AllwinTech H Series User manual
1
H-series VoIP Router
( H3000 / H1100 / H5xx / H6110 )
Web Administor Interface
User’s Manual
Version: 2.03
Date Issued by 2011/08/22
2
目錄
1. Web .............................................................................................................................................5
1.1 Login Web UI.......................................................................................................................5
2. VoIP Function..............................................................................................................................9
2.1 Portall Status .....................................................................................................................9
2.2 Auto Provision ................................................................................................................ 11
2.3 Line Configure ................................................................................................................12
2.3.1 Line Setting ..........................................................................................................12
2.3.2 Line Interface F
F
FX
X
XS
S
S/
/
/F
F
FX
X
XO
O
O.....................................................................................14
2.3.3 Tone Setting .........................................................................................................15
2.3.4 Tone Detect F
F
FX
X
XS
S
S/
/
/F
F
FX
X
XO
O
O........................................................................................17
2.3.5 Line Feature .........................................................................................................18
2.3.6 Line Voltage & Current........................................................................................22
2.3.7 Line Diagnostics .................................................................................................. 23
2.3.8 Line Impedance....................................................................................................24
2.3.9 PSTN Gain ...........................................................................................................25
2.3.10 Message Indicator ..............................................................................................26
2.4 Routing Setup..................................................................................................................27
2.4.1 VoIP Call Out .......................................................................................................27
2.4.2 VoIP Call In..........................................................................................................37
2.4.3 VoIP Call In IVR .................................................................................................. 43
2.4.4 Routing Profile.....................................................................................................46
2.4.5 Forwarding ........................................................................................................... 49
2.4.6 Authorization .......................................................................................................52
2.5 Register Server................................................................................................................53
2.5.1 Register Status......................................................................................................53
2.5.2 Register Server—SIP Protocol.............................................................................55
2.5.3 Register Server—H.323 Protocol ........................................................................ 57
2.6 Advance Setup ................................................................................................................59
2.6.1 NAT Traversal ......................................................................................................59
2.6.2 Listen Port ............................................................................................................ 60
2.6.3 VoIP Package........................................................................................................61
2.6.4 RTP Packet Summary ..........................................................................................63
2.6.5 Flash & Call waiting ............................................................................................64
2.6.6 Gain...................................................................................................................... 65
2.6.7 QoS.......................................................................................................................67
2.6.8 CDR .....................................................................................................................68
2.6.9 FoIP...................................................................................................................... 69
2.6.10 Prompt Voice & Beep......................................................................................... 70
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2.6.11 Call log ............................................................................................................... 72
2.7 Application......................................................................................................................73
2.7.1 Ping Test...............................................................................................................73
2.7.2 Centrex ................................................................................................................. 74
2.7.3 Telnet & SNMP....................................................................................................75
2.7.4 Call Timer ............................................................................................................ 77
3. System Setup.............................................................................................................................78
3.1 System............................................................................................................................. 78
3.1.1 System Status .......................................................................................................78
3.1.2 ystem Settings ......................................................................................................83
3.1.3 Date&Time...........................................................................................................84
3.1.4 Administrator Settings .........................................................................................85
3.2 WAN................................................................................................................................86
3.2.1 WAN Settings.......................................................................................................86
3.2.2 WAN Settings #2..................................................................................................90
3.2.3 DNS...................................................................................................................... 91
3.3 LAN ................................................................................................................................92
3.3.1 LAN Settings........................................................................................................92
3.3.2 DHCP Client List .................................................................................................94
3.4 Wireless...........................................................................................................................95
3.4.1 Basice Settings .....................................................................................................95
3.4.2 Advanced Settings................................................................................................97
3.4.3 Security ................................................................................................................99
3.4.4 Wireless Access Control ....................................................................................100
3.4.5 Site Survey .........................................................................................................101
3.5 NAT............................................................................................................................... 102
3.5.1 Virtual Server .....................................................................................................102
3.5.2 Port Mapping......................................................................................................103
3.5.3 ALG ...................................................................................................................104
3.6 Firewall .........................................................................................................................105
3.6.1 Denial-of-Service ...............................................................................................105
3.7 Routing..........................................................................................................................106
3.7.1 Routing Table ..................................................................................................... 106
3.7.2 Static Routing..................................................................................................... 107
3.8 Bandwidth&VLAN....................................................................................................... 108
3.8.1 Bandwidth Control.............................................................................................108
3.8.2 VLAN.................................................................................................................109
3.9 Backup / Restore ...........................................................................................................110
3.9.1 Configurations.................................................................................................... 110
3.9.2 VoIP module....................................................................................................... 111
5
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.
6
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:
zVoIP Function
Ports Status
Auto Provision
Line Configure
Line Setting
Line Interface F
F
FX
X
XS
S
S/
/
/F
F
FX
X
XO
O
O
Tone Setting
Tone Detect F
F
FX
X
XS
S
S/
/
/F
F
FX
X
XO
O
O
Line Feature
Line Voltage/Current
Line Diagnostics
Line Impedance
PSTN Gain (FXS+PSTN only)
Message Indicator
DAA Hybrid F
F
FX
X
XO
O
O
DAA Hybrid Config
DAA Hybrid Table
Line Rejection
Routing Setup
VoIP Call Out
VoIP Call In
VoIP Call In IVR
VoIP Routing Profile
Forwarding
Authorization
Register Server
Register Status
Server #1
Server #2
Server #3
Server #4
Advance Setup
NAT Traversal
Listen Port
7
VoIP Package
RTP Summary
Flash & Call waiting
Gain
QoS
CDR
FoIP
Prompt Voice & Beep
Call Log
Application
Ping Test
Centrex
Telnet & SNMP
Call Timer
zSystem Setup
System
System Status
System Setting
Date & Time
Administrator Settings
WAN
WAN Settings
WAN Setting #2 (H3000 H5xx series only)
DNS
LAN
LAN Setting
DHCP Client List
Wireless (H6110n only)
Basic Settings
Advanced Settings
Security
Access Control
Site Survey
NAT
Virtual Server
Port Mapping
ALG
8
Firewall
Denial-of-Service
Routing
Routing Table
Static Routing
Bandwidth & VLAN
Bandwidth Control
VLAN
Backup/Restore
Configurations
VoIP module
Reboot
Language/語言
Save Modification
9
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. Port: display line number.
d. Type: Line interface type (divided into 2 types):
FXO: connected to PSTN or PBX analog e extension lines.
FXS: connected to a telephone set or PBX Co. line.
e. Display Name: VoIP call user display name.
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f. 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.
g. Connected IP: display connedted remote side type for this call.
zPstnOut: Outbound call to analog line interface.
zPstnIn: Inbound call from analog line interface.
zrs: call via register server.
zIP: direct VoIP call by IP.
h. Caller ID: Caller ID.
i. Start Time: start time of VoIP call.
j. End Time: End time of VoIP call..
k. Talking Sec: Total VoIP talking time in seconds.
l. Dialed number:
zdial out numbers for out bound call.
zreceived dialed numbers from In bound call.
Ｂ. Error Message
Display the last error message of failure call.
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2.2 Auto 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.
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2.3 Line Configure
2.3.1 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 (divided into 2 types):
FXO : connected to PSTN lines or PBX analog extension lines.
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.
d. Line Number: Define line extension number, can be given to each line as the
extension number.
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.
13
h. Outbound: Enable or Disable Outbound (VoIP Call In) function. default is Enable.
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 (FXO: Ring in or FXS: off
hook phone).
For example, If we want the first line to hotline, each time user seizes the first line (FXO:
ring in, 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 Area Code Min Digits Max Digits Destination Strip Prefix Profile Delete
1 Hot_Line hl1 10.1.1.1 3 600 Delete
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2.3.2 Line Interface F
F
FX
X
XS
S
S/
//F
F
FX
X
XO
O
O
When using H3000-2 model, you can configure each port to use FXS or FXO.
2.3.2 /Line Configure/Line Interface/
This function must click / Save Modification /to re-start to take it effective.
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2.3.3 Tone Setting
2.3.3/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. Tone: Tone item index, Maximum 15 items.
b. 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.
c. Low freq: low frequency setting, set the lower frequency of tone.
d. High freq: high-frequency setting,set the higher frequency of tone.. Each tone can include
two frequencis if only one frequency need, set the High Freq. to zero.
e. T_ON_1, T_OFF_1, T_ON_2, T_OFF_2:
16
tone cadence interval time : includes dual-band beat interval to four intervals (see
below), the lowest is 30 milliseconds. (Unit in mS)
17
2.3.4 Tone Detect F
F
FX
X
XS
S
S/
//F
F
FX
X
XO
O
O
When the device has FXO interface, here define the buys tone and ring back tone detection
parameters.
2.3.4 /Line Configure/Line Detect/
Device will disconnect the VoIP call when the busy tone was detected.
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2.3.5 Line Feature
2.3.5.1 /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.
C. When using FXS to answer, decide to bring out the phone number by setting the
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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. CID signal detection method for FXO Interfacel:
Disable: turn off, do not detect the CID
DTMF: detection signal of the CID by DTMF
FSK Bell: detection signal of the CID by FSK Bell
FSK ETSI: detection signal of the CID by FSK ETSI
F. 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
G. 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.
H. Define how the FXS interface to ring the phone line when VoIP call in:
Free Random: Any unused available line.
Line number Priority: The 1st 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.
Rotation: 1st line ring first, then 2nd 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.
I. Polarity Detect Auto F
F
FX
X
XS
S
S/
/
/P
P
PS
S
ST
T
TN
N
N：
When enable, the device will detect the FXO interface polarity of the Tip and Ring endpoint,
20
and then automatically adjust the FXS interface on the Tip, Ring polarity to the same polarity
as FXO endpoint. This feature is used for some PBX don’t allow to change the polarity on its
Co. line during call making..
J. Define the default FXS phone port interface standby method, the following options
F
F
FX
X
XS
S
S/
/
/P
P
PS
S
ST
T
TN
N
N:
PSTN: Phone set on FXS port is standby on PSTN line connected on FXO interface.
When the device set accept a VoIP call, the phone will switch to the FXS interface,
and will be automatically on-hook the FXO line to avoid the call from PSTN. But if
enable the Calling Waiting function on / Routing Setup / VoIP Call In / , the FXO line
still can ring the phone on busy with call waiting tone.
If your device is not connected to the PSTN line, the phone set connected on FXS port will not
hear a dial tone and can not be used in this mode.
FXS: Standby the phone set on FXS interface. When make a PSTN call, the phone
will switching to the PSTN Interface through relay.
FXS PCM: Standby the phone set on FXS interface, when make a PSTN call, the
device will connect FXS and FXO by internal PCM bus rather than relay,.
K. Re-hook timeout for PSTN: (H3000-1 / H1111p)：
When FXS port standby on PSTN port, The PSTN port connected on FXO port will become
busy by off hook automatically if no enable call waiting function. Here define the interval time
for FXO re-hook the PSTN line.
L. Re-hook dela (H3000-1 / H1111p)：
Define the on hook second for re-hook duration.
M. Line OffHook Voltage (H3000-1 / H1111p)：
When standby on PSTN, define the voltage value for hook detection. When voltage value is
less then the define value, it will become off hook status.
This manual suits for next models
4
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