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DVG-4032S
VOIP Gateway
User Manual
Version 1.0
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Contents
1. Introduction....................................................................................................1
Product Overview.......................................................................................................................................... 1
Product Features........................................................................................................................................... 2
Hardware Description.................................................................................................................................... 3
2. Installation and Applications........................................................................4
Network Interface .......................................................................................................................................... 4
Gateway Assigned with a Public IP Address...................................................................................... 4
Gateway in a NAT network................................................................................................................. 5
Telephone Interface Description....................................................................................................................6
Example for Phone Ports: .................................................................................................................. 6
Example for Line Ports: ...................................................................................................................... 6
3. Setting the Gateway through IVR.................................................................7
IVR (Interactive Voice Response) .................................................................................................................7
Recorded Voice File ...........................................................................................................................8
IVR Functions Table: .......................................................................................................................... 9
IP Configuration Settings—Setting IP Configuration of WAN Port................................................... 11
4. Setting a Gateway with WEB Browser .......................................................14
Network Settings ......................................................................................................................................... 14
QoS Settings ............................................................................................................................................... 19
NAT/DDNS .................................................................................................................................................. 20
Telephony Settings...................................................................................................................................... 22
SIP............................................................................................................................................................... 25
Calling Features .......................................................................................................................................... 29
Advanced Options....................................................................................................................................... 31
Line Settings..................................................................................................................................... 33
Voice................................................................................................................................................. 33
FAX................................................................................................................................................... 34
Digit Map ..................................................................................................................................................... 35
Local Phone Book .......................................................................................................................................35
Speed Dial...................................................................................................................................................36
Caller Filter .................................................................................................................................................. 36
ACL for Management ..................................................................................................................................36
CDR Settings............................................................................................................................................... 37
Language ....................................................................................................................................................37
Transit Call Control...................................................................................................................................... 37
Long-Distance Control Table .......................................................................................................................39
Long Distance Exception Table ...................................................................................................................39
CPT/Cadence Settings................................................................................................................................ 39
Busy Tone Cadence Measurement.................................................................................................. 39
CPT parameters Table ..................................................................................................................... 40
System Information .....................................................................................................................................41
RTP Packet Summary................................................................................................................................. 41
STUN Inquiry............................................................................................................................................... 42
Ping Test ..................................................................................................................................................... 42
SNMP ..........................................................................................................................................................42
NTP ............................................................................................................................................................. 43
Backup/Restore........................................................................................................................................... 43
System Operations...................................................................................................................................... 44
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Software Upgrade ....................................................................................................................................... 44
Logout .........................................................................................................................................................44
5. Coding Principle..........................................................................................46
Instruction.................................................................................................................................................... 46
Dialed Number Processing Flow................................................................................................................. 46
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1. Introduction
Product Overview
The stand-alone VoIP Gateway carries both voice and facsimile over the IP network. It supports
SIP industry standard call control protocol to be compatible with free registration services or VoIP
service providers’ systems. It works in two different modes: UA (User Agent) or Server. As a
standard user agent, it is compatible to all well-known Soft Switches and SIP proxy servers.
While running the optional server software, the gateway can be configured to establish a private
VoIP network over the Internet without a 3
The gateway can be seamlessly integrated to existing network by connecting to a phone set,
PBX, key telephone system, fax machine or PSTN line. With only a broadband connection such
as ADSL bridge/router, Cable Modem or leased line router, it allows you to gain access to voice
and fax services over IP in order to reduce the cost of international and long distance calls.
With the support of DDNS, it makes the gateway reachable by its domain name where the ISP
dynamically assigns the IP address. It helps users to host a web site or mail server in a PPPoE
or DHCP network. By enabling the CDR function & setting up a simple server, administrators are
allowed to log and view all call records such as call duration, time and date of calls and latency,
etc.
The gateway can be assigned with fixed IP address or by DHCP, PPPoE. It adopts the G.711,
G.726, G.729A or G.723.1 voice compression format to save the network bandwidth while
providing real-time and toll quality voice.
rd
party SIP Proxy Server.
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Product Features
● SIP (RFC 3261) compliant
● Optional server enables small businesses to build up private VoIP network (SIP model)
● QoS support guarantees voice bandwidth in a busy network
● Supports IP TOS (Type Of Service)
● T.30 (G III) / Real Time T.38 / Secured T.38 Fax Relay
● Feasible for Fixed IP address or dynamic IP address network ( PPPoE / DHCP client, support
DDNS)
● Configurable Hot Line feature
● Supports IP-to-PSTN / PSTN-to-IP applications
● NAT traversal - STUN and UPnP (optional)
● Pass through NAT
● Call Detailed Records (CDR)
● Web-based firmware upgrade
● Caller ID Delivery
● Easy Configuration by IVR and Web-based GUI
● Greeting message
● Echo Cancellation: G.168 compliant
● Voice Activity Detection (VAD) & Comfort Noise Generation (CNG)
● Dialing: DTMF, PULSE (optional)
● Signaling Protocol: Loop Start
● Adaptive Jitter Buffer and Programmable Gain Control
CALL features (optional)
● Call hold
● Call waiting
● Call forward
- Unconditional (follow me)
- Busy forward
- No answer forward
● Call transfer
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Hardware Description
Front Panel
Rear Panel
Power Indicator: Green light indicates a normal power supply.
Run Indicator: Blinking green light indicates normal operation.
Alarm Indicator: When the system starts up, the red light will blink. It also indicates the
gateway’s abnormal operation.
Phone 1 – Phone 16 stands for Port 1 – Port 16: Connect to your analog telephone,
Line 1 – Line 16 stands for Port 17 – Port 32: Connected to your original telephone line
on the wall jack with RJ-11 cable.
WAN stands for the WAN Port Indicator.
LAN stands for the LAN Port Indicator.
When starting up the system, the Alarm, Run, and Power indicators will light up.
After about 40 seconds, the Alarm indicator will go off, the Run indicator will blink
in green, and the Power indicator will stay green under normal operational
conditions. If the Alarm indicator continues to blink, it means the system is
currently communicating with ISP and has yet to obtain an IP address.
When the WAN is connected, the WAN indicator will light up in green and if data is
being transmitted over the Internet, the indicator blinks in green and orange.
NOTE: Do not connect Phone ports to each other. Also, do not connect any Phone ports
directly to a PSTN line or internal PBX. If any of these are done, your DVG-4032S may be
damaged.
Restore to factory default: (IP address, User’s. Na me and Password)
(1) Pull off the power plug.
(2) Press reset (do not let go of the reset button).
(3) Plug the plug back into the socket (do not let go of the reset button).
(4) Let go of the reset button after 6 seconds. Factory settings will be restored.
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2. Installation and Applications
Network Interface
The network interface is divided into 3 basic modes as described below:
Gateway can be assigned with a Public IP Address
Gateway can be built under the existing NAT
Gateway Assigned with a Public IP Address
The gateway will have a Public IP address for Internet connection regardless of whether it is a
static IP address, DHCP (using a Cable Modem), or PPPoE (Dialup / ADSL).
Gateway IP Settings
NAT/STUN Settings Unnecessary (Disabled)
Need to set up as static IP,
DHCP, or PPPoE
DDNS Settings Unnecessary (Disabled)
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Gateway in a NAT network
The gateway uses a virtual IP address and the IP sharing function of other systems to connect to
the Internet.
LAN IP address of IP sharing
Gateway IP Settings
NAT /STUN Settings
Please avoid IP address 192.168.0.1-192.168.8.254 (You may need
to change the settings of IP sharing or change SIP series Gateway
LAN Port IP address)
Set as static IP address, and assign the LAN IP address of the IP
sharing to the Default Gateway.
Enable
If the WAN of the IP sharing device has static IP address,
then the NAT IP address is set as the Public IP address of
the IP sharing.
If the WAN of the IP sharing device uses a dynamic IP
address, then it has to comply with the DDNS settings.
When suing NAT, you must enter the URL (Uniform
Resource Locator) that is registered to the DDNS server.
DDNS Settings
The WAN of the IP
sharing device has a
static IP address.
The WAN of the IP
sharing device has a
dynamic IP address.
Disabled
Enabled: enter the registered URL (Uniform
Resource Locator) into the network settings
-> under NAT
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Telephone Interface Description
Example for Phone Ports:
DVG-4032S connecting directly to phone sets
After connecting telephone sets to P1-P8, users can make direct calls, (P1-P8 are FXS
interfaces). Each set acts as an independent extension line.
Integrating the DVG-4032S with PBX
P1-P16 is FXS interfaces, and some of them can be connected to telephone sets for direct calls.
Others can be connected to the PBX so other extension lines can make VoIP calls.
Example for Line Ports:
DVG-4032S connecting directly to the Telephone Line of a PSTN
P9-P16 is FXO interfaces and can all be connected to a PSTN to serve as a bridge between the
PSTN and other VoIP telephones. The system also allows a call to be made from a traditional
telephone line to connect with a user behind the Gateway.
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3. Setting the Gateway through IVR
VoIP transmits voice data (packet) via the Internet to achieve telecommunications. This means
that the telecommunication quality is closely related to the whole network environment. If any
one of the telecommunicating parties has insufficient bandwidth or frequent packet loss, the
telecommunication quality will be poor. Therefore, an excellent telecommunication can only be
created when Gateway is connected to the Internet and when network environment is stable.
Preparation
Install the Gateway according to instructions. Connect the power supply, telephone set,
telephone cable, and network cable properly as described in Chapter 2.
If a static IP is used, confirm the desired IP settings of the WAN Port (IP address, Subnet
Mask, and Default gateway). Please contact your local Internet Service Provider (ISP) if
you have any questions.
If using dialup ADSL (PPPoE) for network connection, confirm the dialup account
number and password.
If users wish to build Gateway under the NAT, Gateway WAN Port IP address and LAN
Port should not use the same range. This is to avoid phone failures.
Basic Settings of a Gateway
IP Settings— Connecting Gateway to the Internet.
Gateway provides two setting modes:
1. Telephone IVR Setting Mode
2. Browser Setting Mode
The IVR provides basic query and setting functions, while the browser provides a full setting
function.
IVR (Interactive Voice Response)
Gateway provides convenient IVR functions. Users only need to pick up a handset and enter the
function code for the query and setting without using a PC.
NOTE: After finishing the settings, make sure the new settings are saved. This is so that
the new settings will take effect after the system is restarted.
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Instructions
FXS Port: Connected to telephones. To enter IVR mode, enter “ * * password #” after
hearing the dial tone. When you hear a second dial tone, the system is in IVR mode,
enter the function code. (Please refer to the Advanced Settings on page 35 for these
codes)
FXO Port: to use IVR functions, dial the phone number of FXO Port using an external
line. You will hear the instruction “enter value”, and then enter a PIN number. The factory
default code is blank. Enter “**#” as above. You are now in IVR setting mode.
Example: The factory default code is blank. Enter “**#”
. You are now in IVR setting mode, enter
the desired code. E.g.: if the code is 1234, then enter “**1234#”. Or If your password is abc123
then you access IVR by pressing ***414243010203#
Once the first setting or query has been completed, you will hear a dial tone. Then use
the same procedure to make a second query or setting. To exit IVR mode, simply hang
up the phone.
Example: enter “**#”
(You are now in IVR mode) enter 101 (to query IP address) the
system responds with an IP address you can continue with more settings or queries: enter 111
(to set IP address) enter 192*168*1*2 (IP number).
Save Settings
After entering IVR mode, dial 509
(Save Settings). Wait for about 3 seconds and after hearing a
confirmation tone “1”, hang up the phone. Please reboot the Gateway to enable the new
settings.
To inquire about current Gateway’s WAN Port IP address
After entering IVR mode, dial 101
. The system will repeat the current WAN Port IP address.
If the system does not repeat the IP address, it indicates that the Gateway is not currently
connected to the Internet. Please check if the cable connection, account number, and password
are correct.
Recorded Voice File
The gateway allows users to record their incoming call greeting messages, when calling
via FXO.
After entering IVR mode, dial 132. After hearing “Enter value”, record the incoming call
greeting message. To end recording, simply hang up.
After recording, to listen to the recorded message, press 131. Press 133 to save the
message.
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IVR Functions Table:
Function Code Description Example
111/101 WAN Port IP address Set/Query
112/102 WAN Port Subnet Mask Set/Query
113/103 WAN Port Default Gateway Set/Query
114/104
Current Network IP Access Set/Query (1: Static IP,
2.DHCP, 3.PPPoE)
115/105 DNS IP address Set/Query
116/106 Phone books manager IP address Set/Query
117/107
Set/Query whether or not to use Public Telephone
Book (0: Disable 1:Enable)
199/099
Set/Query whether or not this Gateway acts as the
phone books manager (0: Disable 1: Enable)
066 Querying the connection to Phone books manager
118 Restart
121 Setting PPPoE Account
122 Setting PPPoE Password
123 Setting NAT IP address
124 Uses NAT (0: Disable 1: Enable)
151/141
Register to Proxy Server Set/Query (0: Disable 1:
Enable)
152/142 Proxy Server IP address Set/Query
153/143 Proxy Server Port Set/Query
125 Set Proxy Server account
126 Set Proxy Server password
154/144 Uses STUN Set/Query (0: Disable 1: Enable)
155/145 STUN IP address Set/Query
156/146 STUN Port Set/Query
311/301
312/302
131/132
133 Saving greeting message
211/201 Set/Query International Prefix code
212/202 Set/Query Country Code Setting country code, e.g. 886
213/203
Set/Query Area Prefix Code (Long-Distance Prefix
Code)
214/204 Set/Query Area Code eg. “2” for Node B area.
Use in conjunction with function
code 114, select 1 for a Static IP
function.
Must use 116/106, 117/107 in
conjunction with each other.
Use in conjunction with function
code 114, select 3 for a PPPoE
function
Must use 123 and 124 in
conjunction with each other.
Prefix dialed before making an
international call e.g. 002 and 005.
Prefix dialed before making a
long-distance call e.g. 0.
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Function Code Description Example
215/205
Set/Query Gateway Telephone Number
(Representative Number)
216/206 Set/Query the extension number of Line 1.
217/207
109 Restoring factory default IP address configuration A static IP address for WAN Port
IP：192.168.1.2
Mask：255.255.255.0
Gateway：192.168.1.254
409 Restoring factory default settings
509 Save settings
900 Setting IVR and the language used on the Web GUI
(1: English, 2: Traditional Chinese, 3: Simplified
Chinese)
209 Soft Upgrade
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IP Configuration Settings—Setting IP Configuration of WAN Port
Static IP Settings
NOTE: Complete static IP settings should include a static IP (Option 1 under114), IP
address (111
Internet Service Provider (ISP) if you have any questions.
Function Command
Select a Static IP
IP address Settings
Subnet Mask Settings
Default Gateway Settings
Save Settings and Restart
Dynamic IP (DHCP) Settings
After entering IVR mode, dial 114
After hearing “Enter value”, dial 2 (to select DHCP).
Saving settings –press 509
restarted, press 101
If the system does not repeat the IP address, it indicates Gateway has not been properl y
connected to the Internet. Please check the cable connection.
ADSL PPPoE Settings
), Subnet Mask (112), and Default Gateway (113). Please contact your local
After entering IVR mode, dial 114.
After hearing “Enter value”, dial 1 (to select static IP)
After entering IVR mode, dial 111. After hearing “Enter value”, enter your IP
address, followed by “#”.
Example: If the IP address is 192.168.1.200, dial 192*168*1*200#.
After entering IVR mode, dial 112. After hearing “Enter value”, enter your
subnet mask, followed by “#”.
Example: If the mask value is 255.255.255.0, dial 255*255*255*0#.
After entering IVR mode, dial 113. After hearing “Enter value”, enter your
default gateway’s IP address, followed by “#”.
Example: If the Default Gateway is 192.168.1.254, dial 192*168*1*254#.
To save settings, dial 509
(Save Settings). The system will save the
settings. Please restart the system. Wait for about 40 seconds for the
system to restart, and then enter 101
to check if the IP address is retained.
If the IP address is not repeated, it indicates that Gateway has not been
properly connected, please check if the cable connection, account, or
password are correct.
.
(Save Settings). Please restart the system. After the system is
to check if the IP address is retained.
NOTE: Complete PPPoE settings should include: Select PPPoE (Option 3 of 114), PPPoE
account (121
) and PPPoE password (122).
Please contact your local Internet Service Provider (ISP) if you have any questions.
Select a PPPoE
After entering IVR mode, dial 114
After hearing “Enter value”, dial 3 (to select PPPoE).
.
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PPPoE Account Settings
After entering IVR mode, dial 121
.
After hearing “Enter value”, enter the account number, followed by ”#”.
Example: If the account is “84943122 @ hinet.net”, please enter 08 04 09 04 03 01 02 02 71 48 49 54 45
60 72 54 45 60 #.
Please note that it is necessary to enter two digits for each character/number; for
example, enter “01” for “1” and “11” for “A”.
PPPoE Password Setting
After entering IVR mode, dial 122
after hearing “enter value” followed by “#”.
Example: If the password is “3ttixike”, please enter “03 60 60 49 64 49 51 45#”.
Save Settings and Restart
To save settings, dial 509
(Save Settings). The system will save the settings. Please restart the
system. Wait for about 40 seconds for the system to restart, and then enter 101 to check if the IP
address is retained. If the IP address is not repeated, it indicates that Gateway has not been
properly connected, please check if the cable connection, account, or password are correct.
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PPPoE Character Conversion Table
Number Input Key
Upper Case
Letter
Input Key
Lower Case
Letter
Input Key Symbol Input Key
0 00 A 11 a 41 @ 71
1 01 B 12 b 42 • 72
2 02 C 13 c 43 ! 73
3 03 D 14 d 44 " 74
4 04 E 15 e 45 $ 75
5 05 F 16 f 46 % 76
6 06 G 17 g 47 & 77
7 07 H 18 h 48 ' 78
8 08 I 19 i 49 ( 79
9 09 J 20 j 50 ) 80
K 21 k 51 + 81
L 22 l 52 , 82
M 23 m 53 - 83
N 24 n 54 / 84
O 25 o 55 : 85
P 26 p 56 ; 86
Q 27 q 57 < 87
R 28 r 58 = 88
S 29 s 59 > 89
T 30 t 60 ? 90
U 31 u 61 [ 91
V 32 v 62 \ 92
W 33 w 63 ] 93
X 34 x 64 ^ 94
Y 35 y 65 _ 95
Z 36 z 66 { 96
| 97
} 98
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4. Setting a Gateway with WEB Browser
The gateway allows users to make settings using a web browser. After opening a browser, enter
Gateway’s IP address as the website address in order to enter the Web configuration screen as
shown in the following diagram.
The factory default WAN IP address for Gateway is 192. 168. 1. 2.You can also enter ”101” from the
handset to inquire about the current WAN Port IP address. The factory default LAN Port IP address is
192.168.8.254.
Instructions
Open an Internet browser.
Enter gateway’s WAN Port IP address in the website address area (If the PC is
connected to the LAN Port, enter the LAN Port IP address. The default is 192.168.8.254)
The following registration screen will appear (The factory default settings for Login ID
and Password are set to blank).
After completing and confirming the settings, some of the settings will take effect
immediately. But network related settings would take effect after the gateway is
restarted. Please go to System Operation to save the settings before restarting the
system.
To avoid several people simultaneously configuring the web and causing problems to users,
please enter the correct Login ID and password. If a user logs into the system, other users from
different IP addresses cannot login at the same time. Please remember to logout or restart the
system if not using the web configuration function.
Network Settings
The network settings are used to set the gateway’s communication ports, IP configurations, and
Phone Books Manager IP etc.
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Current WAN IP Address: The IP address of WAN port.
Listen Port UDP: It is not necessary to change the protocol of the communication port
used by DVG-3016S.
RTP Starting Port UDP: The initial value of port number for transmitting voice data
among Gateway(s). Each line requires 2 ports. It is not necessary to change these.
For example, If the starting port is 9000, then Line 1 is using 9000 and 9001, and Line 2 is using
9002 and 9003, and so forth.
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IP Configuration (Setting WAN Port)
There are four methods of obtaining a WAN port IP address:
1. Static IP
2. DHCP, means a Dynamic IP (Cable Modem)
3. PPPoE (Dialup ADSL)
4. PPTP.
Using the DHCP and PPPoE for obtaining an IP address may vary. If not familiar with the
network connection, please contact your local ISP.
Setting Dynamic IP (DHCP)
Click “DHCP” to obtain a Dynamic IP address, then click the “Accept” button at the bottom of the
screen.
Save the settings: Click System Operation to select “Save Settings”,
”Restart”, and then click the ”Accept” button. Wait for a while (about 40 seconds), and the system
will obtain the related IP value from the DHCP Server.
NOTE: After the system has obtained a new IP address, if using WAN Port to enter the
Web Configuration Screen, a new IP address has to be used. The same applies to the
following two settings.
Setting Static IP
Select “Static IP” and enter the IP address, Subnet Mask and Default Gateway values. Then
click the “Accept” button at the bottom of the screen.
Save the settings, and then restart the system. Wait for about 40 seconds for the system to
restart.
ADSL PPPoE Settings
Select “PPPoE” Enter the Account Number, Password and Reenter Password to confirm. Then
click the “Accept” button at the bottom.
Save the settings, and then restart the system. The system will take about 49 seconds to restart.
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PPTP※
Select “PPTP” and enter the IP Address, Subnet mask, PPTP Server, PPTP ID and Password.
Then click the “Accept” button at the bottom.
Save the settings, and then restart the system. The system will take about 40 seconds to
restart.
If not familiar with the network connection, please contact your local ISP.Domain Name
Server.
BigPond (for Australia use only)
Click “BigPond Cable” Enter User Name and Password. Login Server is optional. Then click
the “Accept” button at the bottom.
(DNS) Settings
Domain Name Server (DNS): While a gateway is accessing another gateway or computer with a
hostname, it will look up the IP address from the DNS provided by ISP. The ISP whilst
negotiating with PPPoE or DHCP usually assigns the DNS information. In the case that the DNS
is not assigned automatically or WAN port is assigned with a static IP address, the DNS
information must be assigned manually.
Auto : Gateway learns primary & secondary addresses from ISP’s DHCP server or PPPoE
server.
Manual : Enter the primary & secondary addresses manually. Please be sure the IP
addresses are correct otherwise the gateway will not be able to access hosts with its
hostname.
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Clone MAC
Some Internet Service Providers (ISP) assigns the bandwidth via the MAC (Media Access
Control) Address. You can click the " Clone" button to copy the MAC address of the Ethernet
Card installed in the computer used to configure the device. It is only necessary to fill in the field
if required by your ISP.
The “Your MAC Address” will be blank as you log in through WAN port.
Using Phone Books Manager
Enable Phone Books Manager Server: It allows other Gateway users to register the IP
address and telephone number in this Phone books manager. It is recommended that
the unit appointed as the Phone Book Manager use static IP.
Share Local Phone Book: While this option is enabled and the gateway is performing as
a Phone Books Manager, this gateway will append its local Phone Book entries to the
Manager for other clients to lookup.
TTL (Time to Live): If a Gateway system that is controlled by the Phone Books Manager
does not report back within the deadline set by TTL, the system will be excluded from
the user’s list. Each Gateway should report to the Phone Books Manager once every 30
seconds.
Register to Phone Books Manager: To register to the Phone Books Manager.
Gateway Name for Phone Book Manager: The alias registered with the Phone Books
Manager.
Phone Books Manager Login Password: Enter the registered password. If this system is
serving as the Phone Books Manager, the set password is also the password used for
registering other Gateway systems.
Phone Books Manager IP/Domain: Enter the IP address for the Phone Books Manager.
It supports URL (Uniform Resource Locator).
Phone Books Manager Port: The protocol communication port for transmitting signals
between the Phone Books Manager and other Gateway systems. Please confirm
whether the setting is the same as that of the Phone Books Manager.
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Network Settings (LAN)
Network Settings (LAN): Gateway LAN Port IP address and the subnet mask value.
Port of Web Access from WAN: Http port for WAN. To make this setting, the LAN Port
QoS Settings
Please note that Gateway is built under NAT: Gateway LAN Port IP address cannot be
in the same section as the NAT LAN Port IP address, or else it is unable to make or
receive calls. For example, if the NAT LAN Port IP address is 192.168.8.1, then
Gateway LAN Port cannot be ranged between 192.168.8.1 ~ 192.168.8.254. If so,
please change the LAN port IP address, (e.g. setting the IP address to 192.168.99.254.)
must be used. Settings cannot be made using the WAN Port. Always use port 80 when
connecting to LAN port.
QoS (Quality of Service): Sets an external bandwidth to ensure sound quality during
transmission (When this function is enabled, the voice packet has the highest priority to
ensure telecommunication quality while less bandwidth is assigned for data
transmission).
ToS/DiffServ (Type of Service/DSCP): The voice packet has the highest priority to
ensure telecommunication quality; the larger the value you set, the higher priority you
will get.
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NAT/DDNS
NAT Traversal
If a Gateway is set up under an IP sharing you can select NAT or STUN protocol.
NAT: The IP address used by Gateway should be a virtual address. Further, users must
set the Virtual Server Mapping in the NAT Server (A virtual server is defined as a
Service Port，and all requests to this port will be redirected to this specified the server IP
address ).
The default port is listed below:
Listen Port (UDP): 5060
RTP Port Base (UDP): 9000~9031 (Channel used for telephone communication).
Http Port (TCP): 80 (The port can remain disable if users are not going to login from a
remote end for the setting).
NAT IP/Domain: Enter the NAT Server IP address (Real External IP address of NAT
Server). If using DDNS (Please refer to the next setting item), then fill in the URL
(Uniform Resource Locator).
STUN: Use STUN protocol prevents problems setting IP sharing, but some NAT do not
support this protocol.
STUN Server and Port: Enter the STUN server IP address and Listen Port number.
Uses UPnP: Add a new function to enable the gateway’s IP traffic to pass through a NAT
server. This function only works when the NAT server support UPnP and has it enabled.
Setting up the DDNS (Dynamic Domain Name Service) to solve the problem of general Gateway,
which cannot be set up under NAT (which uses a dynamic IP address).
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These settings are only necessary when Gateway is set up under a NAT, which not only uses a
dynamic IP address but also does not support DDNS.
Choose a DDNS Server: The current system allows users to choose either DynDNS、TZO、
3322.org、PeanutHull or a private server. Please apply for a user account before choosing a
service provider.
Server: Set up the IP address or URL (Uniform Resource Locator) of the DDNS Server.
Hostname: The URL (Uniform Resource Locator) of the system (or NAT) – apply for
from a domain name registration providers.
ID and User’s Password: The ID and password are used to login the DDNS server.
IP Auto Detect: Select only when the system is set up under NAT.
NOTE: If the Gateway is set up under NAT, then enter the hostnam e into the NAT
IP/Domain.
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Telephony Settings
Prefix Number Rules
Trunk Dial Out Verify/ Trunk Dial Out Replace: VoIP gateway will check (verify) the dial
out prefix from dial out numbers and change (replace) the prefix to transit out through
FXO port.
For example:
If you transit out with 01
with 008621123456 the system will replace it with 190200 8621123456.
907123456, the system will trans to 190601 907123456. If you transit out
Trunk Dial Out Deny: The system will deny the call with the leading number filled in this
column.
FXS Caller ID Generation: Select this option to enable the caller ID display function.
When enabled, the caller’s phone number will be displayed on your phone set when the
call comes through. FSK is preferred in some countries.
FXO Caller ID Detection: To detect the Caller ID delivered from PSTN to the FXO port.
While enabled, the Caller ID detected on the FXO port will be send to the SIP Proxy
Server on dialing out calls.
Detection Level: The gain volume occurs on Caller ID detection.
Note: You have to enable “Wait for Caller ID before FXO / Trunk pick up” to ensure Caller
ID is detected correctly.
FSK Caller ID Type: Select FSK type. In most cases, Bellcore is preferred in North
America and ETSI in Europe.
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Anonymous Caller ID (CLIR): When enabled, the caller’s phone set will not display your
number.
Note: If you register the gateway to a Proxy, you may be unable to make a call. This is
due to the fact that the gateway doesn’t send the number for authori zatio n.
Anonymous Transit in W/O Caller ID: FXO won’t detect caller ID, and the gateway will
dial out with anonymous caller identification. If the call needs caller ID to be identified for
Proxy, Proxy will reject this call without caller id.
Trunk Incoming Prompt Voice: Select the greeting (must use the IVR 132 function to
record a voice file) when FXO receives an inbound call.
Upload Greeting: It is able to upload the recorded voice file. The format must be G.723.
FXO Hunting VoIP call in option: To set FXO dial-out mode by using the default setting
or waiting caller to dial out when the VoIP call calls FXO hunting number.
FXO Hunting Default Dial-Out: To set FXO default dial-out number.
FXO Line VoIP call in options: To set FXO dial-out mode when the VoIP call indicates
the FXO extension number.
Enable: Enable a line; if some lines are not used, disable them (Pause Function) to
Hotline Functions
FXS port: When the user picks up the phone, the gateway automatically dials your
FXO port: When receiving a call from an outside line, the gateway will divert the call to
Hot Line No.: Enter the hot line number for an automatic dialing function.
Warm Line: When the warm line function is in use, user can dial a number. Otherwise
Example:
1. The assigned hotline for Line 1(Port 9) is 701 and the Warm Line(Hot Line Delay) is 5
seconds. If no extension number is dialed within 5 seconds, the call will be automatically
diverted to the assigned hotline (ext 701). The system allows users to record a voice prompt
(e.g. “please enter an extension number or wait for the operator to connect you”) to use in
this situation.
2. The assigned hotline for Line 2(Port 10) is 702 and the wait time is 0 second. When Line 2
receives a call from an outside line, it will be automatically diverted to extension 702.
avoid unnecessary waiting when an incoming call is diverting to this line.
assigned hotline number. When in hotline mode, other lines cannot be used.
the assigned hotline number.
the system will divert incoming calls from an outside line to the Hot Line Number after a
set wait time.
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Dial-out Prefix: It is the number dialed automatically by the system when the FXO
interface diverts a call to the PSTN by VoIP.
FXO Line Default Dial-Out: To set the dial-out number when this FXO line is indicated.
Example:
If PBX extension needs to dial “0” to make a PSTN call, and the FXO are connected to PBX
extension. In this case, the Dial-out prefix should be set to “0”. If the PBX requires some delay
time before capturing a line, then the trunk prefix should be set as “0,” so that after dialing a 0, it will
pause for 1 second before dialing the destination number. Each comma represents a 1 second
delay. If more delay time is required, simply add more commas. Please note that if a Dial-out
prefix is set, the line won’t be able to dial to any PBX extension line (FXS interface does not have a
trunk prefix function). Please refer to Section 6 if required to dial to a PBX extension line and
PSTN concurrently.
Enable FAX: Enable this line to detect if there is a FAX tone to transfer the Codec.
Trunk Hunting Order: To set FXO dial-out mode when there is an incoming call dialed
FXO representative number.
First Idle: The gateway will assign each unassigned call from first FXO line.
Sequential: The gateway will automatically assign the first unassigned call to the first
FXO line. The second FXO line will dial the second unassigned call out. Each line will be
used.
Enable FXO/Trunk Extension Number: Select this function only when FXO receives 2 or
more different PBX or PSTN, or under special circumstances. Users are free to call out
from a desired channel, if assigned. If you register to a Proxy it MUST be checked.
Description:
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If the user at Phone 1 (Port 1) of this system wants to assigns Line 8 (FXO) to make a call, he/she
can dial 708 22520199.
If this item isn’t checked, the gateway will select a line automatically to call out from. For example,
dial 22520199 without adding the extension number of the FXO port.
Pick up Line by Dialing Extension Number: Allows user to dial just the FXO extension –
708 - to use when the PSTN line is connected on the FXO port. If you are registered to a
Proxy, it MUST be checked.
Wait for Caller ID before FXO / Trunk pick up: To detect caller ID from FXO port.
Transit in Busy Tone Limit: The duration gateway plays a busy tone before FXO hook-on.
To notify the caller from PSTN that this call is finished.
Ring Time Limit(10 - 600secs)：The timeout to cancel a call when no one answers.
Enable End of Digit Tone：The gateway will play a “Beep-Beep” tone to notify the call is
in progress.
VoIP Calling Notification: The gateway will play a tone to notify the call is through VoIP.
Force Calling Thru PSTN code: Dial the code to get a PSTN line for dial out. For
example: If you would like dial “23456789” through PSTN and Force Calling Thru PSTN
code is *33, just dial “*33 23456789”
SIP
Early Media Treatment: If it is disabled, the system will send RTP immediately when the
connection with Proxy is set up. The default is enabled. If communicating with other
Gateway has problem, please disable this function.
Compare SIP 'To' Header for Transit Out: When FXO is callee and the number of
Request line and “To” is different, the system will use the number of “To” to dial out.
Please consult your Proxy Server Provider or ITSP about the format of invite packet
from Proxy.
All Call through OutBound Proxy：An outbound proxy server handles SIP call signaling
as a standard SIP proxy server would. Furthermore, it receives and transmits phone
conversation traffic (media) in between two talking gateways. This option tells the
gateway to send and receive all SIP packets to the destined outbound proxy server
rather than the remote gateway. This helps VoIP calls to pass through any NAT
protected network without additional settings or techniques. Please make sure your
VoIP service provider supports outbound proxy services before enable it.
Session Expiration: It is to avoid the billing of abnormal dropping the call because of
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Internet. The default is disabled.
Session Refresh Request: to send the packet of UPDATE or re-INVITE to
Session Refresher: It is the gateway’s role in Session Timer. UAS is an originator, and
UAC is a replier.
Enable P-Assert: It is for caller id protect.
Privacy Type: Privacy requested for Third-Party Asserted.
SIP Message Resend Timer Base: SIP packet will resend if response didn't arrive in the
base time set in this column. It will send again at "base time" * 2, and send again at
"base time" *2 *2. The max of resend time is 4 sec. Resend will stop/restart when total
resend 20sec has reached.
Max. Response Time for Invite: If the destination does not reply in the set time, this call
is failed.
Invite URL need ‘user=phone’: There is ‘user=phone’ in invite packet.
Reliability of Provisional Responses: Provide information on the progress of the request
processing if ticked.
Compact Form: It decreases the size of SIP header if ticked.
E.164
International Call Prefix Digit: Enter the International call prefix.
Country Code: Users please select the desired country code.
Long Distance Call prefix Digit: The long-distance prefix digit for making a long-distance
call.
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Area Code: Please enter the area code.
E.164 Numbering: To invite Proxy to follow the E.164 rule. It depends on the Proxy. If
you fail to make a call, please contact your ITSP.
Enable Support of SIP Proxy Server / Soft Switch: Enable the functions to inter-work with
Proxy Server / Soft Switch. When SIP Proxy 1 and 2 are enabled, the system will
register to SIP Proxy 2 after all lines are failed to register to SIP Proxy 1. SIP Proxy 2 is
a backup system.
Proxy Server IP/Domain: Enter the Proxy Server IP address or URL (Uniform Resource
Locator). You can set 3 redundant Proxy spread by semicolon.
EX: 61.123.231.1;12.34.56.78;proxy.sip.sip
Proxy Server Port: Enter the Proxy Server listen port number. (The factory default value
is 5060)
Proxy Server Realm: Enter the correct registered Proxy Server Realm name to avoid
registration failure. If you fail to make a call, please contact your ITSP.
TTL: Enter the desired time interval at which the gateway will report to you Proxy Server.
SIP Domain/Use Domain to Register: Enter the correct SIP domain to avoid registration
failure (it is not necessary to set with some Proxy Servers). If you enable “Uses Domain
to Register” the gateway will register to Proxy with the domain name you filed. Else, the
gateway will register to a Proxy with the IP it resolves. If you fail to make a call, please
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contact your ITSP.
Bind Proxy Interval for NAT: This function is able to keep the binding is existed when the
gateway is behind NAT and SIP Proxy is not able to keep the binding.
Initial Unregister: After rebooting, it is unregistered first and then do the general registry
process.
Enable Message Waiting Indication: The system will play a tone to remind users that
there are messages in SIP Server.
Proxy-Require: Some SIP Sever need SIP UA to add this header to it's sip message.
FXS/ FXO Representative number registers to Proxy:
Assuming that your registered ID and password are individual, the settings should be as above.
FXS Representative Number: Register all FXS ports as a hunting group.
FXO Representative Number: Register all FXO ports as a hunting group.
All the grouped FXO ports will be hunted automatically. It is available when FXO
registers to Proxy.
Register: Register to Proxy if ticked.
Invite with ID / Account: DVG-4032S can be invited to a VoIP trunk gateway w/o register
to a Proxy. Please contact your ITSP
NOTE: Please ensure that if Proxy Server allows one account for many ports using before using
representative number to register.
Each line registers to Proxy independently:
Invite with ID / Account: The gateway can be invited to a VoIP trunk gateway w/o register
to a Proxy. Please contact your ITSP
As there are various Proxy Server providers, our company has designed the gateway to be compatible
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with them, and according to RFC standards. If any registration problem occurs, please consult your
Proxy Server provider.
NOTE: When you register with a Proxy Server, dialing principles may vary with different Proxy Servers,
especially when dialing through a remote end FXO port. Please consult your Proxy Server
Provider.
Calling Features
Unconditional Forward: All incoming calls will be forwarded to the “Forwarding Number”
automatically.
Busy Forward: Forward the incoming call to “Forwarding Number” when the port is busy.
No Answer Forward: Forward the incoming call to “Forwarding Number” after ring
timeout expires without answer.
Call Hold: Enable the call hold feature on the specific FXS port.
Call Transfer: Enable the call transfer feature on the specific FXS port.
Call Waiting: Enable the call waiting feature on the specific FXS port.
Three-Way Calling / Service ID: Feature code of conference call defined on Nortel Soft
Switch
Calling Feature Instructions:
Call Hold: Ongoing call will be put on hold after FLASH button pressed on the phone set.
The gateway will play a repeating music to the remote end.
Call Transfer: Ongoing call will be put on hold after FLASH button pressed on local
phone set (gateway plays a repeating music to the remote end). Meanwhile, local user
can dial out to another number after dial tone observed. After the handset is back on the
hook, the call on hold will then be transferred to the new call regardless of the status of
the new cal. If wrong number is dialed for the new call, just press the FLASH button to
get back the call on hold. In another case, if the local user doesn’t hang up the phone
after new call sets up, press FLASH button to switch between the first call and the new
call. Please be informed that PBX between phone sets and the gateway must support
FLASH features to make this function work correctly. If a phone set is connecting directly
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to the FXS port of the gateway and not functioning to FLASH, please adjust the settings
in “Flash Detect Time” in category “Advanced Options”.
Example of a Three-Way calling:
1. Alex dials to Bob, Bob answers that call.
2. Alex presses Flash and call to Coral (Bob is on hold), Coral answers that call.
3. Alex dials *61 then presses Flash, thus conference call is created.
Or
1. Alex dials to Bob, Bob answers that call.
2. Coral dials to Alex (Call Waiting), Alex presses Flash to pick the second call and talk to
Coral.
3. Alex dials *61 then presses Flash, thus conference call is created.
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Advanced Options
There are two levels to enter Web. Administrator is able to change all settings. Web UI
only changes some settings.
Web UI auto log out: When logging in a web page, if a user does not act within the
effective time range, the user will be disconnected from the web page to allow others to
login.
Dial Wait Timeout: Use it to set the waiting time for the user’s first key pressing when
dialing a number. The user will hear a busy tone if he/she does not press the first key
within the set time frame.
Inter Digits Timeout: Set the waiting time between each key pressing. The inputted
numbers will be dialed after the timeout.
Minimum DTMF ON Length (Dial on)/ Minimum DTMF OFF Length (Dial off - between
tones): Used to set dial tone when a call is being diverted to another extension.
DTMF Detection Sensitivity: Used to adjust the sensitivity of the telephone keys.
FXO Dial Type: Choose dialing type of FXO. There are DTMF and Pulse. ※
Pulse Dial Mark/Space Ratio: Duration and break of pulse dial ration. ※
FXO/FXS Impedance: Choose correct impedance in your country/area. ※
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Enable Out-of-Band DTMF: To send DTMF keys (0~9, *, #,) follow the RFC2833 rules or
via SIP Info.
Enable Hook Flash Event: The gateway will deliver the flash signal to remote party via
RFC2833 or SIP Info.
Payload Type：Payload type of RFC2833.
Uses Second CPT for VoIP Call: This function is usually applied when the user selects
VoIP as the primary path for outgoing calls and PSTN as the backup. By enabling this
function, the gateway will generate a different set of tones to inform the user that VoIP is
in service. Should VoIP fails and fallback to PSTN, the user will hear PSTN tones
instead of the second set CPT. (for CPT related settings, please refer to Trunk
Management -> CPT Settings)
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Line Settings
Listening Volume: Adjusts the hearing volume.
Speaking Volume: Adjusts the speaking volume.
Tone Volume: Adds a new option to make tone volume adjustable. This setting will be
applied to all tones generated by the gateway including Dial Tone, Busy Tone, and so
on.
Flash Time:
FXS: Used to adjust the detecting period of flash signal from the phone set connected to
the FXS port. For example, if pressing the HOLD key will disconnect a call, increase
the “Flash Detect Time” should fix this issue.
FXO: Used to set the time frame that FXO generates a FLASH signal.
Voice
Enable Polarity Reversal:
FXS: As the remote site answer this call or hook on the FXS port will reverse the polarity.
FXO: This option forces the gateway to detect the reversal of polarity on FXO port as the
primary signal to drop a call. Some telephone switches or PBX reverse the line
polarity to inform the remote end to drop an ongoing call. Please consult with the
telephone service provider for availability of this feature.
PSTN Answer Detection: This is only used with ITSP.
PSTN Ring OFF Length: FXO/PSTN make out if the call from PSTN hangs up before
FXS answer the call. ※
Preferred Codec Type: Since different voice codec have different compression ratios, so
the sound quality and occupied bandwidths are also different. It is recommended to use
the default provided (G.723.1) because it occupies less bandwidth and will provide
better sound quality.
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Jitter Buffer: Adjusts the jitter to receive a packet. If the jitter range is too wide, it will
delay voice transmission.
Silence Suppression: If one side of a connection is not speaking, the system will stop
sending voice data (package) to decrease bandwidth usage.
Echo Canceling: Prevents poor telecommunication quality caused by echo interference.
Packet Time: Defines how long the DVG-3016S send a RTP packet-voice packet- to the
other side. The smaller the value, the more bandwidth usage. The larger the value, the
more voice delay.
Approximate Bandwidth Require: The bandwidth required varies with Codec format and
packet time.
FAX
T.38: The T.38 protocol is used for better and faster facsimile transmission. When this
function is enabled, the following fax and voice parameter settings will be disabled, so it
is recommended to enable this function to gain better fax quality. When this function is
enabled, please select UDP, TCP, or AUTO. If selecting TCP and some routers cannot
use the Fax function, please select UDP instead.
Enable High Quality: To ensure better quality.
If T.38 is disabled, then the system will use T.30 as the protocol for fax transmission.
The parameter settings are the same as for voice transmission. However, enabling the
fax function will consume more network resources and will affect transmission quality.
FAX Detect sensitivity：used to adjust the sensitivity of detect a phone call whether be
FAX or not.
This is used as a standard to determine whether or not to hang up the phone. The system will
hang up the phone automatically to avoid keeping the line engaged if the detected volume is
below the Silence Detection Threshold and the time exceeds the Drop Silent Call Timeout.
Silence Detection Threshold: Set the volume as a standard.
Drop Silent Call Timeout: Set the time to hang up the phone.
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Digit Map
There are 50 sets of leading digit entries to choose voice routing interface – Auto select, PSTN or VoIP.
Default Call Route: The default call route can be Auto, VoIP, PSTN and Deny.
Auto (VoIP first): The call route is VoIP first, and the next is PSTN.
VoIP: The call route is VoIP only.
PSTN: The call route is PSTN only.
Deny: The call will be deny if the dial-out number is not in the table.
Enable: Enable detection of this entry.
Leading Digits: The leading digits for DVG-3016S to scan while user is dialing.
Total Digit Count: Total number of digits that DVG-3016S should accept. 0 disables this
feature.
Route: The interface calls should go through if above conditions satisfied.
Local Phone Book
This system can set up and store 100 phone numbers into a phone book and provides an IP address
query when calling to other Gateway(s). If no Phone books manager is set within a Gateway group,
then all Gateway systems have to set up phone data for each VoIP Gateway to communicate with each
other.
Gateway Name: Enter other Gateways’ code or an easy-to-remember name.
Gateway Number: Enter the desired number of other Gateways.
IP/Domain Name: Enter the IP address or URL (Uniform Resource Locator) of other
Gateways.
Port: Enter other Gateways’ listen port.
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Speed Dial
This system can set up 100 numbers for speed dialing. Setting methods are as follows:
Method 1- Single mapping: Fill a short code into the “
desired phone number into the “Number To Dial” column.
For example, pick up the handset and dial 55＃ and the system will dial 32568791.
Method 2- Multi mapping; Fill the prefix code into the ” Speed Dial Code” column and the format to transfer
into the “Number To Dial” column.
For example, pick up the handset and dial 301
If the user dial 00
1657987456321, the system will DIAL 856 1657987456321
＃, and the system will dial 521301.
Caller Filter
This function is used at allow or deny SIP Invite from the Proxy list ONLY.
Filter IP Address: Fill up with the start IP you would like to allow/deny.
Subnet mask: Fill up with the subnet mask you would like to allow/deny.
Speed Dial Code” column, and enter the
ACL for Management
You can use ACL to allow the user that is from some IP address to enter Web.
Enable ACL for Management: Enable ACL for Management if ticked.
Filter IP Address: Fill up with the start IP you would like to allow.
Subnet mask: Fill up with the subnet mask you would like to allow.
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Web: Enable management from Web if ticked.
CDR Settings
The user can set up a CDR Server to record call detail for every phone call.
The present CDR provides the call detail recording in a text file and imports the text file to prepare for an
analysis report, if needed.
Send record to CDR Server: Enables the call detail recording function.
CDR Server IP: Enter the IP address of the CDR server.
Port: Enter the listen port of the CDR server.
Language
The system provides English, Traditional Chinese, and Simplified Chinese to display text on Web pages.
Meanwhile, it will change the language for IVR (Interactive Voice Response).
Transit Call Control
If you wish to restrict a general user (one who is not required to enter the PIN code) to local calls only
and prohibit him/her from making long-distance calls started with a prefix “0”, do the following steps:
1. Enable the Outbound Call Control function,
2. Set the PIN code for Outbound Level 5 to blank,
3. Set the Long-Distance Control Table to correspond with the Outbound Level 5 to prohibit
making any call with the prefix “0” (as shown above).
Note: Transit Call Control is effective when it cooperate with Long-Distance Control Table.
Inbound Call Control: To determine when users make a phone call from a PSTN to
Gateway FXO whether or not they check the inbound PIN code while using a VoIP －
only effective for incoming calls calling from a PSTN trunk.
Outbound Call Control: To determine when users utilize Gateway FXO interface to divert
to a PSTN whether or not they check the outbound PIN code － only effective for
outgoing calls being diverted to a PSTN Trunk.
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PIN Code: Enter the PIN code (4-6 digits or leave blank. A blank indicates no PIN code
is required at this level. Generally, the PIN at level 5 can remain blank to simplify the
phone number.)
Enable: Enables the PIN code at each level.
Privileges: The level is divided into 0~5 (The levels are in descending order; 0 stands for
the highest authority and 5 stands for the lowest.)
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Long-Distance Control Table
This table controls the level of authority of an outgoing call through FXO.
If Level 0 (the highest level) is set to prohibit dialing any number started with prefix 0204, then any level
below 0 (including Levels 1 to 5) is also prohibited.
If Level 1 is set to prohibit dialing any number with prefix 0, then any level below 1 (including Levels 2 to
5) is also prohibited. Since Level 0 is not restricted to any prefix, therefore at level 0 users can dial a
number with the prefix 0.
Note: Downward Restriction —If the users at a higher level cannot dial a number with a certain prefix,
then users at lowers level also cannot dial a number with the same prefix.
Long Distance Exception Table
This table handles any exceptions to the long-distance call table.
According to the Long Distance Control Table, users at Level 0 are prohibited to dial a number with the
prefix 0204. But, if the number 020488988 is set in the Exception Table as above, then users could then
dial this number.
Note: Upward Opening —If the users at a lower level can dial a number with a certain prefix, then the
users at higher levels can also dial a number with the same prefix.
CPT/Cadence Settings
CPT/Cadence setting parameters serve as the basis of an FXO interface to determine whether
or not a PSTN-call receiving party has hung up the phone. If the following parameters differ from
the parameters of the actual assigned lines, it could cause the FXO to continue to engage a line.
Busy Tone Cadence Measurement
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Busy Tone Cadence Measurement: Provide a best solution of FXO integrated with
PSTN or PBX. FXO will learn the busy tone automatically.
BTC Detection Sensitivity: The more sensitivity, the more quickly the system will cut off
the call. If the system often cut off an un-finished call, select less sensitivity.
CPT parameters Table
The CPT has 3 sets of parameter tables. Please adjust the CPT based on local PSTN or PBX.
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System Information
RTP Packet Summary
Displays the information of the last finished call. It contains peer IP, peer port, packets sent, packet
received and packet lost.
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STUN Inquiry
Ping Test
Use “ping” to identify if the remote peer is reachable. Fill in remote IP address and click “Test” will start
the test.
SNMP
Enable SNMP Agent: Enable SNMP if ticked.
Get/Set/Trap Community: Enter Community name to Read, Write and Trap.
Trap Host: Enter the IP of Trap Host.
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NTP
Time Zone: Set the Time Zone where DVG-3016S resides.
Time Server #1~#3: Set the Time Server where DVG-3016S should sync up during start
up. (NTP protocol)
Backup/Restore
You can backup settings to a file and restore settings from that file. You also can restore all settings back
to default by selecting Restore Default Configurations and click Restore.
Note: It needs to Save Settings and Restart, and all settings will back up default settings or have
new setting that you upload.
Configuration File: Backup the all settings.
Configuration Template File: Backup the settings as template file for editing.
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System Operations
Save Settings: Save settings after completing. The new settings will take effect after the
system is restarted. Please select “Save Settings”.
Restart: If it is necessary to restart the system, please select “Restart” and click the
“Accept” button.
Software Upgrade
Gateway provides software upgrade function for a remote end.
Software Upgrade Server IP address: Please enter the software server IP address.
Software Upgrade Server Port: The default setting is 6001(Do not change this setting).
Logout
Gateway only allows one user to login at a time, so whenever a change is made, please save the
settings, restart the system, or logout to avoid the situation where other users cannot login to change
settings.
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r
r
5. Coding Principle
Instruction
After a phone number is entered, dial # to call out immediately or, wait until the “Inter
DTMF Timeout” expires (defined in “Advanced Options”, default=4 seconds).
If the phone number fits the setting of Digit Map, the gateway dials out the phone number
through the assigned interface automatically.
The phone number should have at least 2 digits (not including * and #).
Dialed Number Processing Flow
To maintain maximum flexibility, the number dialed will be looked up from several tables defined
in the gateway. Once no match can be found, it will look up again from the registered SIP Proxy
Server. The look up flow is shown below:
Speed Dial
Table
A complete flow chart is on the next page.
Extension
Numbe
Local Phone
Book
Phone Book
Manage
SIP
Proxy
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d
n
k
Start
Enter a phone
number (D#)
Dial the number
defined in
SpeedDial table
Ye s
Is (D#)
efined in Spee
Dial table?
No
Is (D#)
defined in Extension
table?
No
Is (D#)
efined in Phoneboo
table?
No
Is (D#)
defined in Phonebook
Manager?
No
Ye s
Ye s
Ye s
Dial (D#) through
the first available
FXO port to PSTN
Ye s
Is (D#)
defined in SIP proxy
Does this
ateway has a
FXO port?
server?
No
No
End
Ye s
Dial out as defined
in the first match
case through the
gateway
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