LevelOne VOI-8001 User Manual
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VOI-8001
VOI-8002
VOI-8003
8-Port H.323/SIP VoIP Gateway
User Manual
Ver. 1.01-0610
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Table of contents
CHAPTER 1. INTRODUCTION ...............................................................3
1.1 OVERVIEW .....................................................................................................................................................3
1.2 PACKAGE CONTENTS .....................................................................................................................................3
1.3 KEY FEATURE.................................................................................................................................................4
CHAPTER 2 GETTING STARTED ..........................................................5
2.1 FRONT PANEL ................................................................................................................................................5
CHAPTER 3 CONFIGURATION.............................................................7
3.1 CONNECTION .................................................................................................................................................8
3.2 SYSTEM STATUS ..........................................................................................................................................10
3.3 REGISTER SERVER ......................................................................................................................................11
3.4 VOIP CALL OUT ..........................................................................................................................................12
CHAPTER 4 WEB UI MANAGEMENT...................................................14
4.1 ACCESS TO WEB UI....................................................................................................................................14
4.2 WEB UI MANAGEMENT ...............................................................................................................................15
4.2.1 Overview..........................................................................................................................................15
4.2.3 Register Server Setting ...........................................................................................................54
4.2.4 Auto Provision function .........................................................................................................60
4.2.5 Advance Setup.............................................................................................................................62
4.2.6 Application ..................................................................................................................................76
4.2.8 Route Function(/System Setup) ...........................................................................................88
4.2.9 Bandwidth & VLAN .................................................................................................................. 113
4.2.10 Backup/Restore ..................................................................................................................... 114
4.2.11 Save Modification ................................................................................................................... 117
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Chapter 1. Introduction
1.1 Overview
The VOI-800x series VoIP Gateway is equipped with 8 standard phone ports, one
10/100BaseTX Fast Ethernet WAN port, and one 10/100BaseTX Fast Ethernet
LAN ports. With the integration of both voice and data, the offers ability to route
data information into network solution
VoIP Gateway has voice support that includes Quality of Service (QoS), voice
compression, echo cancellation, dynamic latency (jitter) buffers, silence
suppression, and comfort noise generation.
The VoIP Gateway is compatible with xDSL and Cable-modem Broadband
service providers with built-in support for DHCP Client, MAC Address Cloning,
PPPoE and multiple auto-provisioning methods.
1.2 Package Contents
- 8-Port VoIP Gateway
- CD User Manual
- Cat.5 Cable
- Power Adapter, 12VDC / 2A
Model No List
VOI-8001 8-Port FXS VoIP Gateway
VOI-8002 8-Port FXO VoIP Gateway
VOI-8003 4FXS+4FXO VoIP Gateway
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1.3 Key Feature
VoIP
Support 8 simultaneous VoIP calls
Support T.38 FAX relay
Support QoS(ToS) for VoIP
Compliant with H.323 / SIP VoIP standard protocol
Extensible by external IVR/CDR/Billing servers for value- added application
Support register up to 4 Gatekeepers / Proxy servers
Support worldwide off net call by ITSP service
Support Multiple dialing plan / Call hunting group
Adaptive Jitter Buffer function
Multiple call profile for adjust VAD, Audio CODEC, H.245
tunneling, DTMF In/Out band, FAX relay, Frame Size, Q.931
Fast start parameters
VoIP Gateway
Support static and dynamic IP from DHCP, PPPoE
Built-in DHCP Server
Support TCP/UDP Port Mapping (Local Server Mapping)
Support User-definable Static Routing Table
Support Network Access Rules (LAN-to-WAN & WAN-to-LAN)
Self-Protection against DoS Attacks
Dynamic DNS Support
WAN: 10/100Mbps RJ-45 connector, auto-sensing
LAN: 10/100Mbps Ethernet Switch. (Auto MDI-II/MDI-X)
VoIP: 8 port FXO/FXS ports
LED Indicators : Power, Status, Ready, WAN linking , LAN linking, Phone,
Line
Supported Protocol: UDP, TCP, Standard H.323,SIP, NAT,BOOTP, TFTP, FTP,
HTTP, TELNET, IEEE 802.3/ IEEE 802.3u
Selectable Coders: G.711, G.723.1, G.726, G.729A
DTMF / Call progress tone detection and generation
G.168 echo cancellation
1 Reset button for load factory default IP parameters setting
User friendly Web configure interface
Configuration/Upgrade Web and APS (Auto Provision Server)
Build-in watching dog for auto recovery
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Chapter 2 Getting Started
2.1 Front Panel
VOI-8001
VOI-8002
VOI-8003
(Ready) Flash: It means when the VoIP Gateway registration fail.
(Ready) Constant: It means when the VoIP Gateway registration successes.
(Status): This LED will flash quickly when the VoIP Gateway is either
performing a self test of booting up.
(Power): The LED light on when power on.
(WAN): The LED light on when WAN port was connected, flash quick when WAN
port transmission Voice or data. (Only for VOI-8002, VOI-8003)
(LAN): The LED light on when LAN port was connected, flash quick when LAN
port transmission Voice or data. (Only for VOI-8002, VOI-8003)
Note:
VOI-8001 displays 1~8 Phone LED only
VOI-8002 displays 1~8 Line LED only
VOI-8003 displays 1~4 Line and 5~8 Phone LED only
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2.2 Rear Panel
VOI-8001
VOI-8002
VOI-8003
DC12V: For the included power adapter. Be sure to use only the 12VDC/2A
power adapter included with the product. Using the wrong power adapter can
damage the product and void the warranty.
Reset: Clear all settings and restore them to the initial values present when
the device was purchased. After performing the reset, make sure to redefine
the IP settings for the device in the ‘Connection’.
WAN: A 10/100 dual-speed Ethernet port fitted with an RJ-45 connector
used to connect the to WAN device (usually a DSL / Cable Modem).
LAN : A 10/100 dual-speed Ethernet port fitted with an RJ-45 connector
used to connect the VoIP Gateway to a LAN device.
Phone [P]: Normal RJ-11 phone jacks used to connect analog telephones
and fax machines.
Line [L]: Normal RJ-11 phone jacks used to connect analog phone line or
PSTN (landline)
Note:
Do not place heavy objects on the VoIP Gateway. Placing the VoIP Gateway in a well ventilated
area is very important. Not doing so may cause damage to the unit.
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Chapter 3 Configuration
The default setting of DHCP Server inside VoIP Gateway is turn ON, So please set up
your PC TCP/IP network as “Get IP Automatically” from DHCP to get internal IP from
VoIP Gateway. By default, The VoIP Gateway will become the network gateway and
default IP is 192.168.22.1 and will assign your PC IP as 192.168.22.X.
Please go to “Control Panel”→”Network”. In the “Configure” page, choose the TCP/IP of
LAN card, and press “Properties” please choose “Obtain IP Address Automatically”
Launch your browser and open the Internal UI WAP page as 24H24H24Hhttp://192.168.22.1
The default User name is voip
The default Password is 1234
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3.1 Connection
Click “Connected Type” option below “System Setup\Wan” item:
Please select the type of Internet connection you have
and set up the VoIP Gateway to use the Dynamic IP
Address, Static IP Address, PPPoE, PPTP or L2TP
connection.
If your ISP has not given you an IP address, select
Dynamic IP Address (default). If you have been given
a specific IP address, select Specify an IP Address.
To use Static-IP ADSL connection, please select “Static IP Address” and enter
WAN IP settings.
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To use PPPoE ADSL connection, please select Yes in use “PPPoE” ADSL
service and enter the “username” and “password” in the PPPoE setup
section. Most of ADSL ISPs assign dynamic-IP settings to the VoIP Gateway
when using PPPoE. Please select Obtain IP Address Automatically(default
setting). You can leave the Primary and Secondary DNS IP settings in default.
The VoIP Gateway automatically obtains these settings from your ISP when
the PPPoE connection is successfully established.
Please remember to setting the Primary and Secondary DNS IPs, supplied
by your ISP
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3.2 System status
This page reveals the status of the VoIP Gateway including WAN, LAN and some
hardware/firmware information.
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3.3 Register Server
If this VoIP Gateway wants to use SIP Proxy or GateKeeper service to transfer
the VoIP call, you can input the server information here. The VoIP Gateway can
register to up to four servers simultaneously.
This page reveals the status of the server registration information.
Here is the server configuration page, please ask the information form your
ITSP.
Remark: For Notify remark for this rule. Please use UNDERLINE to
replace the SPACE due to HTTP protocol limitation.
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3.4 VoIP Call Out
User key in the phone number through phone set dial pad, then VoIP Gateway
translate the phone number by the routing table setting here to destination IP & dial
out number then Call out via network protocol
Remark: For Notify remark for this rule. Please use UNDERLINE to
replace the SPACE due to HTTP protocol limitation.
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 Gateway will check the rule list from top to bottom one by one, any rule
item matched with the prefix digits that user key in 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.
Min Digits: The length of the dialed number should not less than this
digits. For example, if the field is entered into ‘3’, the length of the dial
number should be 3 digits at least.
Max Digits: The length of the dialed number should not more than this
digits. For example, if the field is entered into ‘10’, the length of the dial
number should be 3 digits at most.
a. IP Address: Define the destination IP for call out number fit this rule, user
can input below format:
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IP address, for example: 168.56.9.22
URL, route via URL. For example: www.inphonex.com .This VoIP
Gateway can setup to register to DDNS service (/System Setup
/Advanced/ Dynamic DNS/) to let user call out to another VoIP
Gateway with dynamic IP by URL.
rsn , route via server, it will get the destination IP by server setting
(/VoIP Setup/Register server/) in advance. For example: rs1 for
server 1. rs2 for server 2. rs for all the server available ( search
sequence: rs1 > rs2 > rs3 > rs4). rs3_2_1 will try rs3 first, then rs2,
then rs1.
IP address, for example: 168.56.9.22
All the setting above can be added by port number, for examples:
168.56.9.22:8495 will call to 8495 port.
Strip: the number of digits will be ignored by user input. For example, if
user key in the number is 886212345678 and the STRIPE field is setting to
4, the first 4 digits 8862 will be truncated and actually call out number will
be 12345678.
Prefix: The numbers will be added on the prefix of user key in number.
For examples, if user key in the number is 12345678 and the PREFIX field is
setting to 0028862, the actually call out number will be 002886212345678.
Another example, if user key in the number is 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). This example is especially for
speed dial function.
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 “Save Modification”.
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Chapter 4 Web UI Management
4.1 Access to Web UI
The VoIP Gateway provide user friendly Web interface to let you configure your VoIP
Gateway function
The default setting of DHCP Server inside VoIP Gateway is turn ON, So please set up
your PC TCP/IP network as “Get IP Automatically” from DHCP to get internal IP from
VoIP Gateway. By default, The VoIP Gateway will become the network gateway and
default IP is 192.168.22.1 and will assign your PC IP as 192.168.22.X.
Please go to “Control Panel”→”Network”. In the “Configure” page, choose the TCP/IP of
LAN card, and press “Properties” please choose “Obtain IP Address Automatically”
Launch your browser and open the VoIP Gateway Internal UI WAP page as
http://192.168.22.1
The default User name is voip
The default Password is 1234
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4.2 Web UI Management
The VoIP Gateway provide user friendly Web interface to let you configure your VoIP
Gateway function and VoIP function. There are a help on line content within each
setting page. Please press Help hyperlink to view the on line help. There are 3 main
functions for web, VoIP, System Setup (VoIP Gateway) & System maintenance. Each
function is setup by the function below:
4.2.1 Overview
Route function
Connection (Setting WAN connecting)
LAN Setting
Firewall Basic setup
Networks System Status Display
Dynamic DNS Setting
DHCP Server Setting
Static Routing Setting
Local Server Setting
DMZ Setting
Bandwidth&VLAN
VoIP function
Port Status Display
Line Configure Setting
Line Setting
Tone Setting
VoIP Call Out Routing Table Setting
VoIP Call In Routing Table Setting
VoIP Call In IVR
VoIP Routing Profile Setting
VoIP Forwarding Profile Setting
Authorization
Register Status
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System Maintenance function
Configurations Backup/Restore
VoIP Module Backup/Restore
Reboot System
Save Modification
Gateway Manual overview
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4.2.2 VoIP Function
4.4.2.1 VoIP Setup/ Port Status/
This page will display the current and last time VoIP call status & result.
a. The PC time : will show the date & time that your connected PC now.
b. The VoIP Gateway time : will show the date & time on this VoIP Gateway,
the date& time may get from SNTP server or setting from your PC. You may
set the SNTP server from /System Setup/Administrator/Date & Time/.
A. Ports Message
a. Port: display the port number, e.g. 1 or 2.
b. Type: Telephone interface type:
FXO: (DAA interface) for connect to telephone line or PBX extension
line.
FXS: (SLIC interface) for connect to regulate phone set.
c. Display Name: display the remote party name of this VoIP call.
d. Status: Current status of this port.
Idle: Standby for make a phone call.
Signal: Waiting for DTMF press or VoIP protocol connecting.
In: There is a phone call made from phone port and call out to Network
by VoIP.
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Out: There is a phone call made from Network VoIP and pick up by
phone set.
e. Connected IP: The remotely party IP of this VoIP call.
f. Caller ID: Caller ID received from telephone line port.
g. Start Time: Date & time of this VoIP call begin on this port.
h. End Time: Date & Time of last VoIP call End on this port.
i. Talking Sec: Total talked seconds of last VoIP call on this port.
j. Dialed number:
On the VoIP call out (line status display “In”). This will display the real
dial out number for VoIP call.
On the VoIP call in (line status display “Out”). This will display the
number will dial out to phone line.
Release by: This will display the reason of this call termination.
B.Error Message
For some reason,(ex. All lines of this VoIP Gateway are busy) here will display
the failure information about the last failure VoIP Call.
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4.2.2.2 VoIP/Line Configure/ Line Setting
/VoIP Setup/Line Configure/Line Setting/
This page will setup the phone line information each port.
a. Port: display the port number, e.g. 1 or 2.
b. Interface: Telephone interface type:
FXO: for connect to telephone line or PBX extension line.
FXS: for connect to regulate phone set
a. Name: Line name for this port. This will send and display on the remote
side during VoIP call
b. Line number: Telephone number assigned to this line.
c. TxGain: Transmitter Gain. This will adjust the speaker volume of local
phone set. The adjust range is from +3 to -13dB. Higher value will cause
louder sound come from local phone set.
d. RxGain: Receiver Gain. This will adjust the microphone volume of local
phone set. The adjust range is from -3 to +13dB. Higher value will increase
amplifier the sound get from local phone set.
e. Inbound: Enable or disable the VoIP call to Internet. Disable the inbound
option will not allow any call made from phone set to Internet.
f. Outbound: Enable or disable the VoIP call from Internet. Disable the
Outbound option will not allow any call made from Internet to phone set.
g. Hotline: When Enable, it will allow you to make a VoIP call without Press
any number. That mean it will direct call out by VoIP when you off hook the
phone of this line.
For example, if you want line 1 to become a hot line for VoIP call, every time
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when you off hook the phone connected to the line 1, it will directly call to
another VoIP gateway location at 168.56.09.22 and dial 601. You can enable the
line 1 as hot line, and add a routing rule on the routing table on /VoIP
Setup/Routing Setup/VoIP Call Out/ to assign the AREA CODE to hl1 to
handle the VoIP Gateway rule for hot line function. And please also remember to
Strip 3 digits to stripe the “hl1” symbol and remember add real phone number
you want to dial on Prefix. In this case, the setting example on call out routing
(/VoIP Setup/Routing Setup/VoIP Call Out/) for hot line application is as
below:
Index
1 Hot_Line_Call Hl1 10.1.1.1 3 601 25H25H25HDelete
Remark Area Code IP Address Strip
Prefix Profile Delete
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4.2.2.3 Line configure/ Tone Setting
/VoIP Setup/Line Configure/ Tone Setting
A.Call Progress Tone
This page defines the tones generated to the phone connected to the phone port.
The cadence of CPT is been defined here also. All lines use same tone
parameters. After modify the tone parameters, you must 26H26H26Hsave modify then
27H27H27HReboot to let the modified parameters work.
Detect Voice Busy Cycle: Use the parameters to automatic detect cadence
busy tone. When detected a voice cadence repeat over the number setting
in sequence, the VoIP Gateway will treat it like busy tone and disconnect
automatically. Please do not set this parameter less than 5 to avoid
unexpected erroneous disconnect.
B.Tone define Table
You can set up to 15 tones set for generation. For the generation, the first entry
will be used. The call progress tones, ranging from 300 Hz to 2000 Hz. Tone:
Maximum 15 tones can be defined.
a. Type:
Dial: Define the generated dial tone.
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Busy: Define the busy tone for generate.
Ring: Define the ring back tone for generate
b. Low freq: Lower frequency for defined tone
c. High freq: Higher frequency for defined tone. Each tone can define two
frequencies, if only one frequency needed, please leave High Frequency to
0.
d. T_ON_1, T_OFF_1, T_ON_2, T_OFF_2:
The cadence pattern of up to four intervals for each dual-frequency.
Minimum Cadence value is 30msec.
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4.2.2.4 Line configure/ Line Feature
/VoIP Setup/Line Configure/ Line Feature
This page defines the feature on the phone port of the VoIP Gateway.
A. Dial Pause signal length(as ,)[100~3000] ms:
Define the pause time (ms) of the “,” on the /Routing Setting/VoIP Call
Out/. This pause time is usually for time delay when connect to PBX and
used for seize the CO line. The default pause time is 1000ms. The input
range is between 100 to 3000 ms. User can use more then one “,” to get
longer delay time.
B. Loop Current Drop & Polarity Reversal Generate：
Define the signal generated on local side when remote side disconnects:
Disable: Disable the Loop current Drop and Polarity Reversal
Generate signal, only generate busy tone.
Polarity Reversal-> Enable: Enable FXS interface to generate the
Polarity Reversal Signal.
Current Drop-> 1 S: Enable FXS interface to generate one second
Current Drop signal.
Current Drop-> 2 S: Enable FXS interface to generate two seconds
Current Drop signal.
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Current Drop-> 3 S: Enable FXS interface to generate three seconds
Current Drop signal.
C. Called Number Relay on FXS ：
Define when use the FXS interface to outbound call, resend or Drop out the
dialed number.
Drop out: Do not send the dialed number. When use the FXS port
direct connect to phone set for outbound call, please enable the
“Drop out” function to avoid hear the unnecessary dialed number
when answer the phone call.
Resend: Resend the dialed number. When use the FXS port to
connect to PBX line for outbound call, please enable the “Resend”
function to redial the destination number by DTMF, this will cause the
PBX transfer to the call to the final user.
D. Caller ID Generate type:
Define the Caller ID (CID) signal generate format:
Disable: Disable, do not send CID signal.
DTMF: Send CID signal by DTMF format.
FSK Bell: Send CID signal by FSK Bell format.
FSK ETSI: Send CID signal by FSK ETSI format.
E. Caller ID Detect Mode：
Define the CID detect format of FXO interface:
Disable: Disable, Do not detect any CID signal
DTMF: Enable detect CID signal by DTMF format.
FSK Bell: Enable detect CID signal by FSK Bell: format.
FSK ETSI: Enable detect CID signal by FSK ETSI: format.
F. When VoIP call out, send ANI by:
Define when VoIP call out, use the below number as the Caller ID (ANI):
Register Number: Use the gateway register number as ANI.
Line Number: Use the line number setting on the /VoIP
Setup/Line Configure/Line Setting/ as ANI.
PSTN CID: Use the received Caller ID number from PSTN line as
ANI.
G. FXS Ring Method：
Define how the FXS interface to ring the phone line when VoIP call in:
Free Random: Any unused available line.
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Line number Priority: The 1
st
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: 1
st
line ring first, then 2
nd
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.
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4.2.2.5 Line configure/ Line Polarity
/VoIP Setup/Line Configure/ Line Polarity
This page defines the Polarity on the phone port of the VoIP Gateway.
If use the normal phone set to connect gateway, please select “Normal”.
If use PBX or special PSTN line (support polarity invert), then please select
“Invert”.
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/).
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4.2.2.6 Routing Setup/ VoIP Call Out Setting
/VoIP Setup/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 Gateway 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 Gateway, 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 Gateway 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
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translate for call out rule without needed to press # key.
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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 Gateway wait for the defined seconds and
there is no any digits press, the VoIP Gateway 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 Gateway 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 by the PSTN line.
Index Remark
1
2
Normal
rule
Backup
rule
<The example that use “v” prefixes for reroute the call out>
Timeout for Re-entry route:
Area
Code
Min
Digits
Max
Digits
Destination Strip Prefix Profile Delete
second.
8 8 8 168.11.22.33 28H28H28HDelete
v8 PSTN 29H29H29HDelete
When user enable the hot line function on /VoIP Setup30H30H30H/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
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b. Remark: Remark for this routing rule. Please use UNDERLINE to replace
the SPACE due to HTTP protocol limitation.
c. 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
Gateway 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 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 Gateway can service the hot line direct call.
d. 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 Gateway will wait time
defined on OtherDigitTime then go to translate for call out rule without
needed to press # key.
e. 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.
f. 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 please add sip: before ip address, for example sip:168.56.9.22
2. for h323 please add h323: before ip address , for example
h323:168.56.9.22
URL, route via URL. For example: sip.fwd.com .This VoIP Gateway can
setup to register to DDNS service (/System
Setup/Advanced/Dynamic DNS/) to let user call out to another VoIP
Gateway 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
3
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.
srn, rsn: same as gkn , basically, it is used for SIP register
server.
PSTN: route this call via PSTN line interface. This is usually used
for the backup route for the rule setting on /Routing setup /VoIP
Call out/ with “v” prefix.
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.
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
f. Strip: the number of digits will be ignored by user input. For example, if
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