Accton Technology Network Router VG3300 Series User Manual
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SIP Gateway  
VG3300 Series  
User Guide  
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SIP Gateway VG3300 Series  
User Guide  
Update: 2005/06/20  
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VG3300 series user guide  
Contents  
1.  
2.  
Safety Instructions..................................................................................................................................................... 3  
Preface ...................................................................................................................................................................... 3  
2.1.  
What is SIP................................................................................................................ 3  
2.1.1. Components of SIP ................................................................................................ 4  
3.  
4.  
Package Contents ..................................................................................................................................................... 6  
Panel Descriptions..................................................................................................................................................... 6  
4.1.  
4.2.  
Front Panel................................................................................................................ 6  
Rear Panel ................................................................................................................ 7  
5.  
6.  
7.  
8.  
LED Indicators........................................................................................................................................................... 8  
Connectors ................................................................................................................................................................ 9  
IDC Connectors (Only for VG3310/3318).................................................................................................................. 9  
Information required before Installation ................................................................................................................... 10  
8.1.  
8.2.  
8.3.  
IP Address ............................................................................................................... 10  
SIP Information.........................................................................................................11  
Prepare a password for Web Management..............................................................11  
9.  
Installation and Configuration .................................................................................................................................. 12  
9.1.  
Confirming the Region ID........................................................................................ 12  
9.1.1. Phone Setting....................................................................................................... 12  
9.1.2. System console settings (Only VG3306/3310/3318)............................................ 13  
IP Address Settings ................................................................................................. 13  
9.2.1. Static IP Mode ...................................................................................................... 14  
9.2.2. DHCP Mode ......................................................................................................... 14  
9.2.3. PPPoE Mode........................................................................................................ 15  
9.2.  
10.  
SIP Configuration............................................................................................................................................. 21  
10.1.  
10.2.  
10.3.  
10.4.  
10.5.  
10.6.  
10.7.  
Channels and SIP entity.......................................................................................... 22  
SIP Proxy and Register Parameters........................................................................ 23  
SIP Entity................................................................................................................. 24  
SIP Outbound Authentication .................................................................................. 24  
Configure STUN...................................................................................................... 25  
Check SIP entity Status........................................................................................... 27  
Phone Book............................................................................................................. 28  
10.7.1. General Phone Book ............................................................................................ 28  
10.7.2. Hotline Function.................................................................................................... 28  
10.8.  
10.9.  
Make SIP Calls........................................................................................................ 31  
Make Inbound Transit Call....................................................................................... 32  
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10.10.  
Contact Address ...................................................................................................... 34  
11. Other Parameters.................................................................................................................................................... 35  
11.1.  
11.2.  
11.3.  
11.4.  
11.5.  
11.6.  
11.7.  
Dialing Plan ............................................................................................................. 35  
Call Forward............................................................................................................ 37  
Inbound Authentication............................................................................................ 39  
FAX.......................................................................................................................... 39  
Non-SIP Call port seizure preference...................................................................... 42  
Call Waiting ............................................................................................................. 42  
Target the Media (RTP) ........................................................................................... 44  
12.  
WEB MANAGEMENT INTERFACE................................................................................................................. 46  
12.1.  
BASIC / GENERAL ................................................................................................. 47  
IP SETTING............................................................................................................. 50  
ADVANCED / GENERAL......................................................................................... 52  
SIP COMMON......................................................................................................... 54  
SIP OUTBOUND AUTHENTICATION..................................................................... 58  
SIP INBOUND ANTHENTICATION......................................................................... 60  
Dialing Plan ............................................................................................................. 61  
Inbound Transit........................................................................................................ 63  
STUN....................................................................................................................... 64  
CHANNEL ............................................................................................................... 66  
PHONE BOOK ........................................................................................................ 69  
12.2.  
12.3.  
12.4.  
12.5.  
12.6.  
12.7.  
12.8.  
12.9.  
12.10.  
12.11.  
13.  
14.  
Use Private IP (Behind NAT) ........................................................................................................................... 70  
File Management............................................................................................................................................. 71  
14.1.  
14.2.  
File Types ................................................................................................................ 71  
Software Update...................................................................................................... 71  
14.2.1. Software update via FTP...................................................................................... 71  
15.  
Appendix.......................................................................................................................................................... 74  
15.1.  
Appendix A: Phone-Set Command.......................................................................... 74  
Appendix B: Console Command ............................................................................. 76  
Specifications .......................................................................................................... 77  
Mapping table of characters used in PPPoE........................................................... 78  
Region ID ................................................................................................................ 79  
15.2.  
15.3.  
15.4.  
15.5.  
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VG3300 series user guide  
1. Safety Instructions  
WARNING  
1. Do not attempt to service the product yourself. Any servicing of this product should be referred  
to qualified service personnel.  
2. To avoid electric shock, do not put your finger, pin, wire, or any other metal objects into vents  
and gaps.  
3. To avoid accidental fire or electric shock, do not twist power cord or place it under heavy objects.  
4. The product should be connected to a power supply of the type described in the operating  
instructions or as marked on the product.  
5. To avoid hazard to children, dispose of the product’s plastic packaging carefully.  
6. The phone line should always be connected to the LINE connector. It should not be connected  
to the PHONE connector as it may cause damage to the product.  
7. Please read all the instructions before using this product.  
2. Preface  
The VG3300 unit is a personal SIP VoIP gateway developed using the latest in VoIP technology. It is  
also very simple to install and easy to operate.  
2.1. What is SIP  
Session Initiation Protocol (SIP) is the Internet Engineering Task Force's (IETF's) standard for  
multimedia conferencing over IP. SIP is an ASCII-based, application-layer control protocol (defined  
in RFC 2543& RFC 3261) that can be used to establish, maintain, and terminate calls between two  
or more end points. Like other VoIP protocols, SIP is designed to address the functions of signaling  
and session management within a packet telephony network. Signaling allows call information to be  
carried across network boundaries. Session management provides the ability to control the  
attributes of an end-to-end call.  
SIP provides the following capabilities:  
Determine the location of the target end point—Supports address resolution, name mapping, and  
call redirection.  
Determine the media capabilities of the target end point—By using Session Description Protocol  
(SDP), SIP determines the highest level of common services between the end points. Conferences  
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are established using only the media capabilities that can be supported by all end points.  
Determine the availability of the target end point—If a call cannot be completed because the target  
end point is unavailable, SIP determines whether the called party is already on the phone or did not  
answer in the allotted number of rings. It then returns a message indicating why the target end point  
is unavailable.  
Establish a session between the originating and target end point—If the call can be completed, SIP  
establishes a session between the end points. SIP also supports mid-call changes, such as the  
addition of another end point to the conference or the changing of a media characteristic or Codec.  
Handle the transfer and termination of calls—SIP supports the transfer of calls from one end point to  
another. During a call transfer, SIP simply establishes a session between the transferee and a new  
end point (specified by the transferring party) and terminates the session between the transferee  
and the transferring party. At the end of a call, SIP terminates the sessions between all parties.  
2.1.1. Components of SIP  
SIP is a peer-to-peer protocol. The peers in a session are called User Agents (UAs). A user agent  
can function in one of the following roles:  
User agent client (UAC)—A client application that initiates the SIP request.  
User agent server (UAS)—A server application that contacts the user when a SIP request is  
received and that returns a response on behalf of the user.  
Typically, a SIP end point is capable of functioning as both a UAC and a UAS, but functions only as  
one or the other per transaction. Whether the endpoint functions as a UAC or a UAS depends on  
the UA that initiated the request.  
From an architecture standpoint, the physical components of a SIP network can be grouped into two  
categories: clients and servers.  
Architecture  
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VG3300 series user guide  
SIP Clients  
SIP clients include the following:  
Phones—Can act as either a UAS or UAC. Soft phones (PCs that have phone capabilities installed)  
and SIP IP phones can initiate SIP requests and respond to requests.  
Gateways—Provide call control. Gateways provide much functionality. The most common one is a  
translation function between SIP conferencing endpoints and other terminal types. This function  
includes translation between transmission formats and between communications procedures. In  
addition, the gateway also translates between audio and video Codec and performs call setup and  
clearing on both the LAN side and the switched-circuit network side.  
SIP Servers  
SIP servers include the following:  
Proxy server—The proxy server is an intermediate device that receives SIP requests from a client  
and then forwards the requests on behalf of the client's. Basically, proxy servers receive SIP  
messages and forward them to the next SIP server in the network. Proxy servers can provide  
functions such as authentication, authorization, network access control, routing, reliable request  
retransmission, and security.  
Redirect server—Provides the client with information about the next hop or hops that a message  
should take, then the client contacts the next hop server or UAS directly.  
Registrar server—Processes requests from UACs for registration of their current location. Registrar  
servers are often co-located with a redirect or proxy server.  
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3. Package Contents  
The VG3300 Gateway  
Power Cord  
X
1
X
X
X
X
1
1
1
4
Accessories for fixing support  
System CD-ROM  
(For VG3310/3318)  
(For VG3310/3318)  
5 IDC Connector  
Rubber footer  
RJ-45 Ethernet Cable  
RJ-11 Telephone Cable  
X
X
1
1
4. Panel Descriptions  
4.1. Front Panel  
REGISTERED STUN  
VG3318 Front Panel  
VG3310 Front Panel  
REGISTERED STUN  
VG3306 Front Panel  
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4.2. Rear Panel  
There is a button on the rear panel of gateway for special maintenance. Please don’t touch this  
button under normal operation.  
VG3318 Rear Panel  
VG3310 Rear Panel  
VG3306 Rear Panel  
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5. LED Indicators  
LED  
Label  
Description  
On  
10/100  
Ethernet  
LNK/ACT  
Link up  
Off  
Link down  
Sending/Receiving  
data packets  
100Mbps  
Flash  
100Mbps  
On (LNK is on)  
Off (LNK is on)  
10Mbps  
LOOP/RING FXS  
On  
Off hook  
Off  
On hook  
Flash  
On  
Ringing out  
Line is active  
Line is inactive  
Ringing in  
FXO  
Off  
Flash  
Device  
Alarm  
Power  
The red light “On” indicates that system has  
some problem; please contact your vender.  
“On” indicates that the power supply is  
working normally.  
CPU/ACT “On” indicates that the CPU is working  
normally.  
Registered “On” indicates that all SIP entities are  
registered successful.  
“Off” indicates that all SIP entities are  
registered fail.  
“Flash” indicates that one of these SIP  
entities is registered fail.  
STUN  
“On” indicates communicate with STUN  
Server once.  
“Off” indicates never communicate with  
STUN Server.  
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6. Connectors  
Ports  
Label  
FXS  
Description  
Voice Ports  
Connects to a telephone set or fax  
machine  
FXO  
Connects to the phone line  
Ethernet  
Ports  
LAN/Internet RJ-45 connector  
MDI-X connects to a Modem  
RJ-45 connector  
PC  
MDI connects to a PC  
Console Port  
Console  
RJ-45 connector/RS-232 Interface  
(Only VG3306/3310/3318)  
7. IDC Connectors (Only for VG3310/3318)  
IDC connector is used for the voice interface (FXS and FXO) on the frame model. IDC connector  
can easily connect PBX line and telephone wire together to the gateway. No special tools are  
required; please follow the instruction to install:  
(Remarks: For IDC connector, it’s better to use No. 24 wire, e.g. CAT 5)  
Get the material ready  
Insert the insulated wires directly into the  
block for wire insertion  
Push from here  
Push the block down until it is locked to  
flush the conductor with the probe  
Cut off the conductor outside the edge to avoid from  
causing the circuit shortage  
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8. Information required before Installation  
You need to prepare the following information before installing the gateway.  
8.1. IP Address  
The gateway requires an IP address for operation. Before installation you need to know how to  
obtain an IP address from your local ISP. Static IP, DHCP or PPPoE can be used. The following  
table helps you to decide what information you need. If your ISP offers static IP, you may need to  
obtain an IP from MIS personnel in order to prevent an IP conflict. Otherwise DHCP (most cable  
broadband providers offer this) and PPPoE (most ADSL broadband providers offer this) will work  
fine.  
IP Environment  
Static IP  
Requiring information  
Public IP  
Address  
IP Address  
Subnet Mask  
Default Gateway  
It is strongly suggested that you obtain an  
IP address from MIS personnel in order to  
prevent an IP conflict.  
Private IP  
Address  
IP Address  
Subnet Mask  
Default Gateway  
It is strongly suggested that you obtain an  
IP address from MIS personnel in order to  
prevent IP conflicts.  
Your private IP requires an IP Sharing  
device and you must configure the IP  
Sharing device to treat the gateway and the  
IP that it is using as a virtual server.  
DHCP mode  
Dynamic IP address (DHCP)  
PPPoE  
Account Number  
Password  
Your ISP normally provides this information.  
If you don’t have this information please  
contact your ISP.  
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8.2. SIP Information  
Before configuring SIP, the VG3300 requires SIP information for operation. The following table  
helps you to decide what information you need.  
Items  
Description  
1. SIP Proxy  
If you want to make SIP calls through SIP proxy  
server, you will need to know the IP address or  
domain name of SIP proxy server. The proxy  
server is an intermediate device that receives  
SIP requests from a client and then forwards  
the requests on the client's behalf. If you don’t  
know which SIP proxy for setting, contact your  
SIP service provider.  
2. Public Address (SIP Account) The public address is like phone number, you  
Example: [email protected] can get the account from your SIP service  
provider.  
3. Outbound Authentication  
You will need the information when the SIP  
proxy server requires authentication. You can  
get this authentication information from SIP  
service provider when you apply for the service.  
8.3. Prepare a password for Web Management  
You will need to prepare a password for Web based Management. It can be a digit and/or letter  
combination ranging from 1 to 6 digits (E.g. 123). For security reason, password must be set to  
enter the Web Management page.  
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9. Installation and Configuration  
After preparing the information you need as specified in section 5, follow the following steps to do  
the basic configuration. You can use either a telephone or a system console to perform basic  
configurations. It is simple to connect a telephone set to FXS port and configures the system. If you  
want to use system console to configure the system (Only VG3306/3310/3318 support), you have to  
configure your VT100 terminal to match the settings of the gateway’s console port. The console  
port’s terminal connection is set to 9600 baud, 8 data bits, 1 stop bit and no parity. Turn on the  
gateway’s power and wait for the terminal to display “Press Enter…” follow the directions to begin.  
Here are several procedures to do:  
1. Confirming the Region ID.  
2. Configure IP address of gateway.  
3. Enter into the WEB page.  
4. Plan and configure the channels into SIP entity.  
5. Configure SIP proxy and register information.  
6. Configure SIP entity information.  
7. Configure Outbound Authentication (If needs).  
8. Configure STUN (If your gateway is behind NAT).  
9. Check the SIP entity if is registered successful.  
10. Configure Phone book (If needs)  
11. Make a SIP call.  
9.1. Confirming the Region ID  
About the Region ID, please refer to Section 15.5 Region ID.  
9.1.1. Phone Setting  
1. Connect the power.  
2. Connect the phone cable to the “Phone” socket on the rear panel as pictured above.  
3. When the CPU/ACT LED is on, pick up the handset and listen for the dialing tone.  
4. Dial “##0000” and listen for 3 short beep.  
5. Dial “9507#”；Assuming you are modifying for China (The last 2 digits are the regional ID)  
6. Dial “971#”；Sets the new regional ID.  
7. Hang up the phone. The device will be updated with the new region setting after it restarts  
(restart time is about 10 seconds)  
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9.1.2. System console settings (Only VG3306/3310/3318)  
SIP-RG>enable  
SIP-RG #configure  
Enter configuration commands, one per line. End with CNTL/Z  
SIP-RG(config)#regional_id 07  
SIP-RG(config)#exit  
SIP-RG#delete nvram  
This command resets the system with factory defaults.  
All system parameters will revert to their default factory settings. All static  
and dynamic addresses will be removed.  
Reset system with factory defaults, [Y]es or [N]o? Yes  
Attention:  
Before Changing the Region ID, the system has to be reset to the default value. Therefore this step  
should be done first.  
The following instruction may keep the IP address unchanged after reset:  
“delete nvram keep_ip”  
9.2. IP Address Settings  
We recommend using a traditional phone to configure the unit’s parameters, as this is the easiest  
way. The following two sections contain the procedures used to configure the gateway according to  
how you obtain your IP address (Static IP; DHCP or PPPoE).  
Every time you set a parameter item and press the “#” key to complete it, a successful setting will be  
confirmed by three equal tones in succession. If your setting is unsuccessful you will be prompted  
with one long tone.  
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9.2.1. Static IP Mode  
The following table shows an example.  
IP Address  
210.67.96.121  
Subnet Mask  
Default Gateway  
Web Management  
Password  
255.255.255.248  
210.67.96.120  
123  
Using the information contained in the example above. The procedure is as follows:  
1. Connect the gateway to a suitable Power source.  
2. Connect a traditional phone set to the “FXS” connector located on the rear panel.  
3. When the CPU/ACT light is on, pick up the phone to hear the dialing tone.  
4. ##0000  
; you should hear three short tones.  
; the digit “0” is used to enable “manual” IP mode.  
; IP address.  
5. 010#  
6. 02210*67*96*121#  
7. 03255*255*255*248#  
8. 04210*67*96*120#  
9. 15123#  
; Subnet Mask.  
; Default Gateway.  
; “123” is the web management password.  
; Warm-restarts.  
10. 981#  
11. Hang up the phone. The system should now restart.  
You can also use console to configure IP address. But phone number can’t be configured by  
console.(Only VG3306/3310/3318)  
SIP-RG>enable  
SIP-RG#configure  
Enter configuration commands, one per line. End with CNTL/Z  
SIP-RG(config)#ip state user  
SIP-RG(config)#ip address 210.67.96.121 255.255.255.248  
System need to restart  
SIP-RG(config)#ip default-gateway 210.67.96.120  
SIP-RG(config)#exit  
SIP-RG#restart  
This command resets the system. System will restart operation code agent.  
Reset system, [Y]es or [N]o? Yes  
9.2.2. DHCP Mode  
1. Connect the gateway to a suitable Power source.  
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VG3300 series user guide  
2. Connect a traditional phone set to the “FXS” connector located on the rear panel.  
3. When the CPU/ACT light is on, pick up the phone to hear the dialing tone.  
4. ##0000  
5. 011#  
; you should hear three short tones.  
; the digit “0” is used to enable “manual” IP mode.  
; “123” is the web management password.  
; Warm-restarts.  
6. 15123#  
7. 981#  
8. Hang up the phone. The system should now restart.  
You can also use console to configure IP address.  
SIP-RG>enable  
SIP-RG#configure  
Enter configuration commands, one per line. End with CNTL/Z  
SIP-RG(config)#ip state dhcp  
SIP-RG(config)#exit  
SIP-RG#restart  
This command resets the system. System will restart operation code agent.  
Reset system, [Y]es or [N]o? Yes  
9.2.3. PPPoE Mode  
If your network environment is using PPPoE, you need to prepare the information as specified in  
section 8. Information required before Installation.  
The following table shows an example.  
PPPoE Account  
[email protected]  
123ab  
PPPoE Password  
Web management password  
123  
There are three ways to configure user name and password of PPPoE  
1. Use phone set to configure:  
You can configure the user name and password by using phone set. The command ‘09’ is used for  
username and ‘10’ is for password of PPPoE. Since the user name and password use characters  
and digits are accepted by phoneset only, you need a mapping between characters and digits. You  
can find them at section 15.4  
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Mapping table of characters used in PPPoE.  
Example user name：[email protected]，Password：123ab，The procedure is below  
1. Connect the phone to the gateway  
2. When CPU/ACT is light, pick up the phone and press  
；You will hear 3 short tones.  
；Set user name：[email protected]  
；Set password is 123ab  
；Save and restart.  
3. ##0000  
4. 0938333732314068696*465742*46*46574#  
5. 103132336162#  
6. 981#  
2. Use Console to configure (Only VG3306/3310/3318)  
SIP-RG>enable  
SIP-RG#configure  
Enter configuration commands, one per line. End with CNTL/Z  
SIP-RG(config)#pppoe username [email protected]  
SIP-RG(config)#pppoe password 123ab  
SIP-RG(config)#exit  
SIP-RG#restart  
This command resets the system. System will restart operation code agent.  
Reset system, [Y]es or [N]o? Yes  
3. Use WEB Interface to configure:  
You can configure the user name and password by using WEB interface. Follow the steps to finish  
configuration.  
Step 1: Using a traditional phone set to configure the web management password and phone  
number  
You will need to use a web browser to perform the PPPoE settings through the gateway’s web  
based management interface. To enter the web based management interface you must have a  
previously configured password. Follow the next procedure to setup your password and phone  
number.  
1. Connect the gateway to a suitable Power source.  
2. Connect a traditional phone set to the “Phone” connector located on the rear panel.  
3. When the CPU/ACT light is on, pick up the phone. You should hear the dialing tone.  
4. ##0000  
5. 15123  
; you should hear three short tones.  
; “123” is the web management password.  
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6. 010#  
; “0” is to enable “manual” IP mode.  
; IP address.  
7. 02192*168*0*2#  
8. 03255*255*255*0# ; Subnet Mask .  
9. 981#  
; Used to restart the gateway.  
10. Hang up the phone to complete the configuration.  
Step 2：Configure IP address of PC  
Use the provided Ethernet cable to connect your PC to the port labeled “PC”, located on the rear  
panel of the gateway. For VG3306, VG3310, and VG3318, it is located on the front panel.  
Because the gateway’s default IP setting of this is 192.168.0.2, you must configure your PC to the  
same subnet. “192.168.0.x” for example. The following example uses 192.168.0.5 for the IP  
address and 255.255.255.0 for the subnet mask.  
After you have completed the PC’s IP address setting, you will be required to restart the PC in order  
for the new settings to take effect.  
Step 3: Using the browser to configure the PPPoE Parameters of the gateway.  
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“WEB” should  
be all Capitals  
The  
gateway’s  
IP address  
(192.168.0.  
2)  
On the PC that is connected to the gateway, enter the gateway’s IP address (Default 192.168.0.2)  
and press enter. The gateway will then prompt you with a dialogue box requesting that you enter a  
password. Use “WEB” (all capitals), for the User field and “123” for the password field that you have  
previously configured. Click the OK button; you should now have access to the gateway’s web  
based management interface page.  
Upon entering the web based configuration interface.  
Click on “IP SETTING” at the top of the page and you will see the page as shown in the following  
image.  
Select PPPoE from the “IP State” pull down menu.  
Fill in the “Account”, “Password”, and “Confirm Password” under the PPPoE Settings. You can  
obtain this information from your ISP.  
Click on the Apply button.  
Click the “BASIC” button at the top to go to the BASIC page and select “Warm Start” to restart the  
gateway. You can also perform a warm start using the phone by picking up the handset and dialing  
“##0000” then “981#”.  
After restarting, the gateway will use PPPoE to obtain it’s IP address.  
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1
Click “IP setting”  
to open this  
display  
4
2
Click the “Apply”  
button to apply  
any changes.  
3
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1
6
Click the “Apply”  
button to apply  
any changes.  
5
At this stage, your gateway should be able to use PPPoE to access the Internet. However, if you  
configured a wrong account number or password, your gateway cannot access the Internet. You are  
not able to use PC to access the gateway by using the IP address of 192.168.0.2 because the  
gateway has been set in PPPoE mode. You have to use phone set to configure the gateway back to  
fix IP mode (##0000 010#) and use PC browser to configure correct parameters.  
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10. SIP Configuration  
VG3300 not only can make regular PSTN calls, it also can communicate with IP Phones or  
Soft-Phones by using SIP protocol. Previous paragraphs have described the way to make regular IP  
calls. This section shows you what parameters you need to configure for SIP calls and how to make  
the SIP calls.  
SoftPhone (Notebook／  
PC)  
VG3300  
VG3300 (SIP)  
IP  
IP Phone (VP3302)  
Notice: These configurations on WEB page, after select or input value in the field, please press  
“Apply” button to save and confirm the setting. Some parameters need “Warm-restart”, please  
process the restart action, thanks.  
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10.1. Channels and SIP entity  
Many Channels can be assigned as on SIP Entity. Single Channel also can be assign as on SIP  
Entity.  
Application example:  
As the figure below, Channel 1-3 belongs to SIP Entity 1: [email protected]. Channel 4 and  
Channel 5 belongs to SIP Entity 2: [email protected]. and Channel 6-8 belongs to SIP Entity 3:  
[email protected]. When other device under SIP network dial into [email protected], the  
phone connect to Channel 1 is ringing. If Channel 1 is under conversation (busy), the line will be  
switched to Channel 2, and so on. So Channel 1~3 become a simple Hunting Group. (This feature  
needs the support of SIP Proxy Server).  
Figure:  
SIP IP Phone  
Internet  
VG3310  
FXS  
Busy  
Ring  
Configuration:  
WEB page: CHANNEL\  
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Notice: Each channel must belong to a SIP entity.  
10.2. SIP Proxy and Register Parameters  
You need to configure IP address or Domain name of Registrar and Outbound Proxy server, please  
check the information is right.  
SIP service provider will give you an IP address or Domain name of Registrar and Outbound proxy  
when you apply for the service.  
Configuration  
WEB Page: ADVANCED\SIP COMMOM  
Notice: The Registrar Server is only for SIP entity registering. If the SIP entity register is fail, please  
check the item. SIP calls are all through Outbound Proxy Server, if the parameter is not configured,  
the SIP call will fail. So the two parameters must be configured. If Outbound Proxy Setting is  
Enabled and Registrar Setting is Disabled, then all SIP call is routed to Outbound Proxy.  
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10.3. SIP Entity  
SIP service provider will assign one or more SIP accounts for you when you apply for the service. In  
standard, the SIP account is called ‘Public Address’, so you need to configure the account  
information in ‘Public Address’ item. The format is like an E-mail address such as  
[email protected].  
The Public Address will be generated automatically with the format below if user keeps the Public  
Address empty.  
"Default account's username" @ "Registrar" if you had enter the information below  
1. Registrar Setting. For example: fwd.pulver.com, which configured at 10.2 SIP Proxy and  
Register Parameters  
2. Username of Default Account. For example: 413189, which is configured at below graph  
For example: If the two data above is created, then the Public Address will be 413189@  
fwd.pulver.com  
Input Username and Password here if SIP Proxy needs it for authentication. This account  
information also helps you to create Realm for SIP Outbound Authentication and Public Address.  
Configuration  
WEB Page: ADVANCED \ SIP COMMON  
You can control the SIP entity on WEB page, just select ‘Enable’ or ‘Disable’.  
10.4. SIP Outbound Authentication  
You need to configure outbound authentication for each SIP entity if SIP proxy server or other SIP  
phone request for authentication. Please check with SIP service provider if you need the setting.  
Please select the entity then input information includes realm, username, and password.  
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"Realm" is a kind of verification for SIP Outbound Authentication. If SIP service provider does not  
provides this information. The gateway will create a default Realm (by string  
USER-UNSPECIFIED-REALM) automatically with your Username and Password mentioned on last  
section for SIP Outbound Authentication. If there are more than one SIP entity is registered on this  
gateway. The gateway creates Realm for each entity. The default Realm helps you to register the  
SIP server successfully.  
Configuration  
WEB Page: ADVANCED \ SIP OUTBOUND AUTHENTICATION  
10.5. Configure STUN  
The STUN (Simple Traversal UDP through NAT) server is an implementation of the STUN protocol  
that enables STUN functionality in SIP-based systems. The STUN server also includes a client API  
to enable STUN functionality in SIP endpoints.  
STUN is an application-layer protocol that can determine the public IP and nature of a NAT device  
that sits between the STUN client and STUN server.  
Notice: If your gateway is behind NAT (Use Private IP), must configure the parameter.  
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After configuring the parameters of STUN, please act Warm-Restart.  
Configuration  
WEB Page: ADVANCED\STUN  
You can enable and disable the service on WEB page.  
The STUN refresh time defines how long the device will send a binding request packet with discard  
flag on to STUN server. A binding packet with discard flag off will be sent each time when the  
number of binding request packet with discard flag on reach the Rebinding counts. The binding  
request packet is used to let the STUN server keep the most fresh client information.  
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10.6. Check SIP entity Status  
You can use the WEB page to check the SIP entity is registered successful or unsuccessful.  
WEB Page: ADVANCED\SIP COMMOM  
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If the status shows “REGISTERED” means successful, otherwise means fail; please notice that.  
When you find the registration is fail, first check the “Registrar Setting” configuration is normal, or  
not “Enable”.  
Then check the “Public Address” and “Outbound Authentication” configuration is in normal status.  
If the configurations are all right, please check the situation with your SIP service provider.  
10.7. Phone Book  
10.7.1. General Phone Book  
Since the SIP phone number is not easy for regular phone to dial, VG3300 provide a SIP phone  
book to let standard phone to make a SIP call easier. The phone book uses index number to map  
SIP account. User also can configure this index number to build the route by SIP Proxy or build the  
route without Proxy if destination gateway use fixed IP (Public IP or private IP in VPN)  
For instance if the phone book is configure as below:  
Index  
100  
Public Address  
[email protected]  
[email protected]  
[email protected]  
Port  
5060  
5060  
5060  
Via Proxy  
No  
<-- GW1  
<-- GW2  
<-- GW2  
200  
Yes  
201  
No  
Notice: If your SIP account is digit type like [email protected] or [email protected], you don’t  
need to configure the items.  
Configuration  
WEB page: PHONEBOOK \  
10.7.2. Hotline Function  
A new Hotline function is added for VG3300 Firmware Version 1.07 or above  
When hotline function is enabled, the FXS channel is connected to specified SIP device or  
VES3302 (if the VG3300 is configured and register to VES3302 as a client) automatically when user  
of VG3300 FXS channel picks up hand-set.  
♦ If the FXS channel is Hotlined to other SIP device (SIP Phone, Softphone), other SIP device  
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rings immediately when FXS channel user of VG3300 picks up hand-set.  
♦ If the FXS channel is Hotlined to VES-3302 Line, FXS channel user of VG3300 hear dialing tone  
from VES3302 when pick up hand-set, and then he/she can dial extension number to other SIP  
device.  
Configuration of Hotline  
♦ Enable Hotline function  
WEB page: PHONEBOOK \  
♦ Setup index number  
WEB page: PHONEBOOK \  
When Hotline function is enabled, user also needs to specify which channels (FXS only) should join  
Hotline function and which SIP number (Public Address) the channel is hotlined to.  
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Hotline mapping table  
Channel (FXS) only  
1st FXS channel  
2nd FXS channel  
….  
Index Number  
Description  
1
Index number “1” maps the 1st FXS channel  
Index number “2” maps the 2nd FXS channel  
…
2
….  
16  
16th FXS channel  
Index number “16” maps the 16th FXS channel  
Available Hotline index number  
Model  
Available Hotline Index Number  
1, 2, 3, 4  
Depends on module used. Please refer to Only FXS channel can be  
table below. counted as index number  
Depends on module used. Please refer to Only FXS channel can be  
Note  
VG3306  
VG3310  
VG3318  
table below.  
counted as index number  
VG3310/VG3318 channel mapping number  
Group  
Model  
Location  
Channel Number (Please  
select FXS port only)  
Group 1  
Group 2  
Group 3  
Group 4  
Group 1  
Group 2  
Lower module(S1), 4 ports of left side  
Lower module(S1), 4 ports of right side  
Upper module(S2), 4 ports of left side  
Upper module(S2), 4 ports of right side  
4 ports from left  
1
5
9
13  
1
5
2
6
10  
14  
2
3
7
11  
15  
3
4
8
12  
16  
4
3318  
3310  
4 ports from right  
6
7
8
Any index number that is not listed in Available Hotline Index Number is recognized as normal  
index number and they are not used as hotline function and not all of the channels have to join  
hotline function. Please see the example below  
Example Model: VG3306  
Index  
1
Public Address  
Port  
Via Proxy  
No  
Description  
Channel 1 Hotline to  
[email protected] without  
proxy  
[email protected]  
5060  
Channel 2 Hotline to  
[email protected] by  
proxy,  
2
[email protected]  
5060  
Yes  
100  
200  
[email protected]  
[email protected]  
5060  
5060  
Yes  
Yes  
No hotline function for channel  
3, 4 to dial  
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300  
[email protected]  
5060  
Yes  
User of 1st FXS channel picks up hand set, and then [email protected] rings immediately  
User of 2nd FXS channel picks up hand set, and then [email protected] rings immediately  
Hotline to VES3302  
Assume the Public Address of VES3302 is [email protected] and it has extension number  
1001 to 1002.  
1002  
SIP Phone  
(Notebook)  
SIP  
VES3302  
Entity:  
[email protected]  
0.145.70  
VTG3306  
VG3300 Series  
1001  
Hotline to  
VTG3306 Line  
So we configure the Phone Book as below  
Index  
1
Public Address  
Port  
Via Proxy  
Description  
Channel Hotline to  
[email protected]  
VES3302 directly  
Channel Hotline to  
[email protected]  
VES3302 directly  
[email protected] 5060  
Yes  
Yes  
2
[email protected] 5060  
User hears dial tone from VES3302 when pick up hand set and then dial extension no. for example  
1002, to other SIP device  
10.8. Make SIP Calls  
After you have configured the SIP phone on the SIP phone book, you can easily make SIP calls.  
You can select one way to make SIP call following these ways:  
Standard Call: Dial <numbers>+<#>.  
1. Compare dialing plan, check the number if it is in setting. Example 050.  
2. If the number is in setting, send the call to proxy. If the calls does not match dialing plan or the  
registration to the proxy is fail, then the call will be sent to PSTN.  
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3. If the number is not in dialing plan, the call will be sent to PSTN.  
Phone Book Call: Dial <#>+ <index>+<#>.  
1. Compare SIP Phone books; check the number if it is in phone book.  
2. If the number is configured in Phone Book and Proxy selection is set to "No", you will hear a  
busy tone. If Proxy selection is set to "Yes", then send the call to proxy.  
3. If the index number you had configured to use Via Proxy but it communicates with proxy failed,  
you will hear busy tone.  
4. If the number is not in phone book, you will hear busy tone.  
Force PSTN Call: Dial <*>+<numbers>.  
Always go through PSTN  
Hotline Call:  
If the channel is configured to use Hotline function, any dialing above is disabled. If the channel is  
hotlined to other SIP device, no dialing is needs after user picks up handset. Other SIP device rings  
immediately.  
Hotline Call to VES3302:  
Dial <SIP extension number> or  
<Prefix number (configured in VES-3302 Line)>  
1. If you dial SIP extension number, other SIP device that register to VES-3302 Line with that SIP  
extension number will ring.  
2. If you dial Prefix number, the call is relay to the IP-PBX network according to the Prefix Map  
specified in VES-3302 Line.  
Notice: If you do not want to dial “#” after numbers, please configure the ‘Dial Ending  
Time’ item. After the seconds, the call will be sent automatically.  
WEB Page: ADVANCED\GENERAL  
10.9. Make Inbound Transit Call  
To make an inbound transit call from PSTN to SIP, you have to enable Auto Answer function of this  
gateway  
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Please enable Auto Answer configuration at  
WEB Page: CHANNEL  
If you don't enable the Auto Answer configuration, the inbound call from PSTN will be assigned to a  
free FXS port of this gateway directly. It makes Inbound Transit Call impossible.  
When Auto Answer function is enabled, the gateway will answer the call and calling side will hear  
the second dial tone. For the Auto Answer function, it is also divided into Enable and Enable w/  
Pincode options. The configuration page is the same as above.  
Dial Inbound Transit Call when Auto Answer is configured as Enable  
Please dial the number below after the second dial tone:  
1. SIP Number + ‘#’, Example: 73797# or  
2. ‘#’ + Index Number + ‘#’, Example: #123#  
If you still need to make a call to the FXS port of this gateway, please press "*" to seize a free FXS  
port.  
Dial Inbound Transit Call when Auto Answer is configured as Enable w/ PIN code  
This Auto Answer mode provides security control for the Inbound Transit call  
Please dial the number below after the second dial tone:  
1. PIN code + ‘#’+ SIP Number + ‘#’, Example: 7742#73797# or  
2. PIN code + ‘#’+ ‘#’ + Index Number + ‘#’, Example: 7742##123#  
If you still need to make a call to the FXS port of this gateway, please press "*" to seize a free FXS  
port.  
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Notice for the Inbound Transit Call  
1. If the SIP number that user dial does not match any prefix code configured in Dialing Plan page,  
the call is disconnected.  
2. If the PIN code does not match any passwords configured in Password For Inbound Transit  
page, the call is terminated.  
3. If the Index Number does not match any pre-configured Phonebook Index in Phone Book page,  
the Index Number will be regarded as SIP number and create a IP call without applying any  
match rule configured in Dialing Plan.  
For which free FXS port that this gateway will seize, please refer to 11.5 Non-SIP Call port seizure  
preference  
The PIN code (Password for Inbound Transit) is configured at chapter 12.8 Inbound Transit  
The Dialing Plan is configured at chapter 11.1 Dialing Plan  
The Index Number is configured at chapter 12.11 PHONE BOOK  
10.10. Contact Address  
The main purpose of Contact Address is making SIP calls without proxy.  
The Contact Address is the same as the "Username" of Public Address if that field is configured. For  
S/W version above 1.05, the value is read only. Generally speaking, "Username" of Default Account  
are digits and it is regarded as SIP number.  
WEB Page: ADVANCED\SIP COMMOM  
Making SIP calls without proxy server:  
The SIP protocol allows you to make SIP calls directly to the destination number without through the  
proxy server. You can simply dial the SIP number to connect other SIP gateway. The typical  
example is: [email protected]. Other SIP gateway that had already configured  
[email protected] in Phone Book can connect this gateway by number 413189 without  
routing through SIP Proxy.  
Notice: For this type of SIP calls, the destination device’s IP address is already known and fixed.  
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11. Other Parameters  
11.1. Dialing Plan  
X means all calls will be sent to SIP proxy, if the SIP call is fail, it is disconnected. Only if Outbound  
Proxy is disabled, then the gateway will try to connect the number by PSTN. Outbound Proxy  
Setting can be configured on Web Page: SIP Common. Please refer to 12.4 SIP COMMON  
If the configuration is only ‘050’ means the numbers like 050xxxxx will send to SIP proxy, if you dial  
any other numbers like 100, the number will send to PSTN immediately.  
Dialing Plan:  
CO  
050 and 070  
FXO  
VG3300  
Dial 82261234  
Dial 050123456 or 070345678  
The call will be defined to SIP account  
and sent to SIP Proxy. If the SIP call is  
fail, then it is disconnected.  
The call is sent to  
PSTN  
FXS  
Configuration  
WEB Page: ADVANCED\Dialing Plan  
Dial In Rewriting Rule  
Number dialed from VG3300 can be converted to different number and sent to SIP Proxy. User can  
pre-define maximum 10 sets of prefix rewriting rule to convert the number that user dials before  
build the connection to SIP Proxy. It is useful to create a user-friendly dialing behavior and also can  
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limit user to dial certain number. The rules below explain the judgment.  
1. System will check the dialing plan on last page in advance to decide whether it is PSTN call or  
SIP call.  
2. If the call will be send to SIP Proxy, then system will exams the number to see if it meets  
Rewriting Rule.  
3. If the SIP call does not meets any Rewriting Rule, system will build the SIP call with the number  
that user dials.  
4. If the numbers of the SIP call meets any Rewriting Rule, then the numbers is converted (or  
limited if it meets barring rule) and system build the SIP call by converted number.  
Here is the example  
Web Folder: ADVANCED \ DIALING PLAN  
Pattern: Add the pattern that user may dial  
Rewrite: Add the converted number if user dials the same digits in pattern column.  
Fill in digits and click the AddDialin button  
By the operation above, we create a Rewriting Rule table below and it controls all SIP call.  
Pattern  
00x  
Rewrite  
X means any digits. ! means the call is terminated.  
If the prefix number dials from user are 001~009, then  
the 3 digits are removed. For example, if user dials  
0028621123456, then the system dials 86211123456 to  
build SIP call.  
If the prefix number dials from user are 0, then the digit  
is replaced with 886. For example, if user dials  
0921123456, then the system dials 886921123456 to  
build SIP call.  
0
886  
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If the prefix number dials from user are 1~9, then add  
8862 in front of the original number. For example, if  
user dials 82263368, then the system dials  
886282263368 to built SIP call.  
x
8862x  
If the prefix number dials from user are 0204, then the  
call is terminated.  
0204  
!
Matching Rule  
1. Best Match rule, the longest digits match first.  
2. Wildcard ( x digits) match last  
11.2. Call Forward  
There are three forward types:  
1. All: All incoming VoIP call to the SIP entity will be forward.  
2. Busy: When the SIP entity is busy, the incoming VoIP call will be forward.  
3. No Answer: When the SIP entity is no answer and after 30 seconds, the incoming VoIP call will  
be forward.  
Notice:  
ꢀ
ꢀ
ꢀ
In order to let the caller identify the port has been configured ”forward”; the caller will hear  
second dial tone, rather than normal dial tone.  
If Auto Answer function is disabled, incoming call from PSTN seizes a free FXS port. The call  
is not forwarded even the seized FXS port is part of Call Forward SIP Entity.  
If Auto Answer function is enabled, Incoming PSTN call dials "*" to seize a free FXS port after  
second dial tone. The call is not forwarded even the seized FXS port is part of Call Forward  
SIP Entity.  
ꢀ
If Auto Answer function is enabled, Incoming PSTN call dials "SIP phone number" of the  
gateway itself after second dial tone. The call is forwarded to other VG3300 or SIP device.  
Configuration  
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WEB page: ADVANCED\SIP COMMOM  
Phone Set: Please refer to section Appendix A: Phone-Set Command.  
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11.3. Inbound Authentication  
You need to configure inbound authentication if you request authentication for other SIP phone to  
call you.  
Configuration  
WEB Page: ADVANCED \ SIP INBOUND AUTHENTICATION  
11.4. FAX  
For VG3300 software version 1.05 or above, SIP-based T.38 Fax protocol is applied. Any brand SIP  
gateway with SIP-based T.38 Fax protocol can transmit FAX with each other. T.38 is FAX protocol  
and it has better performance and better successful transmission rate. However, SIP device that  
does not support SIP-based T.38 still can transmit and receive FAX with VG3300 by G.711 codec.  
G.711 codec uses more bandwidth, so it may not as good as SIP-based T.38 protocol if bandwidth  
control is the key factor of the network.  
Setup method is listed below:  
1. Web folder: “Channel”  
Enable T.38 Fax Relay support. Configure it to Yes  
2. Warm-Restart the system  
Note: For FAX transmission, two gateways will change to SIP-Based T.38 Protocol automatically if  
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both sides support SIP-based T.38.  
Note:  
If VG3300 connects different SIP devices, some have T.38, but some use G.711 codec only, then  
user should enable G.711 codec support for FAX. Setup method is listed below:  
1. The same step as above set Connect Device to Fax  
2. Setup “Codecs Type“, Web Folder: ADVANCED\SIP COMMON  
Select and mark “PCMU” and “PCMA” Codecs (G.711 Standard), than click “Apply” button  
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3. Warm-Restart the system  
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11.5. Non-SIP Call port seizure preference  
For non-SIP Calls, the port seizure preference is listed below  
1. Inbound from PSTN  
If the inbound FXO port was configured as "Fax" device, it will also seize only FXS ports that  
"Connect Device" is configured as Fax. The Voice devices behave the similar way.  
From FXO port to FXS port  
Note  
Connect Device at FXO port  
Connect Device at FXS port  
Select VOICE port only  
VOICE port  
From the lowest port number  
upward  
FAX port  
Select FAX port only  
From the lowest port number  
upward  
2. Outbound to PSTN  
For the calls from FXS to FXO, the ports of the same "Connect Device" type will be the prior  
selection for the calls.  
If there is no correct configured port is available, it will ignore the "Connect Device" setting and  
create a call as the rule below.  
From FXS port to FXO port  
Note  
Connect Device at FXS port  
Connect Device at FXO port  
Select VOICE port (1st priority)  
Select FAX port (2nd priority)  
Select FAX port (1st priority)  
Select VOICE port (2nd priority)  
VOICE port  
FAX port  
From the highest port  
number downward  
From the highest port  
number downward  
For the setting of "Connect Device", please refer to 12.10 CHANNEL  
11.6. Call Waiting  
Call waiting function for a FXS port to answer two SIP calls.  
When D answer a SIP call from other SIP phone or gateway, such as A. In normal condition,  
another incoming call dial to D will be busy, such as B to D. With Call Waiting function, the phone  
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call dials from B to D will not be busy. Here is the possible situation.  
ꢀ
ꢀ
ꢀ
D keeps talking with A and hears Call Waiting Tone if B calls D.  
B hears normal ring back tone without sense any different.  
If D keep talking with A and ignore the Call Waiting Tone for more than 30 seconds, Call  
Waiting Tone stop and the phone call return to normal condition  
ꢀ
ꢀ
If D keep talking with A and ignore the Call Waiting Tone for more than 30 seconds, B keep  
hearing ring back tone for 30 seconds and listen busy tone finally.  
D can talk to B if D presses Flash button when hearing the Call Waiting Tone. Phone A is silent  
when D talk to B.  
ꢀ
ꢀ
D can talk to A or to B by keep pressing Flash button to switch the two side.  
C will hear busy tone when C call to D if there is one line in call waiting status for A.  
3702A  
SIP Phone  
SIP GW  
3702B  
D
E
Configuration  
Enable the Call Waiting function of the FXS port (D) of VG3300 gateway. This function can be  
configured for each FXS port individually.  
Web Folder: Channel\  
43  
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Connection Type  
A: FXS port of VG3300 Series  
B, C: SIP Device (VG3300 Series, other brand SIP gateway. SIP phone...), Normal PSTN phone call  
(special condition is described below)  
Call waiting function works only on SIP call. So PSTN call works when it is transited as SIP call. If  
Inbound transit call is configured on VG3300 (please refer to 10.9 Make Inbound Transit Call), then  
Call Waiting function is available when user dials the SIP number of this VG3300 gateway itself. If  
no inbound transit call function is configured, it is impossible to do call waiting function.  
11.7. Target the Media (RTP)  
For the SIP call passing through NAT, it is possible that the media would not deliver properly; owing  
to the RTP contact information (IP address, port number) is different from original RTP packet. This  
function selects different contact information for VG3300 to send RTP Packets to other SIP device  
within far-end NAT. It designates whether to use the source contact information from the UDP/IP  
header (Symmetric RTP) or the contact information specified within the packet (SDP) when the  
gateway send RTP packet  
Web Folder：ADVANCED\SIP COMMON, Default Value is SDP  
Example 1: Via Symmetric RTP  
The source contact information (IP, port number) of RTP packet is IP: 61.222.217.30, port number:  
10000, but the SDP in the packet is IP: 10.13.6.18, port: 4000. In this case, please Use  
Symmetric RTP  
61.222.217.30  
port: 10000  
VG3300 Series  
(192.72.83.23,  
port: 10000)  
SDP in Packet  
10.13.6.18  
port: 4000  
Network  
VG3300 tries the contact information from SDP first (IP:10.13.6.18, port number: 4000). If VG3300  
finds that the contact information from SDP is different from the source contact information, then it  
will try the source contact information, as the example above, use IP:61.222.217.30, port  
number:10000. It makes SIP call successful.  
Example 2: Via SDP (Default)  
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This selection ignores the source contact information (IP, port number) which VG3300 received. It  
always sends the RTP packet to the contact information (IP, port number) described in the packet  
(SDP) received.  
Send RTP to  
10.13.6.18  
port: 4000  
VG3300 Series  
(192.72.83.23,  
port: 10000)  
Network  
SDP in Packet  
10.13.6.18  
port: 4000  
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12. WEB MANAGEMENT INTERFACE  
The Tree Architecture of Web Management is shown below  
HOME  
BASIC  
GENERAL  
IP SETTING  
ADVANCED  
General  
SIP COMMON  
SIP OUTBOUND  
AUTHENTICATION  
SIP INBOUND ATHENTICATION  
STUN  
Dialing Plan  
Inbound Transit (for gateway has  
FXO port. Gateway without FXO  
port does not have this page)  
CHANNEL  
PHONE BOOK  
ACCESS  
CODE  
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12.1. BASIC / GENERAL  
47  
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Category  
Section  
Description  
Default Setting  
0
Information  
Region ID  
Software  
Version  
Display region ID.(Read only)  
Display software version.(Read only)  
BootRom  
Version  
Display BootRom Version.(Read only)  
Display hardware Version.(Read only)  
Hardware  
Version  
Card Type  
Up-Time  
Display card type. (Read only)  
Display the use time since from system  
reboot.(Read only)  
MAC  
Display MAC address.(Read only)  
Address  
Date  
Show the date  
Time  
Show the time  
Time  
Time  
Select the time server to synchronize  
the time of this gateway  
♦ Registrar: Get the time data from the  
Registrar Server.  
Registrar  
Configuration Source  
♦ NTP Server: Get the time data from  
the NTP Server  
NTP Server Input the address if the system use  
NTP server as time synchronization  
source. The gateway will synchronize  
with the NTP Server once a day. If the  
NTP server inputted here is not  
available or fail to response, the  
gateway will retry it every 5 minutes.  
The gateway has its own clock, so the  
clock will keep going according to last  
synchronization time. For NTP server  
information, please refer to  
http://www.ntp.org  
Time Zone  
Select local system time zone. Select  
correct Time Zone.  
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Daylight  
saving  
Signaling  
Port  
ON: Enable daylight saving.  
OFF: Disable daylight saving.  
OFF  
Auxillary  
protocol  
UDP port to transfer signal packets. It  
can be setting in the range of 0 to  
65535. (Must reboot system to apply  
changes)(Only support VG and VTG  
devices)  
0
RTP  
Base of UDP port to receive RTP  
packets. It can be setting in the range of  
0 to 65534.( Must be Even, after setting  
this item, please reboot system to apply  
changes)  
4000  
Base Port  
System  
Restart  
Restart  
Mode  
None: Not to restart system.  
Cold restart: Cold restart.  
Warm restart: Warm restart.  
None  
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12.2. IP SETTING  
Category  
Section  
IP State  
Description  
Default Setting  
IP Settings  
The way to obtain IP address: Manual  
Manual: Entered by user  
(Static IP)  
Auto(DHCP): Assigned by  
DHCP server  
PPPoE: Assigned by PPPoE of  
ISP  
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Current Setting  
Change To  
Display the configured IP  
192.168.0.2  
address, subnet mask address 255.255.255.0  
and default gateway. (Read  
only)  
192.168.0.1  
Enter the IP address that will  
be used after next restart,  
Including:  
IP Address  
Subnet Mask Address  
Default Gateway  
(This item is used only on  
Manual mode of IP Setting.)  
The user’s account of PPPoE  
protocol, provided by ISP.  
The user’s password of PPPoE  
protocol.  
PPPoE  
Account  
Settings  
Password  
Confirm  
Confirm the user’s password of  
PPPoE protocol.  
Password  
Service Name  
The service name of PPPoE  
account, provided by ISP.  
(Most ISP doesn’t need this)  
The primary address of DNS  
server. The default setting  
would be different according to  
the local area. In Taiwan, the  
default setting is 168.95.1.1.  
The secondary address of  
DNS server.  
DNS Server  
Primary Address  
168.95.1.1  
Secondary  
Address  
Web  
User Name  
The user’s name of Web  
Management Interface.(12  
character)  
WEB  
Password  
Password  
The password of Web  
Management Interface.( 6  
character)  
Password  
Confirm  
Enter the password again to  
confirm it.  
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12.3. ADVANCED / GENERAL  
Category  
Section  
Default Setting  
200 msec  
Description  
Flash Button  
Flash Time  
System confirmed  
“Flash” time.  
Touch Tone (DTMF)  
Duration  
The duration to send a  
DTMF.  
100 msec  
100 msec  
Inter-digit  
The inter-digit time of  
sending string of DTMF  
digits.  
Guard Time  
Line  
The time defines how  
long the system will not  
take incoming call after  
call has been  
0.8 sec  
disconnected.  
Dial Ending Time  
Dial Ending  
Time  
The time specifies how  
long to end the dialing  
4
1-10 (seconds)  
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number if a ‘#’ digit is  
missing.  
Redundancy Number of times to retry  
T.38 Fax protocol. Use  
more Redundant packet  
when network is  
unstable.  
T.38 Fax Relay  
No Redundant packet  
1 Redundant packet  
2 Redundant packets  
3 Redundant packets  
4 Redundant packets  
(300 ~ 3000Hz)  
(100 ~ 5000ms)  
Frequency  
Cadence  
f1, f2  
on, off. The on and off  
duration in playing the  
tone  
Busy Tone Spec  
(300 ~ 3000Hz)  
(100 ~ 5000ms)  
Frequency  
Cadence  
f1, f2  
on, off. The on and off  
duration in playing the  
tone  
Reorder Tone Spec  
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12.4. SIP COMMON  
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Section  
Item Field Description  
Default  
Port and Header  
Port  
The control port number of SIP protocol. 5060  
Header  
Form  
Select ‘Standard’ or ‘Compact’ to be the  
Standard  
header format of SIP packet. When  
Compact is selected, the header will be  
shorter and it saves bandwidth.  
Outbound Proxy  
Setting  
Domain  
Name  
Port  
Domain name or IP address of proxy.  
Empty  
Disable  
5060  
Control port number of SIP protocol.  
Domain name or IP address of proxy  
that you want to register.  
Registrar Setting Domain  
Name  
Empty  
Disable  
Disable  
Out-band DTMF  
Control  
Enable/Disable  
Enable: It “Disable” RFC 2833 DTMF  
Incoming Call  
Screening  
Screening Disable: Accept all incoming SIP call  
Enable: This gateway only accepts  
incoming call through SIP  
Disable  
Proxy.  
NAT Signalling  
Keep Alive  
Control  
Port number mapping may change if the Disable  
connection to pass through some NAT  
device is timeout. This function sends  
Dummy Packet to Proxy server every 50  
seconds to keep the port number via  
NAT intact.  
Disable: Does not send Dummy Packet  
Enable: Send Dummy Packet  
Target the media Via  
(RTP)  
Select the contact information (IP  
Address, Port Number) to pass through  
NAT device. Please refer to 11.7 Target  
the Media  
SDP  
SDP: via SDP  
Symmetric RTP: via Symmetric RTP  
Codecs Selection Codec  
Type  
G.729AB: Mark the selection to Enable Enable  
G.729AB Codec  
G.723.1: Mark the selection to Enable Enable  
G.723.1 Codec  
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Section  
Item Field Description  
Default  
PCMU: Mark the selection to Enable Enable  
PCMU Codec (G.711 u Law)  
PCMA: Mark the selection to Enable Enable  
PCMA Codec (G.711 A Law)  
Codec  
Priority  
You can select the codec priority for  
your requirement.  
G729-G723-P  
CMU-PCMA  
SIP Entity  
SIP Entity Select an entity and click Select button 1  
to display follow items’ setting of SIP  
entity section.  
Select: Select Button  
Register: Register Button  
De-Register: Cancel Register Button  
Entity  
Select Enable/Disable  
Enable  
Empty  
Control  
Register  
Status  
Show the register status, if it shows  
Registered means successful. (Read  
only)  
Register: Register Button  
De-Register: Cancel Register Button  
Calling Line Identification Restriction  
Disable: Send caller ID to SIP proxy  
when user make SIP call  
CLIR  
Disable  
Enable: Don’t send caller ID when user  
make SIP call. Note that for some SIP  
Proxy Server, the SIP call is failed if no  
caller ID is sent. Please set “CLIR”  
Disable for this case. That’s the reason  
why default value is disable.  
Public Address  
Setting  
Address  
Enter SIP phone number of the port.  
The phone number general assigned by  
SIP service provider.  
Empty  
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Section  
Item Field Description  
Default  
Default  
Account information for registering SIP  
Proxy  
Account  
Username: It may the same as your SIP  
number  
Password: Password for Authentication  
Confirm Password: Reconfirm  
Password  
Contact Address  
Setting  
Current  
Setting  
Display current setting of (Read Only)  
Contact Address. It will be  
the same as the  
Username of Public  
Address Setting at this  
page of web if that field is  
configured  
RFC 2833 DTMF 2833  
DTMF  
Enable: Enable RFC 2833 DTMF.  
Negotiate: Encode DTMF to message  
and decode it back at destination.  
Never: Convert DTMF to voice and sent  
by RTP packets.  
Never  
2833 In  
Use  
Display current status of  
DTMF configuration.  
(Read Only)  
Forward To  
Forward  
Address  
Enter a SIP account (Public Address)  
forward. When users dial into the SIP  
Entity, the call will be forwarded to the  
number. Only SIP calls can be  
forwarded.  
Empty  
Type  
N/A: All incoming calls are forward.  
Busy: When the SIP entity is busy, the  
calls will be forward.  
N/A  
No Answer: When the SIP entity is no  
answer about 30 seconds, the calls will  
be forwarded.  
SIP Entity  
Members  
Channel  
Show the all channels  
Depend on  
gateways  
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Section  
Item Field Description  
Default  
Entity  
Show ‘+ ‘ means the SIP entity is for the Empty  
channel.  
12.5. SIP OUTBOUND AUTHENTICATION  
Section  
Item Field Description  
Default  
SIP Outbound  
Authentication  
Maximum Maximum number of entries (Read Only) 50  
allowed  
Entered  
Number of entries of  
authentication entered.  
List of entries  
(Read Only) 0  
Entries  
(Read Only) Empty  
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Section  
Item Field Description  
Default  
List  
Entity: Which entity that you select.  
Realm: Domain name or IP address.  
Username: Username of authentication.  
The gateway creates default entry  
according to the Public Address Setting  
for easy registration. Please refer to 10.3  
SIP Entity and 10.4 SIP Outbound  
Authentication  
Update  
Entry  
Enter the information of outbound  
authentication  
Empty  
Entity: Select an entity.  
Realm: Domain name or IP address.  
Username: Enter Username of  
authentication.  
Password: Enter password of  
authentication.  
Confirm Password: Enter password again  
for confirmation.  
Delete  
Entry  
Delete the information of outbound  
authentication  
Empty  
Entity: Select an entity.  
Realm: Domain name or IP address.  
59  
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12.6. SIP INBOUND ANTHENTICATION  
Section  
Item Field  
Realm  
Description  
Default  
SIP Inbound  
Authentication  
Enter domain name, IP address or word Empty  
string.  
Maximum  
Entered  
Maximum number of  
entries allowed  
(Read Only) 20  
Number of entries of  
authentication entered.  
Display the entries  
(Read Only) 0  
Entries List  
(Read Only) Empty  
Entity: Which entity that you select.  
Username: Username of authentication.  
60  
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Section  
Item Field  
Description  
Default  
Empty  
Update Entry Enter entries of authentication  
Entity: Which entity that you select.  
Username: Username of authentication.  
Password: Password of authentication.  
Confirm Password: Enter password  
again for confirmation.  
Delete Entry  
Delete entries of authentication  
Entity: Which entity that you want to  
delete.  
Empty  
Username: Username of authentication.  
12.7. Dialing Plan  
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Section  
Item Field  
Maximum  
Description  
Default  
DIALING PLAN  
Maximum number of (Read Only) 100  
entries allowed  
Entered  
List  
Number of entries of (Read Only) 1  
authentication  
entered.  
Display the entries (Read Only) x  
The default value “x“ means that  
all numbers that you dial will first  
go through SIP proxy.  
Add Dialing Plan Enter numbers. Example: 050.  
Delete Entry Enter numbers for delete.  
Empty  
Empty  
Dial In Rewriting Control  
Rule  
Digits dialed from VG3300 can be Disable  
rewrite to different digits and sent  
to SIP Proxy.  
Enable/Disable  
Capacity  
List  
The max set of rewrite number  
List the entries of original digits  
and the rewrite digits  
Pattern: the pattern that user may  
dial  
Rewrite: the converted number if  
user dials the same digit in  
pattern column.  
Add Dialin (button) Pattern: Add the pattern that user  
may dial  
Rewrite: Add the converted  
number if user dials the same  
digit in pattern column.  
Fill in digits and click the Add  
Dialin button  
Del Dialin (button) Fill in the Pattern digit that will be  
deleted and click Del Dialin button  
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12.8. Inbound Transit  
Only VG3300 gateway with FXO port has this web page.  
Group  
Field  
Description  
Default Value  
Transit call  
Warning Time  
This gateway will send warning tone periodically to 60  
check if the line is still alive. If calling side fail to  
press any key after hearing the warning tone, the  
line will be disconnected.  
Release Call by This gateway will check the RTP packet  
0
Checking RTP  
periodically to verify if the line is still alive. If no RTP  
packet is found, the gateway will disconnect the  
call. When this value is set to "0", means the  
gateway will not check the RTP packet  
Password  
For Inbound  
Transit  
Maximum  
Entered  
Display no. of password can  
be accepted  
(Read only) 32  
Display the no. of password  
had been entered  
(Read only) 0  
Entries List  
List the detail data of password  
had been entered  
(Display) Only) Blank  
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Group  
Field  
Description  
Default Value  
Add Passwords Enter a new password, any combination of digits Blank  
(0~9), less than 9 characters. The password will be  
used at PINcode for auto answer function  
Delete  
Enter the password to be deleted, refer the detail Blank  
data under Entries List  
Passwords  
12.9. STUN  
Section  
Item Field Description  
Control Enable or Disable STUN Server service.  
Default  
Disable  
STUN Server  
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Section  
Item Field Description  
Address Input this NAT WAN IP helps you to pass  
through NAT without using STUN server.  
Default  
NAT WAN IP  
The port number inside and outside NAT  
should be the same. NAT WAN IP is the  
Public IP that used on NAT device  
Note: If you disable STUN server and  
input NAT WAN IP here, the RTP  
(normally 4000) and Signaling (normally  
5060) port number inside and outside  
NAT must be the same, and Server Port  
need to be configured on NAT device.  
STUN Server  
Setting  
Maximum Maximum number of  
entries allowed  
(Read Only) 5  
Entered  
Number of entries of  
STUN server that have  
been entered.  
(Read Only) 0  
List  
Display all of servers that (Read Only)  
have been entered.  
Add  
Add a stun server  
Empty  
Empty  
IP Address: Enter IP address or Domain  
Name  
Port: Enter port number of service.  
Delete a stun server  
Delete  
Type  
IP Address: Enter IP address.  
Port: Enter port number of service.  
NAT Type  
Display NAT type  
(Read Only) Unknown  
Stun Refresh Time Interval  
It defines how long the device will send 30  
a binding request packet with discard  
flag on to STUN server.  
Mapping List  
List  
My ip/port: shows the  
private IP and port  
number.  
(Read Only) Empty  
Global ip/port: Display  
public IP and port number.  
65  
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