save Modification/ .
40
2.4.4
Routing Profile
2.4.4/Routing Setup/Routing Profile/
This page defines the optional special VoIP parameters when making/received a VoIP
call. For define some special parameters for different VoIP equipment or authorize
purpose, please add a profile at /VoIP Setup/Routing Setup/ Routing Profile/ and
use the same name as the profile on the Call in Routing Table (/VoIP Setup/Routing
Setup/VoIP Call In/) or Call out Routing table (/VoIP Setup/Routing Setup/VoIP
Call Out/).
a.
Name: Specify a profile name. Please use UNDERLINE to replace the SPACE due to
HTTP protocol limitation.
b.
c.
VAD:
„
ON: turn on the VAD(Voice Active Detection) function.
„
OFF: turn off the VAD function, please select ON for save the bandwidth.
CODEC: Select different voice CODEC for VoIP communication. The bit rate of
G.723.1 is 5.3k/6.3k, G.729 is 8k, uLaw and aLaw is 64k per second. The G.723.1
is default CODEC.
d.
e.
H.245 tunneling:
„
ON for enable H.245 tunneling.
„
OFF for disable H.245 tunneling.
DTMF Relay:
41
When select In band to transfer the DTMF during VoIP, the user pressed DTMF
„
tone will be treat as general voice and been compressed then transmit to
remote side to decompress play back, it maybe cause some problem on
duplicate or missing DTMF receive.
When select Out band to transfer the DTMF during VoIP, the user pressed DTMF
„
tone will be decode by local VoIP Router then transmit as signal, after received
on received remote VoIP Router, it will be regenerate by remote VoIP Router.
The default value is Out band.
f.
g.
T.38 FAX Relay:
„
ON: FAX will be transmitted by using T.38 FAX over IP protocol.
„
OFF: FAX over IP is disabled.
Package Frame: Select the number of voice payload frames on each UDP package
VoIP transmit. More frames into one package mean save more bandwidth. The
default frames on each package is 3.
h.
Q.931 Fast Start:
„
ON:
Enable Fast Start capability during Q.931 handshaking.
„
OFF: Disable Fast Start capability during Q.931 handshaking.
i.
ID1: User defines ID #1 during this VoIP call.
j.
As:
„
E.164: Parameter on ID1 field is the E.164 during this VoIP call.
„
H.323 ID: Parameter on ID1 field is the H.323 ID during this VoIP call.
„
Calling: Parameter on ID1 field is DID number during this VoIP call. If this
optional is setting, it will override the LINE NUMBER on line setting menu.
„
Password: Parameter on ID1 field is the password for VoIP call. Parameter
defined here will used as MD5 during H.235 and will not display on the Web UI
k.
ID2,ID3,ID4: there are 4 fields for user define the ID parameters, please
reference the ID1 setting above.
l.
Delete: Delete this rule item on routing table.
To add new profile item on routing table, please assign the number you want to
insert before, input profile NAME then press ADD button to add it on the list.
Then modify the necessary information on the routing table list.
Please remember to press the modify button to take it effect. For store back to
42
flash memory, please press Save Modification (/Syetem Maintenance/Save
Modification/).
Here is VoIP Routing Profile setting examples below:
Index
1
2
Name
VAD
CODEC
H.245
DTMF
T.38
Package
Tunneling
Relay
FAX Relay
Frame
Q.931
Fast
Start
ID1
AS
ID2
AS
ID3
AS
ID4
AS
pf1
ON
G.723.1
ON
Out band
OFF
3
ON
00001
H.323
1001
E.164
pf2
ON
G.723.1
ON
In band
OFF
00002
H.323
1002
E.164
****
Password
Delete
Delete
2
OFF
Delete
When using profile PF1 , the parameters will be used for H.323 ID1 = 0001, E.164
ID=1001, DTMF Relay=Out band, Q.931 Fast Start=ON) to call / answer the VoIP.
When using profile PF2 , the parameters will be used for H.323 ID1 = 0002, E.164
ID=1002, DTMF Relay=In band, Q.931 Fast Start=OFF, Password=1234 but be hidden )
to call / answer the VoIP.
43
2.4.5 Forwarding
2.4.5 /Routing Setup/Forwarding/
This page defines the forwarding behavior include:
z
get an unmatched prefix number for VoIP call in,
z
Line busy
z
No answer
Please add a profile at /VoIP Setup/Routing Setup/Routing Profile/ and put the
name of profile on the Call out Routing table (/VoIP Setup/Routing Setup/VoIP Call
Out/).
a.
Other: Define the forward IP and forward phone number when there is no
match rule setting on VoIP Call Out Routing table. The format is IP/phone
number or URL/phone number. i.e. all the phone number can not match a
prefix rule will be forward to the IP& phone number define on here.
b.
Name: Specify a profile name. Please use UNDERLINE to replace the
SPACE due to HTTP protocol limitation.
c.
Always: Always redirect forward to this IP(or URL)/phone number, All
incoming call will be forward to IP assigned here.
d.
On Busy: Redirect forward to this IP(or URL)/phone number when busy, an
incoming VoIP call will forward to IP assigned here when this line is busy.
e.
No Answer: Redirect forward to this IP(or URL)/phone number when no
answer over the time No Answer Sec , an incoming VoIP call will forward
to IP assigned here when ring time over the defined on No Answer Sec.
44
f.
No Answer Sec. Defined the wait seconds for redirect forward to another
IP(or URL).
g.
Delete: Delete this rule item on routing table.
h.
To add new rule item on routing table, please assign the item number you
want to insert before, input AREA CODE then press ADD button to add it on
the list. Then modify the necessary information on the routing table list.
Please remember to press the modify button to take it effect. For store back to flash
memory,
please
press
Save
Modification
(/System
Maintenance/Save
Modification/).
Example: if user set the VoIP Call in & Routing Profile like below:
Index Area Code Strip
Prefix
Maximum Minimum From To LineNo Gatekeeper
Alert
Profile
Forward
Delete
1
601
1
1
cf1
Delete
2
602
2
2
cf2
Delete
Other: 10.1.1.1/104
No.
Name
Always
OnBusy
1
cf1
hk.big com/301
2
cf2
No Answer
No Answer Sec Delete
Delete
assist.big.com
assist.big.com/610
30
Delete
In this case, when the VoIP router received a VoIP call in number with prefix 501 ( not
601 or 602 prefix defined on Call In Routing table), it will forward this call to the IP&
number defined on Other filed(in this case, 10.1.1.1/104, it mean it will forward this
call to IP 10.1.1.1 and calling
number will change to 104).
When the line 1 user is going to have a tour to another location with same VoIP router
equipment, user setup the called prefix number 601 forward to profile name CF1, and
in CF1 profile, the Always field is set to hk.big.com/301, that mean any call number
with prefix 601 will always be forward to another VoIP Router locate at hk.big.com and
dial out number is 301.
When the line 2 is busy and another VoIP with prefix 602 Call in, it will forward it to the
45
assist.big.com with same number (defined on OnBusy).
When VoIP Call In number with prefix 602, it will ring the line 2 for 30 seconds (defined
on No Answer Sec.), if no one answer line 2 within 30 seconds, it will forward the call
to another VoIP Router located at assist.big.com and dial out number is 610 (defined
on No Answer) .
46
2.4.6
Authorization
2.4.6 /Routing Setup/Authorization/
When this gateway has been used for outbound call, it can enable to check the
remote caller gateway’s IP to decide accept or refuse the call. If define the IP range
here and enable the [Auth] option on the/VoIP Setup/Routing Setup/VoIP
Call In/ , only the IP in range will allow to call out by this gateway.
47
2.5
2.5.1
Register Server
Register Status
2.5.1 /Register Server/Register Status
You Can check the register status of this gateway on this page.
a. MAC: this gateway’ MAC Address。
b. RS1-4: Indicate the status of 4 server register.
„
„
„
„
SIP & H323: The protocol used for registering the server, this
gateway supports both H.323 and SIP protocol.
Green Indicator: Successful to register server and the register phone
number.
Red Indicator: Failure to register server and the failure reason.
Yellow Indicator: Disable the register function.
Example: for Status display as above, it indicates:
1. The register to Server #1 function is disabled (SIP).
2. Use SIP protocol to register to register as RS2, the register method is
4 lines independent. Each lines use different number to register:
25618801, 25618802, 5618803, 25618804. Line 2 and Line 4 are
disabled to register, Line 1 and Line 3 are successful to register.
3. Use H.323 protocol to register as RS3, all 4 lines share same register
102003
4. Use SIP protocol to register as RS4, each lines use different number
as 102002, 102003, 77201111. You can see that line 2 and line 3
48
register failures. The line 2 failure reason is “unauthorized” and Line
4 failure reason is “not number”.
Please setup each register parameters at /VoIP Setup/Register
Status/Server#1~4/。
49
2.5.2
Register Server—SIP Protocol
圖 2.5.2 /Register Server/Server
If you need use this gateway to register to the H.323 gatekeeper or SIP
register/proxy server, you can setup the account for register here. This gateway
can register up to four Servers simultaneously.
a. Protocol: Select use SIP or H.323 protocol to register to server, by
different protocol, the gateway will adjust the page for different
parameters for input.
b. Register Method:
„ Global: All the lines of the gateway share same account to register.
„ Independent: Each lines can set different/same account independently
for register.
c. Enable SIP Proxy :
„ ; Enable Register SIP Proxy server function.
„ … Disable Register SIP Proxy server function.
d. SIP Proxy URL: Please input the IP/URL of the SIP proxy server.
e. Port [1~65535]: Port number used for register to server. The SIP
protocol default is 5060, please make sure you have same port number
setting on the gateway and server.
50
f. Through Outbound Proxy:
When your gateway is installed behind the firewall or NAT, you maybe
need use Proxy server to relay your call. If so, please input the
Outbound proxy server’s IP here.
g. Prot[1~65535]: Port number used for register Outbound Proxy Server.
h. TTL(Registration interval)[10-7200s]: Some SIP Server need you
set the time interval (seconds) for send the expire signal to register
server keep alive.
i. Domain: Some SIP Server need you input the Domain for register,
please input here.
j. Proxy Require: Some SIP Server (Nortel) need you input the more
information for proxy function, please input here.
k. Line: Number index of lines.
l. Type: Interface type of the line. FXS:
connect to phone set or PBX Co. line.
Analog phone interface for
m. Remark: Remark for this routing rule. Please use UNDERLINE to replace the
SPACE due to HTTP protocol limitation.
n. Number: Register phone number, Some SIP Server needs this to
parameters for register.
o. Account: Account for register to SIP server.
p. Password: Password for register to SIP server.
q. Conference ID: Some SIP Server requires an ID to enable the
conference function, please input the ID here to enable that.
r. Enable: Enable or disable independently each line for register.
51
2.5.3
Register Server—H.323 Protocol
1.14 /VoIP Setup/Register Server/Server
When Select use H.323 to register gatekeeper, please input the flow information
for register:
a. Protocol:
„ SIP
„ H323
b.
Register:
„ Global: All the lines of the gateway share same account to
register.
„ Independent: Each line can set different/same account
independently for register.
c.
Enable H323 Gatekeeper :
„ ; Enable Register H.323 Gatekeeper function.
„ … Disable Register H.323 Gatekeeper function.
d.
Gatekeeper URL: Please input the IP/URL of the Gatekeeper server.
e.
Prot[1~65535]: Port number used for register to server. The H.323
protocol default is 1719, please make sure you have same port
number setting on the gateway and server.
f.
GK ID: Some Gatekeeper Server need you input an ID for register,
please input here
52
g.
Proxy for NAT: When your gateway is installed behind the firewall
or NAT, you maybe need use Proxy server to relay your call. If your
gatekeeper supports this proxy function, you can enable gateway
function here to use that.
h.
Line: Number index of lines.
i.
Type: Interface type of the line:
„ FXO: Analog phone interface for connect to PSTN or PBX
extension line.
„ FXS: Analog phone interface for connect to phone set or PBX Co.
line.
j.
Remark: Remark for this routing rule. Please use UNDERLINE to replace
the SPACE due to HTTP protocol limitation..
k.
l.
m.
n.
o.
E.164: phone number used for register to server.
H.323 ID: Account name used for register to gatekeeper.
Account: Account name used for register to gatekeeper
Password: Password used for register to gatekeeper.
Enable: Enable or disable independently each line for register.
Please remember to press the Modify button to take it effect. For store back to
flash memory, please press Save Modification (/Syetem Maintenance/Save
Modification/).
53
2.6
2.6.1
Advance Setup
NAT Traversal
2.6.1 /Advance Setup/NAT Traversal
If your VoIP gateway is installed behind NAT, you may need a special
configuration and server to establish the VoIP communication, this gateway
support several method for NAT Traversal as below:
z
By Outbound Proxy:
User can appoint an Outbound Proxy Server to handle the NAT traversal on
VoIP Setup/Register Sever/Server #/
z
Declare NAT IP address:
Select to enable the input the NAT router IP of the network.
z
Use STUN server
Enable STUN and input the STUN server’s IP for handle the NAT traversal,
you can input 2 sets of STUN servers IP.
The Gateway will display the system found NAT IP address.
For store back to flash memory, please press Save Modification (/System
Maintenance/Save Modification/).
54
2.6.2
Listen Port
2.6.2 /Advance Setup/Listen Port
In this page, user can define the usage port for setup the VoIP communication.
Both side of gateways need use the same port for begin VoIP communication.
a.
SIP Listen Port : Define the listen port for SIP protocol, the default
port is 5060, input range from 1024 to 65535.
b.
H.323 Call Signal Port: Define the Call signal port for H.323
protocol, the default port is 1720, input range from 1024 to 65535.
c.
.H.323 Gatekeeper Listen Port: Define the Gatekeeper listen port
for H.323 protocol, the default port is 1719, input range from 1024 to
65535.
d.
RTP Initial Port: Define the RTP package initial port, the input
range from 1024 to 65535. the gateway will display the used UDP
ports due to multiple lines connection.
After modify and press Modify, system will save and reboot automatically
to take it effective.
55
2.6.3
VoIP Package
2.6.3/Advance Setup/VoIP Package
User can define the parameters relative about VoIP package on this page.
z
Jitter Buffer(ms):
Define the Jitter buffer size, input range is from 20 to 200ms.
z
VoIP DTMF Relay Mode:
Define the relay mode for DTMF signal:
„ In band: When local gateway detects a DTMF signal, it will not decode it .
The DTMF signal will been compress/decompress as VoIP voice package.
„ Out band: When local gateway detects a DTMF signal, it will decode it,
and relay it as a data package separately. The remote gateway will
regenerate the DTMF signal after receive the DTMF data package. System
default is relay DTMF by out band mode.
z
VoIP DTMF Relay Mode (Out band),:
Define 2 methods to relay DTMF when select Out band relay mode:
„ by SIP:RFC2833 (SIP protocol) or H.323:H.245 (H.323 protocol)
„ by SIP INFO
(SIP protocol) or Q.931 (H.323 protocol)
56
z
RFC2833: Payload number for DTMF[96~127]:
Define the DTMF token on RFC2833, input range form 96 to 127.
z
RTP Packet NAT Detect by:
User can select the packet transmit method when under the NAT network
environment, by IP (NAT 255 fake IP), or transmit by TCP port, usually
suggest select the option “By IP or By Port”.
z
Extension Number SIP Header To:
Use the number in the SIP header column as the dial out number.
z
Supported SIP 100rel:
The name of the option tag for the SIP extension that is supported by this
SIP protocol is 100rel, it is used for Sending Reliable Provisional Responses.
Enable this item if needed.
z
RTP Packet Monitor Time [0 Disable](sec)
If set 60 sec, when network failed over 60 seconds, device will cut this call
automatically.
z Silence Detection / Suppression:
Enable or disable the Silence Detection/VAD function. When Enable, if local
gateway detect a silence situation ( no talk), it will send a VAD package rather
than a full voice package for remote side to active CNG ( Comfort Noise
Generation) to save the bandwidth. The default is Enable to save the
bandwidth.
z
Prefer CODEC :
In this table, you can define the prefer CODEC. The priority 1 selection is
highest priority. By different CODEC, user can select different payload size
per package as below:
„ G.711 uLaw:
20,30,40,50,60,70,80ms
„ G.711 aLaw:
20, 30,40,50,60,70,80ms
„ G.723.1:
30,60,90ms
„ G.729a:
20,30,40,50,60,70,80ms
„ G.726:
20,30,40,50,60,70,80ms
„ None:
none
The gateway will calculate and show approximately bandwidth for one VoIP
call.
57
2.6.4
RTP Packet Summary
2.6.4 /Advance Setup/RTP Packet Summary
On this page, user will know the RTP package summary about last VoIP call.
z
z
Line#: number of line
Using CODEC: ex.: G.729a
z
z
z
z
z
z
Source IP: Remote side IP
Source Port: Remote side port
Packet Interval: interval time between 2 packets.(ms)
Packet Send: number of packets sent.
Packet Received: number of packets received.
Packet Lost: number of lost packets.
58
2.6.5
Flash & Call waiting
2.6.5 /Advance Setup/Flash & Call waiting
On this page, user can define the parameters relative to the FLASH key and Call
Waiting function. These functions usually work with PBX
z
Token for flash key on VoIP(!):
Define the token for flash key during VoIP protocol ( use “! “ by default).
z
Flash Signal generate length :
Define the pause time (ms) for one “,” symbol at /Routing Setting/VoIP Call
Out/. This pause till is useful for PBX seize the trunk line from extension line. The
default time is 1000ms, Input range from 100 to 3000ms.
z
Flash Signal Detect Threshold:
Define the threshold for valid FLASH signal. Only the flash time length between
setting between min. to max. is accept by the gateway.
z
VoIP User Agent:
You can fill in the value to this packet, usually it used for the communicating
recognization.
z
VoIP Transfer:
Select the proper transfer method for VoIP in this item, Unattended (For INVITE),
Attended (For INVITE) , Attended (For INFO Mode1) ,Attended (For INFO Mode2)
or Flash+N(For CHT).
59
2.6.6 Gain
2.6.6/VoIP Setup/Advance Setup/Gain
This page defines different function gain on the gateway.
z
Gain when Dial tone phase:
When phone off hook, user will hear the dial tone generated from the
gateway, sser can adjust the play/record gain during this phase for
stable DTMF detection. After connection, the gains setting here is no use,
the gateway will adjust the gain setting on /Line Configure/Line
setting/
„ Play: Transmit gain from network to line. Adjust the speaker volume
on the handset. Higher value will louder the speaker on local side.
„ Record: Receive gain from line to network. Adjust the microphone
volume on the handset. Larger value which will amplifier the MIC
volume on local site.
Incorrect value will cause the gateway can not receive the DTMF user
pressed on phone set, please use the default 0dB if no other issue.
z
DTMF Generate DSP play Gain [-29~3]
Setting the internal gain used by DSP for generate the DTMF signal,
incorrect value will cause the DTMF can not accept by other telephone
equipment. Please use the default value if no other issue.
z
Call progress Tone DSP play Gain[-31~0]
Setting the internal gain used by DSP for generate the CPT ( Call Progress
Tone), Incorrect value will cause the DTMF can not accept by other
telephone equipment. Please use the default value if no other issue.
60
z
Caller ID Detection record Gain [13~-3]
Setting the Caller ID Receiver gain. Incorrect value will cause the Caller ID
signal can not be receive, please use the default value if no other issue.
z
Gain when VoIP DSP Talking phase
This item allows user to adjust the talking volume of the device, which
means it will affects all ports of this device, User can select the db value of
Play and Record, it will increase or decrease the talking and listen volume.
User can adjust individual port talking or listen volume in page Line
Configure Æ Line Setting.
61
2.6.7 QoS
2.6.7 /Advance Setup/QoS
User can define the ToS field on the VoIP packet for Quality of Service control.
The ToS field is included these 2 parameters:
z Precedence: 0~7
z DSCP(Diffserv Code Point):0~46
User can select inptut either IP Precedence or DSCP value. Or input the ToS
binary code directlly.
The QoS function works depending on the router in front of the VoIP device,
which means the router need to support the Tos, diffserv recognition also.
User can select Tos or diffserv to define the QoS method, voip device need to
match the Tos tag or value of diffserv with the router, so that the voice packets
can be sent with priority.
62
2.6.8 CDR
2.6.8 /Advance Setup/CDR
The Gateway can export all the CDR (Call detail Record) to external CDR server
by HTTP protocol. The gateway supports up to 2 CDR servers for keep the
record.
z
z
z
Export to CDR Server: Please install and enable this function if you want to
keep CDR of this gateway.
CDR Server IP1: Please input the IP of first CDR server, if installed.
CDR Server IP2: Please input the IP of second CDR server, if installed.
63
2.6.9 FoIP
2.6.9/Advance Setup/FoIP
User can define the parameters relative FAX Over IP function.
„
„
FXS Mode:Disable , T.38 , T.30 , T38+T30
Maximum FoIP Rate
Appoint the maximum FAX transceiver rate during FoIP:
‹ Disable: Only the VoIP function is supported on the gateway. FoIP is
disabled.
‹ Auto: Gateway will negotiation the maximum speed for FoIP with
FAX machine..
‹ Appoint the Maximum speed:2400,4800,9600,12000, 14000
„
CED Tone(2100 Hz) Detection Gain: [-45~0]
64
2.6.10
Prompt Voice & Beep
2.6.10 /Advance Setup/Prompt Voice & Beep
This gateway can use voice or beep to prompt the user different situation. User
can download/upload their own prompt voice wave files also.
For prompt beep enable function, it can be enable by:
„ VoIP Call out Beep:
When enable, the gateway will generate a beep for call out for VoIP
VoIP Call out Failure twice Beep
When enable, the gateway will generate twice beep if failure to call out
for VoIP
„
„
No PSTN line warming twice Beep :
When enable, the gateway will generate twice beep if failure to call out
for PSTN, usually mean there is no trunk line connect to the PSTN line
port.
„
Can not register to server warming twice Beep:
When enable, the gateway will generate twice beep when end user off
hook the phone if the gateway failure to register to Register Server.
For prompt voice enable function, it can be enable by:
65
„
„
„
„
„
Prompt voice for replace dial tone:
Use a customize voice file to replace the dial tone
Warming Prompt after VoIP out failure:
Annunciate a customize voice file when VoIP call failure.
Can not register to server warming prompt :
Annunciate a customize voice file when the gateway failure to register to
Register Server.
Prompt voice for VoIP call in :
When Remote gateway call in the gateway, if enable this function, the
gateway will annunciate a customize voice to remote gateway user to
ask the destination number , this function must work with the setting
rule on /VoIP Setup/Routing Setup/VoIP Call In IVR.
Warning Prompt after Check ANI number failure :
Annunciate a customize voice file when check ANI number failure.
For enable the prompt beep or voice annunciation function, please select ;
to
enable the function and click Modify
Caution: If enable both prompt beep and Voice annunciation on the same
function, only
the voice annunciation will work and will not hear the beep sound.
Procedure to upload customize voice wave file:
1.Select the function index you want to modify.
2.press brows , select the content voice file.
3 Press Restore to upload and save.
4.To keep the voice file permanently, press Save Flash to save it
*. The gateway only accepts the G.723.1 or G.711 format voice file, and all
the 5 files size totally can not exceed 384KB.
66
2.6.11
Call log
2.6.11/Advance Setup/Call log
„
„
The gateway can export status and performance to remote system.
Syslog supports eight leavels:
Disable、Emergency、Alert、Critical、Error、Warning、Notice、Info、
Debug
67
2.7
Application
2.7.1
Ping Test
2.7.1 /Application/Ping Test
User can use the Ping Test function to test the network status or remote
device
„
Ping Destination:
Target device IP for ping test.
„
Number of Ping[1-100]:
Number of ping test, maximum is 100.
„
Ping Packet Size[56-5600 bytes]:
Size of ping test packet, input range is between 56 to 5600 bytes。
68
2.7.2 Centrex
2.7.2/VoIP Setup/Application/Centrex
If you are connected to the PSTN device is applied Centrex service , please
enable Centrex access code, when you press the access code , you will hear a
dial tone.
69
2.7.3
Telnet & SNMP
2.7.3 /Application/Telnet & SNMP
User can enable and set the user account for the SNMP and Telnet function of
this gateway on this page.
z
a.
Enable telnet server
„ Enable: Enable telnet service function, user can telnet this
gateway.
„ Disable: Disable telnet service function.
b.
Enable SNMP Server
„ Enable: Enable SNMP service function, user can use SNMP on the
gateway.
„ Disable: Disable SNMP service function.
c.
d.
e.
User Name:Set a user name for Telnet & SNM login.
Login Password:Set the password for Telnet & SNMP login.
Confirm Password:Check the password again.
How to use Telnet :
1. Using telnet login:
70
voip login:
password:
welcome
2. telnet command:
Help
sini
cat
save
ping
tracert
help
exit
;
3. Use the ping command to test ping time:
ping www.hinet.net
pinging www.hinet.net (203.66.88.89) With 32 bytes:
32 byes from 203.66.88.89: seq=1 ttl=249 time=40ms
32 byes from 203.66.88.89: seq=2 ttl=249 time=40ms
32 byes from 203.66.88.89: seq=3 ttl=249 time=40ms
ping statistics www.hinet.net:
packets: send=4 received=4 lost=0
4. Use the tracert command to test the routing rule :
tracert www.hinet.net
[tracer
hinet.net]-10...
tracert www.hinet.net
tracert to www.hinet.net (61.219.38.89), 30 hops max, 40 byte packets
1 172.16.7.1 (172.16.7.1) 0ms 0ms 10ms
2 220-139-40-254.dynamic.hinet.net. (220.139.40.254) 40ms 40ms
50ms
3 tp-sc-c6r1.router.hinet.net. (168.95.170.34) 40ms 40ms 50ms
4 tp-sc-c12r1.router.hinet.net. (203.75.134.2) 40ms 50ms 40ms
5 tp-s2-c12r1.router.hinet.net. (210.65.2.162) 50ms 40ms 40ms
6 tp-s2-c6r9.router.hinet.net. (211.22.35.1) 50ms 40ms 50ms
7 ***
tracert end
71
2.7.4 Call Timer
2.7.4 /Application/Call Timer
If youneed to limit the user’s talk time per call , you can set the call timer.
„ Pre Alam.
„ Line
„ Remark
„ Area Code
„ Miniute
„ Enabel
„ Delete
72
3. System Setup
3.1
3.1.1
System
System Status
3.1.1 /System/System Status
This page reveals the status of the gateway including WAN, LAN and some
hardware information.
Internet
This sub-block shows the Internet information of your home gateway. It
depends on the WAN mode connecting to your ISP. The different items
correspond to each WAN mode will be revealed after the common part of the
Internet status sub-block.
73
Cable/DSL
This field indicates the Internet connection status.
Connected , Disconnected or Connecting.
Its value is
WAN IP
Connected to the Internet through Cable or ADSL modem, the ISP will
offer the home gateway a WAN IP address to communicate with other
hosts in the Internet.
Subnet Mask
This field indicates a mask used to determine what subnet the WAN IP
address belongs to. An IP address has two components, the network
address and the host address. For example, consider the IP address
192.168.168.182 with subnet mask is 255.255.255.0, the first three
numbers (192.168.168) represent the Class C network address, and the
forth number (182) identifies a particular host on this network.
Gateway
“Gateway” is a node on a network that serves as an entrance to another
network. For the home gateway, The “Gateway” is the next device, which
routes the traffic to the Internet.
DNS
Domain Name System (or Service or Server) is an Internet service that
translates domain names into IP addresses. Because the domain names
are alphabetic, they are easier to remember. However, the Internet is
based on IP addresses. Every time you use a domain name, a DNS
service must translate the name into the corresponding IP address. For
example, the domain name www.example.com might translate to
198.105.232.4. The DNS system has its own network. If one DNS
server doesn't know how to translate a particular domain name, it will ask
its upper stream server, and so on, until the correct IP address is returned
or timed-out.
Secondary DNS
This is the secondary DNS to use when the primary DNS does not work.
Domain Name
Domain name is a name, which identifies one or more IP addresses.
field represents the domain name obtained from your ISP.
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This
Connection Type
There are five ways to get the WAN IP address. They are DHCP, STATIC,
PPPoE, PPTP and L2TP. This field indicates the way to get the WAN IP
address. Through Figure 3-2 to 3-6 detail all specific items of each mode.
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Gateway
IP Address
This field is the LAN IP address of the home gateway.
Subnet Mask
This field is the subnet mask of the network in the LAN side.
DHCP Server
The home gateway supports DHCP service. This field indicates the enabled
status of the DHCP Server.
NAT
This field shows whether the NAT is enabled or not.
Firewall
The gateway supports firewall service. This field indicates firewall service is
enabled or not.
Information
System Up Time
Shows the time in hh:mm:ss format from when the home gateway was
powered up to the web browser requests this page.
System Date
Shows the data and time in mm/dd/year hh:mm:ss when the web browser
requests this page.
Connected Clients
This field shows how many clients in the LAN clients connect to the home
gateway.
Runtime Code Version
Shows the version of runtime code.
Boot Code Version
Shows version of boot code.
LAN MAC Address
Short for Media Access Control address, a hardware address that uniquely
identifies each node of a network. This field indicates gateway’s LAN MAC
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address.
WAN MAC Address
This field indicates gateway’s WAN MAC address.
Hardware Version
Tells the version of hardware of the gateway.
Serial Number
This field indicates serial number.
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3.1.2
System Settings
3.1.2 /System/System Settings
This page is used to configure the names given by the ISP, if any, to represent
the gateway.
‹
‹
Host Name
Some ISPs request the host name to represent the home gateway. Fill the
host name given by the ISP, or you may not be able to access the Internet
successfully. The maximum length of the host name is 32 bytes.
NTP Server
Network Time Protocol is used to obtain the time from the Internet NTP
server. The home gateway will resolve the NTP server from the internal
URL lists. If you know a better NTP server, you can enter it. Domain name
and IP address format are both acceptable.
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3.1.3 Date&Time
3.1.3/System Setup/System/Date&Time
This page is used to setting the system time of VoIP gateway; it can define the
correct time by which ways.
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3.1.4
Administrator Settings
3.1.4 /System/Administrator Settings
This page allows you to change the user and administration password used to
manage this router for security reasons.
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3.2
VDSL
3.2.1 VDSL Settings
3.2.1.1/VDSL/VDSL Settings
Operating Mode: You can select CO(COT) or CPE(RT) Mode.
Profile Enable: You can select 8a, 8b, 8c, 12a, 12b, 17a, 30a.
G.hs Carrier Ser: You can select Auto, A43, B43 or V43
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3.2.2 VDSL Status
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3.3
WAN
3.3.1
WAN Settings
3.3.1.1/WAN/WAN Settings
There are four ways to connect to the Internet.
They are Dynamic IP , Static IP, PPPoE, Bridge.
IP
Dynamic IP
3.3.1.2/WAN/WAN Settings/Dynamic IP
‹
‹
‹
Domain Name
User configured domain name of the network maintained by gateway.
DHCP Vendor Class ID
This is a packet label which can be used to recognize for different Vendors,
fill in this area if needed. The value of this class ID can be checked by
software like ethereal.
Client ID (Option 61) , Option 61 allows the DHCP Server to be setup to
provide a preprogrammed IP address to a field device based on its Media
Access Control (MAC) Address or an ID entered into the field equipment.
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‹
‹
Request IP address
You can specify the IP address you desired. But the ISP has the right to
neglect it and provides you a different one.
MTU (576-1500)
You can specify the MTU (maximum transmission unit) of your home
gateway. The default value is 1500 bytes and in normal case, you don’t
have to change.
Static IP
3.3.1.3/WAN/WAN Settings/Static IP
‹
IP address assigned by your ISP
Set the IP address that assigned by the ISP.
‹
Subnet Mask
Set the subnet mask of the network.
‹
ISP Gateway Address
Set the ISP’s gateway IP address. This address routes packets to Internet.
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PPPoE
3.3.1.4/WAN/WAN Settings/PPPoE
This page is the PPPoE configuration page. Most of the ISPs request the
user to connect to central office (CO) side via PPPoE, acronym of
Point-to-Point Protocol over Ethernet, which provides authentication,
authorization and accounting.
‹
‹
‹
‹
‹
User Name
Enter the user name provided by your ISP to identify the computer using
PPPoE.
Password
Enter the password provided by your ISP to identify the computer using
PPPoE.
Please retype your password
Retype the password to make sure type correct password.
Service Name
Some ISP provides the service name of this PPPoE connection. If so, enter
this item, or make it blank.
MTU ( 546-1492 )
Maximum Transmission Unit (MTU) is the largest physical packet size
measured in bytes, which a network can transmit. Any messages larger
than the MTU are divided into smaller packets before sent. In ordinary,
that the user does not have to worry about the MTU size, the gateway
routing engine will handle the MTU differences between PPP and the LAN
Ethernet side. But for some old PPPoE server, you have to make the MTU
size of the PPPoE side smaller than the default value, or some Web side is not
able to access.
PPPoE MTU should be set between 546 and 1492.
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Bridge
3.3.1.5/WAN/WAN Settings/Bridge
User can set this device work as a bridge, set the necessary information.
A network bridge connects multiple network segments at the data link
layer (layer 2) of the OSI model, and the term layer 2 switch is often used
interchangeably with bridge.
Bridging group IP address
Enter the bridging IP address in this column.
Bridging group Subnet Mask
Enter the bridging Subnet Mask in this column.
Bridging group Gateway Address
Enter the bridging gateway IP address in this column.
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3.3.2 DNS
3.3.2/WAN/DNS
This page sets the primary and secondary DNS servers, which were given by
your ISP or known to you. When a domain name request received, the gateway
tries to resolve to it from the Primary Domain Name Server. Resolving failed,
the gateway tries the Secondary sever again.
‹
DNS Proxy
Enable or disable DNS Proxy.
‹
Domain Name Server
Set your primary DNS in this field.
‹
Secondary DNS Address(optional)
Set your secondary DNS to use when the primary DNS does not work
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3.4
LAN
3.4.1
LAN Settings
3.4.1 /LAN/LAN Settings
The home gateway is an IP sharing device, which provides the home users
share the same public IP address. While in the LAN side, each network
device must have one private IP address to do network communication.
This page is to set the configuration of the LAN interface of the gateway.
‹
‹
‹
‹
‹
IP Address
Set this to be gateway’s LAN interface IP address. The LAN interface
address is also aced as the default gateway address to the computers in you
private network.
Subnet Mask
This field indicates a mask used to determine what subnet the LAN IP
address belongs to.
The Gateway acts as DHCP Server
Check this item, if the home gateway supports DHCP server service. This is
the normal case that can make you free from installing a DHCP server in
your home network. And you can connect to the Internet just as what you
have done in your company environment.
IP Pool Starting Address
The starting address provided by DHCP service
IP Pool Ending Address
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‹
‹
‹
The ending address provided by DHCP sevice.
Lease Time
Set the lease time of the IP address to renew.
Local Domain Name
Set the gateway’s local Domain Name.
DNS Proxy
The home gateway acts as a DNS Proxy. In this case, the DHCP service will
set LAN interface IP address as the DNS server address, and inform the
clients in the DHCP renew process.
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3.4.2 DHCP Client List
3.4.2 /LAN/DHCP Clients List
This page lists all the DHCP clients in the LAN side.
capable of administering 253 clients.
‹
‹
‹
‹
The DHCP server is
Host Name
This column shows the host name of the DHCP clients.
IP Address
This column shows the IP Address of the DHCP clients.
MAC Address
This column shows the MAC Address of the DHCP clients.
Remaining Time
This is the amount of time the lease is valid if not renewed before it expires.
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3.5
3.5.1
Wireless
Basic Settings
3.5.1/Wireless/Basic Settings
Enable Wireless LAN Interface:On / Off Wireless.
‹ Band:Set Wireless Band.
‹ Mode:You can select Accecess Point mode or AP Client mode for your wireless
environment.
‹
SSID: SSID is a name that identifies a wireless network. You can define SSID for
your own wireless network.
‹
‹
‹
‹
Channel Width:You can select 20MHz or 40MHz.
Control Sideband: You can select Upper or Lower.
Channel Number:You can select Wireless channel number.
Broadcast SSID: You can broadcast you SSID that every wireless device around
will see it, and connect to this wireless network. If you don’t want the SSID seen by
others, clear the check box.
‹
Data Rate: 1M bps~54Mbps.
‹
Associated Client: This page lists all the DHCP clients in the WLAN side.
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3.5.2
Advance Settings
3.5.2/Wireless/Advance Settings
‹
Fragment Threshold : This value indicates how much of the Access Point’s
resources are devoted to recovering packet errors. The value should remain at its
default setting of 2346. If you have decreased this value and experience high packet
error rates, you can increase it again, but it will decrease overall network performance.
The setting range is 256 ~ 2346.
‹
RTS Threshold : This value should remain at its default setting of 2347. Should
you encounter inconsistent data flow, only minor modifications are recommended.
The setting range is 0 ~ 2347.
‹
Beacon Interval: The Beacon Period is the amount of time between access point
beacon transmissions. Default value is generally 100, which means 100 beacons sent
every second. You can increase the beacon period and have lower overhead on the
network, but then roaming will likely suffer. It's better to leave this setting alone.
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3.5.3 Security
3.5.3/Wireless/Security
Select one security mode for your wireless device
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3.5.4
Access Control
3.5.4/Wireless/Control
There are two way to filter MAC address
‹
‹
Allow Listed: Allow only MAC address listed to access the wireless network.
Deny Listed: Deny MAC addresses listed from accessing the wireless network.
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3.5.5 Site Survey
You need Station list only work in AP Client mode. When you define your wireless device
as a client, check station list you can find all AP stations around your wireless receiving
area.
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3.6
NAT
3.6.1 Port Mapping
3.6.1/NAT/Port Mapping
The Port Mapping page does a port range mapping.
‹
‹
‹
‹
‹
Server IP
Set one of LAN client’s IP address to do port mapping.
Mapping Ports
Set the ports to mapping from WAN to LAN. You can enter multiple ports,
delimited by comma, or a range of port by dash. White space will be
neglect.
Type
Defines the protocol type, TCP/UDP/Both, of the public ports.
Comment
Let you put some notes to describe this entry.
Enabled
Enable this entry.
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3.6.2
ALG
3.6.2 /NAT/ALG
Some applications have to do Application Level Gateway (ALG) to monitor the
transaction or monitor the payload of the packet. This page let you setup them.
We use mnemonic nouns to describe the specific ALGs.
be enabled by the ALG processing.
The checked entries will
For the Non-Standard FTP Port, you have to set the port number to let the ALG
take it as an FTP session.
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3.7
Firewall
3.7.1 Denial-of-Service
3.7.1 /Firewall/Denial-of-Service
‹
Enable DoS Prevention
DoS Setup Denial of Service A "denial-of-service" (DoS) attack is
characterized by an explicit attempt by hackers to prevent legitimate users of
a service from using that service.
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3.8
3.8.1
Routing
Routing Table
圖 3.8.1/ Routing/Routing Table
‹
‹
‹
‹
‹
Destination LAN IP
This field indicates the destination IP of this route entry.
Subnet Mask
This field indicates the Subnet Mask of this route.
Gateway
If a packet’s destination IP address do “bit and” operation with the “Subnet
Mask” equals to the “Destination LAN IP”, the packet will send to the
Gateway.
Interface
This field indicates which network interface deal the connection.
Reload
Reload the routing information from the home gateway again.
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3.8.2 Static Routing
3.8.2/ Routing/Static Routing
‹
Destination LAN IP
This field indicates the destination IP of this route entry.
‹
Subnet Mask
This field indicates the Subnet Mask of this route.
‹
Gateway
If a packet’s destination IP address do “bit and” operation with the “Subnet
Mask” equals to the “Destination LAN IP”, the packet will send to the
Gateway.
‹
Add
Click this button to add this entry to routing table statically.
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3.9
3.9.1
Bandwidth&VLAN
Bandwidth Control
3.9.1/ Bandwidth&VLAN/Bandwidth Control
You can enable WAN or LAN bandwidth control in this configuring page, so far only the Upstream
Bandwidth is workable, you can fill in the limit value in Upstream Bandwidth to make limitation.
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3.9.2 VLAN
3.9.2/ Bandwidth&VLAN/VLAN
This function is used for special purpose of application. For more details about
VLAN, please refer to application note for VLAN, which is in the end of this
manual.
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3.10
Backup / Restore
3.10.1
Configurations
3.10.1 / Backup/Restore /Configurations
This page let you backup / Restore all of your configuration parameters on the VoIP
Router. It is very good idea to back up all of your VoIP Router configuration parameters
after install.
a.
To Backup, press Download setting backup file, and input the file name you
want and file location to save.
b.
To Restore, press the Browse button the select the backup configuration
parameters file to upload then press Restore . After you upload the file,
Press Saved modification to save your current configuration to Flash ROM
(Usually used to save currently WAN configuration).After save, please
remember to Reboot the VoIP Router to let the restored parameters take
effective.
*** Caution: Never power off the VoIP Router when during Restore configure or
upgrade VoIP module or System, it will cause permanent damage when power off
during writing Flash inside VoIP Router.
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3.10.2
VoIP module
3.10.2 / Backup/Restore /VoIP Module
This page displays the current firmware module version and let you backup /
Restore all of your VoIP firmware module on the VoIP Router. Please use this
page to update the VoIP module firmware if necessary.
a.
To Restore from local file, press the Browse button the select the VoIP
module file to upload then press Restore . After you upload the file, Press
Saved modification to save your current configuration to Flash ROM
(Usually used to save currently WAN configuration).After save, please
remember to Reboot the VoIP Router to let the restored parameters take
effective.
b.
To Restore from Upgrade server, please input the URL address of upgrade
server and press the APServer to link to upgrade server to get latest version
firmware and upgrade automatically.
*** Caution: Never power off the VoIP Router when during Restore configure or
upgrade VoIP module or System, it will cause permanent damage when power off
during writing Flash inside VoIP Router.
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3.11
Reboot
3.11 /Reoobt
Use the Reboot button on this page to reboot your VoIP Router, before you reboot,
please make sure you have to press the Saved modification to save your current
configuration to Flash ROM, otherwise all the change will be disappear after reboot.
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3.12
Save Modification
3.12 / Save Modification/
Most of the VoIP Router parameters will take effective after you modify, but it is just
temporary stored on RAM only, it will disappear after your reboot or power off the VoIP
Router, to save the parameters into Flash ROM and let it take effective forever, please
remember to press the Save Modification button after you modify the parameters.
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