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SIP Gateway
VG3300 Series User Guide www.edge-core.com
SIP Gateway VG3300 Series
User Guide
Update: 2005/06/20
VG3300
series user guide
Contents
System console settings (Only VG3306/3310/3318)............................................ 13
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SIP OUTBOUND AUTHENTICATION ..................................................................... 58
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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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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.
R egistrar 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 X 1
Rubber footer
RJ-45 Ethernet Cable X 1
RJ-11 Telephone Cable X 1
4. Panel Descriptions
REGISTERED STUN
VG3318 Front Panel
REGISTERED STUN
VG3310 Front Panel
VG3306 Front Panel
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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
10/100
Ethernet
LNK/ACT
100Mbps
Flash Sending/Receiving data packets
On (LNK is on)
Off (LNK is on)
100Mbps
10Mbps
LOOP/RING FXS
Device
FXO On
Off
Line is active
Line is inactive
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
Voice Ports
Ethernet
Ports
Console Port
(Only VG3306/3310/3318)
FXS Connects to a telephone set or fax machine
Connects to the phone line FXO
LAN/Internet RJ-45 connector
MDI-X connects to a Modem
Console
MDI connects to a PC
RJ-45 connector/RS-232 Interface
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 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
Push from here
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8. Information required before Installation
You need to prepare the following information before installing the gateway.
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 Requiring information
Static IP 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
Dynamic IP address (DHCP)
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
Password
Your ISP normally provides this information.
If you don’t have this information please contact your ISP.
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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.
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”
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
Subnet Mask
210.67.96.121
255.255.255.248
Default Gateway
Web Management
Password
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.
5. 010# ; the digit “0” is used to enable “manual” IP mode.
6. 02210*67*96*121# ; IP address.
7. 03255*255*255*248# ; Subnet Mask.
8. 04210*67*96*120# ; Default Gateway.
9. 15123# ; “123” is the web management password.
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
1. Connect the gateway to a suitable Power source.
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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.
5. 011# ; the digit “0” is used to enable “manual” IP mode.
6. 15123# ; “123” is the web management password.
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
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
PPPoE Password
123ab
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
3. ##0000 ； You will hear 3 short tones.
4. 0938333732314068696*465742*46*46574# ；Set user name：[email protected]
5. 103132336162# ； Set password is 123ab
6. 981# ； Save and restart.
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 ; you should hear three short tones.
5. 15123 ; “123” is the web management password.
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6. 010# ; “0” is to enable “manual” IP mode.
7. 02192*168*0*2# ; IP address.
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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The gateway’s
IP address
(192.168.0.
2)
“WEB” should be all Capitals
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
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4
Click the “Apply” button to apply any changes.
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6
Click the “Apply” button to apply any changes.
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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
IP
VG3300 (SIP)
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:
. 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
Busy
Configuration:
WEB page: CHANNEL\
Ring
Internet
VG3310
FXS
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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
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
Port
5060
Via Proxy
No <-- GW1
Yes GW2
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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♦ 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
1 st FXS channel
2 nd FXS channel
Index Number
1
2
Description
Index number “1” maps the 1 st FXS channel
Index number “2” maps the 2 nd FXS channel
16 th FXS channel 16 Index number “16” maps the 16 th FXS channel
Available Hotline index number
Model
VG3306
VG3310
VG3318
Available Hotline Index Number Note
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 table below. counted as index number
VG3310/VG3318 channel mapping number
Model
3318
3310
Group Location Channel Number (Please select FXS port only)
Group 1 Lower module(S1), 4 ports of left side 1 2
Group 2 Lower module(S1), 4 ports of right side 5 6
3
7
4
8
Group 3 Upper module(S2), 4 ports of left side 9 10 11 12
Group 4 Upper module(S2), 4 ports of right side 13 14 15 16
Group 1 4 ports from left
Group 2 4 ports from right
1
5
2
6
3
7
4
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
5060 No
Description
Channel 1 Hotline to
without proxy
[email protected] by proxy,
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300 [email protected] 5060 Yes
User of 1 st FXS channel picks up hand set, and then [email protected]
rings immediately
User of 2 nd 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)
VTG3306
VES3302
Entity:
0.145.70
SIP
VG3300 Series
1001
Hotline to
VTG3306 Line
So we configure the Phone Book as below
Index
1
Public Address
Port
5060
Via Proxy
Yes
Description
Channel Hotline to
VES3302 directly
5060 Yes Channel Hotline to
VES3302 directly
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.
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
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:
050 and 070
CO
Dial 82261234
The call is sent to
PSTN
VG3300
Configuration
WEB Page: ADVANCED\Dialing Plan
FXO
FXS 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.
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 Rewrite
If the prefix number dials from user are 001~009, then the 3 digits are removed. For example, if user dials
00x
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
0 886
0921123456, then the system dials 886921123456 to build SIP call.
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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 x 8862x user dials 82263368, then the system dials
886282263368 to built SIP call.
If the prefix number dials from user are 0204, then the
0204 ! call is terminated.
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
Connect Device at FXO port Connect Device at FXS port
VOICE port Select VOICE port only
Note
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
Connect Device at FXS port
VOICE port
FAX port
Connect Device at FXO port
Select VOICE port (1 st priority)
Select FAX port (2 nd priority)
Select FAX port (1 st priority)
Select VOICE port (2 nd priority)
Note
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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VG3300
series user guide 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
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
VG3300 Series
(192.72.83.23, port: 10000)
SDP in Packet
10.13.6.18 port: 4000
61.222.217.30 port: 10000
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
VG3300
series user guide
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Category Section Description
Information Region ID Display region ID.(Read only)
Display software version.(Read only) Software
Version
BootRom
Version
Hardware
Version
Display BootRom Version.(Read only)
Display hardware Version.(Read only)
0
Time
Configuration
Card Type Display card type. (Read only)
Up-Time Display the use time since from system reboot.(Read only)
MAC
Address
Date
Display MAC address.(Read only)
Time
Time
Source
Show the date
Show the time
Select the time server to synchronize the time of this gateway
♦ Registrar: Get the time data from the
Registrar Server.
♦ 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.
Registrar
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Auxillary protocol
System
Restart saving
Signaling
Port
RTP
Base Port
Restart
Mode
VG3300
series user guide
OFF
OFF: Disable daylight saving.
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)
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)
None: Not to restart system.
Cold restart: Cold restart.
Warm restart: Warm restart.
0
4000
None
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12.2. IP SETTING
Category Section
IP Settings IP State
Description
The way to obtain IP addr ess:
Manual: En tered by user
(Static IP)
Auto(DHCP): As signed by
DHCP server
PPP oE: Assigned by PPPoE of
ISP
Manual
S etting
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PPPoE
Settings
DNS Server
Web
Password
Current Setting
Change To
Account
Password
Confirm
Password
Service Name
Primary Address
Secondary
Address
User Name
Password
Password
Confirm
Display the configured IP address, subnet mask addr ess and default gateway. (Read only)
Enter the IP address th at will be used after next restart,
Including:
IP Address
Subnet Mask Address
Default Gateway
(This item is used only on
Manual m ode of IP Setting.)
The user’s account of PPPoE protocol, provided by ISP.
The user’s password of PPPo E protocol.
Confirm the user’s password o f
PPPoE protocol.
1 92.168.0.2
255.255.255.0
192.168.0.1
The service name of PPPoE account, provided by ISP.
(Most ISP doesn’t need this)
168.95.1.1 The primary address of DNS server. The default setting would be diffe rent according to the local area. In Taiwan, the default setting is 168.95.1.1
.
The second ary address of
DNS server.
The user’s name of Web
Manageme nt Interface.(12 character)
WEB
The passw ord of Web
Management Interface.( 6 character)
Enter the password again to confirm it.
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12.3. ADVANCED / GENERAL
Flash Button
Touch Tone (DTMF)
Guard Time
Dial Ending Time
Flash Time System confirmed
“Flash” time.
Duration The duration to send a
DTMF.
Inter-digit
Line
Dial Ending
Time
The inter-digit time of sending string of DTMF digits.
The time defines how long the system will not take incoming call after call has been disconnected.
The time specifies how long to end the dialing
Default Setting
200 msec
100 msec
100 msec
0.8 sec
4
1-10 (seconds)
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T.38 Fax Relay
Busy Tone Spec
Reorder Tone Spec number if a ‘#’ digit is missing.
Redundancy Number of times to retry
T.38 Fax protocol. Use more Redundant packet when network is unstable.
No Redundant packet
1 Redundant packet
2 Redundant packets
3 Redundant packets
4 Redundant packets
Frequency f1, f2
Cadence on, off. The on and off duration in playing the tone
Frequency f1, f2
Cadence on, off. The on and off duration in playing the tone
(300 ~ 3000Hz)
(100 ~ 5000ms)
(300 ~ 3000Hz)
(100 ~ 5000ms)
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12.4. SIP COMMON
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Port and Header
Outbound Proxy
Setting
Port
Header
Form
Domain
Name
Port Control port number of SIP protocol.
Registrar Setting Domain
Name
Domain name or IP address of proxy that you want to register.
Out-band DTMF Control Enable/Disable
Enable: It “Disable” RFC 2833 DTMF
Default
The control port number of SIP protocol. 5060
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.
Domain name or IP address of proxy. Empty
Disable
5060
Empty
Disable
Disable
Incoming Call
Screening
NAT Signalling
Keep Alive
Target the media
(RTP)
Screening Disable: Accept all incoming SIP call
Enable: This gateway only accepts incoming call through SIP
Proxy.
Disable
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
Via Select the contact information (IP
Address, Port Number) to pass through
SDP
NAT device. Please refer to 11.7 Target the Media
SDP: via SDP
Symmetric RTP: via Symmetric RTP
Codecs Selection Codec
Type
G.729AB: Mark the selection to Enable
G.729AB Codec
Enable
G.723.1: Mark the selection to Enable Enable
G.723.1 Codec
55
SIP Entity
Public Address
Setting
PCMU: Mark the selection to Enable
PCMU Codec (G.711 u Law)
Default
Enable
PCMA: Mark the selection to Enable
PCMA Codec (G.711 A Law)
Enable
Priority your requirement.
G729-G723-P
CMU-PCMA
SIP Entity Select an entity and click Select button to display follow items’ setting of SIP entity section.
Select: Select Button
Entity
Control
Register: Register Button
De-Register: Cancel Register Button
Select Enable/Disable
Register
Status
Show the register status, if it shows
Registered means successful. (Read only)
Register: Register Button
De-Register: Cancel Register Button
1
Enable
Empty
CLIR Calling Line Identification Restriction
Disable: Send caller ID to SIP proxy when user make SIP call
Enable: Don’t send caller ID when user
Disable 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.
Address Enter SIP phone number of the port.
The phone number general assigned by
Empty
SIP service provider.
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Contact Address
Setting
RFC 2833 DTMF
Forward To
SIP Entity
Members
Account Proxy
Username: It may the same as your SIP number
Password: Password for Authentication
Confirm Password: Reconfirm
Password
Default
Current
Setting
2833
DTMF
2833 In
Use
Forward
Address
Display current setting of
Contact Address. It will be
(Read Only) the same as the
Username of Public
Address Setting at this page of web if that field is configured
Enable: Enable RFC 2833 DTMF. Never
Negotiate: Encode DTMF to message and decode it back at destination.
Never: Convert DTMF to voice and sent by RTP packets.
Display current status of
DTMF configuration.
(Read Only)
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.
No Answer: When the SIP entity is no
N/A answer about 30 seconds, the calls will be forwarded.
Channel Show the all channels Depend on gateways
57
Entity
Default
Show ‘+ ‘ means the SIP entity is for the Empty channel.
12.5. SIP OUTBOUND AUTHENTICATION
SIP Outbound
Authentication
Default
Maximum Maximum number of entries (Read Only) 50 allowed
(Read Only) 0 Entered Number of entries of authentication entered.
Entries List of entries (Read Only) Empty
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List
Update
Entry
Delete
Entry
Default
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
Enter the information of outbound authentication
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.
Empty
Delete the information of outbound authentication
Entity: Select an entity.
Realm: Domain name or IP address.
Empty
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12.6. SIP INBOUND ANTHENTICATION
SIP Inbound
Authentication
Realm
Maximum
Entered
Entries List
Maximum number of entries allowed
Default
Enter domain name, IP address or word Empty string.
(Read Only) 20
Number of entries of authentication entered.
Display the entries
(Read Only) 0
(Read Only) Empty
Entity: Which entity that you select.
Username: Username of authentication.
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Default
Update Entry Enter entries of authentication
Entity: Which entity that you select.
Username: Username of authentication.
Password: Password of authentication.
Empty
Confirm Password: Enter password again for confirmation.
Delete Entry Delete entries of authentication
Entity: Which entity that you want to delete.
Username: Username of authentication.
Empty
12.7. Dialing Plan
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DIALING PLAN
Dial In Rewriting
Rule
Maximum
Entered
Maximum number of (Read Only) 100 entries allowed
Number of entries of (Read Only) 1 authentication entered.
List
Add Dialing Plan Enter numbers. Example: 050. Empty
Delete Entry Enter numbers for delete. Empty
Control
Display the entries (Read Only) x
The default value “x“ means that all numbers that you dial will first go through SIP proxy.
Capacity
List
Digits dialed from VG3300 can be Disable rewrite to different digits and sent to SIP Proxy.
Enable/Disable
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.
VG3300
series user guide
Group Field
Transit call
Description Default Value
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
Checking RTP
Password
For Inbound
Transit
Maximum
Entered
Entries List
This gateway will check the RTP packet periodically to verify if the line is still alive. If no RTP
0 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
(Read only) 32 Display no. of password can be accepted
Display the no. of password had been entered
(Read only) 0
List the detail data of password had been entered
(Display) Only) Blank
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Group Field Description
Add Passwords Enter a new password, any combination of digits
(0~9), less than 9 characters. The password will be used at PINcode for auto answer function
Default Value
Blank
Delete
Passwords
Enter the password to be deleted, refer the detail data under Entries List
Blank
12.9. STUN
Section Item Default
STUN Server Control Enable or Disable STUN Server service. Disable
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Section Item Default
NAT WAN IP Address Input this NAT WAN IP helps you to pass through NAT without using STUN server.
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
(Read Only) 5 Maximum Maximum number of entries allowed
Entered
List
Number of entries of
STUN server that have been entered.
Display all of servers that have been entered.
(Read Only) 0
(Read Only)
Add
NAT Type
Mapping List
Delete
Type
Stun Refresh Time Interval
List
Add a stun server
IP Address: Enter IP address or Domain
Empty
Name
Port: Enter port number of service.
Empty Delete a stun server
IP Address: Enter IP address.
Port: Enter port number of service.
Display NAT type (Read Only) Unknown
It defines how long the device will send 30 a binding request packet with discard flag on to STUN server.
My ip/port: shows the private IP and port number.
Global ip/port: Display public IP and port number.
(Read Only) Empty
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12.10. CHANNEL
Type Phone: FXS Interface, connect to telephone set or Fax machine.
Line: FXO Interface, connect to phone line.
Default
Setting
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Channel
Control
For FXS port:
Bothway: Can make and accept IP call and PSTN call from this channel
Disable Disable all functions of this port.
For FXO port:
IN_Only: Accept calls from
PSTN only
Bothway: Accept call from
PSTN or call dial from FXS
Disable: Disable all functions of this port.
Current State Display the current state of this port. (Read only)
Enable/ Disable.
Do not
Disturb
Enable/Disable does not disturb function
Enable
Disable
Silence
Suppression
Enable/Disable the function.
2833 In use Yes
No
(Read only)
Enable
Join SIP
Entity
Select an Entity for SIP.
Both FXS and FXO ports can join SIP Entity
1
Connect
Device
Phone: Connect to this port is regular phone
FAX: Connect to this port is
FAX machine. Codec will be fixed on G.711 if SIP-based
T.38 codec negotiation fails.
Both FXS and FXO ports can select their Connect Device
Phone
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T.38 FAX Relay
Voice
Battery
Reverse
This mechanism will reverse the polarity promptly that help some PBX to identify the start and end of each call
ON: Enable the function
OFF: Disable the function
Auto Answer This unit auto answer the call from FXO
Disable: Disable Auto Answer
Enable: Enable Auto Answer
Enable w/ Pincode: Enable
Auto Answer and Pincode verification.
Call Waiting Call waiting function for answering two incoming SIP
VoIP phone calls
Enable: Enable call waiting
Disable: Disable call waiting
Control
Input Gain
Yes: Use T.38 as FXS protocol
No: Don't use T.38 as FAX protocol. If user send or receive
FAX by this port, gateway can use G.711 (PCMU, PCMA) to pass-through FAX, please refer
Adjust Voice input Gain
Output Gain Adjust Voice output Gain
OFF
Disable
Disable
No
0
0
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SIP Phone Book Maximum Maximum number of entries allowed
Default
Apply to Hotline Control Enable or Disable the hotline function to
VES-3302 Line or other SIP device to make
Disable hotline call.
(Read Only) 200
Entered Number of entries of phone books entered.
Entries
List
Display phone books
Index: Dialing number
Public Address: SIP account.
Port: Port number.
Via Proxy: Via proxy or not.
(Read Only) 0
(Read Only) Empty
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Update
Entry
Delete
Entry
Enter entries
Index: Enter dialing number
Public Address: Enter SIP account.
Port: Enter port number
Via Proxy: Select via Proxy or not
Delete entries
Index: Enter the index for delete.
Default
Empty
Empty
13. Use Private IP (Behind NAT)
Using a Private IP in a NAT Environment
The gateway is able to communicate with other gateways under a NAT environment using Private
IP addresses on the LAN side of your IP Sharing device. However you must configure the IP
Sharing device to treat the gateway as a Virtual Server using UDP port 5060,2000.
You will have to ask MIS personnel to enable the ports listed in the following table.
Packet Modes Using Ports
SIP Signal Packets UDP 5060
Gateway Signaling Port UDP 2000
Gateway RTP Base Port UDP 4000
FTP software upgrade TCP 21
Web management TCP 80
If you want to use private IP behind NAT and Proxy Server is in Internet, you must need to enable
STUN service. If the system is installed in VPN, it is not necessary to Enable Stun.
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14. File Management
14.1. File Types
The naming convention to the file type of VPS3302 is listed in the following table:
File Type Description File Name
SIP3302.CFG
SIP3304.CFG
SIP33XX.CFG
SIP3302.RUN
SIP3304.RUN
SIP33XX.RUN
SIP3302.WEB
SIP3304.WEB
SIP33xx.WEB
System configuration file
Executing file
Web file
File of system configuration
System Software
Page for web browser
14.2. Software Update
14.2.1. Software update via FTP
Preparation before Updating FIRMWARE
1. Power on the Conference Bridge
2. Get Windows based PC ready
3. LAN cable is well connected (for FTP)
4. Configure the IP, Subnet, and Default Gateway of this gateway and PC
5. Get the file of update “GW FIRMWARE” ready
71
Software Update by FTP for File Type RUN and WEB
1. Execute FTP Client Software, e.g. CuteFTP
Enter IP Address, User Name (default is FTP), Password (the password of FTP and
Console is same, and the default is blank), and the Port Number to 21 gateway will be displayed on the window if the connection is successful.
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3. Select the file with extension of .RUN and click button Upload and then Yes to overwrite.
(Please notice that the file name must be same as the file name in the Gateway, e.g.
SIP3304 .RUN
).
4. After the file is overwritten (you may check if the time of the file is updated), Gateway has to run Cold Start to store the configure file, then the updating is effective.
5. Select the file with extension of .WEB and click button Upload (Please notice that the file name must be same as the file name in the Gateway, e.g. SIP3302 .WEB
). And repeat the step 3 ~ 4.
6. Check if the uploading is successful, you enter the Web Management Page to examine the version of software. (Web Folder: BASIC\GENERAL)
Check if the version is
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05
06
07
08
09
10
11
15. Appendix
15.1. Appendix A: Phone-Set Command
Pick up the handset and listen for the dialing tone. Dial “##0000 and listen for three consecutive tones before setting the following parameters. After input the parameters, please dial ‘# to end the configuration.
Command Description
01 IP State
Parameters
0 : static; 1: DHCP; 2: PPPoE xxx*xxx*xxx*xxx
03 Subnet xxx*xxx*xxx*xxx
12
14
15
Primary DNS Server
IP
Second DNS Server
IP xxx*xxx*xxx*xxx xxx*xxx*xxx*xxx
Select Signaling Port 0~65535
Select RTP Base Port 0~65534 (limit to even port number only)
PPPoE username
PPPoE password
User name (use the mapping table to map character into digits)
Password (use the mapping table to map character into digits)
DND
SIP Forward State
Do not Disturb, this line accept dial out call only.
All incoming call is terminated. 0 : Disable ; 1:
Enable
0 : Disable ; 1: Enable; 2: Busy; 3: No Answer
Number
The SIP number that this line will forward to. The
Forward To address is "key in phone-set number@SIP proxy registered". For example,
, 73796 is the number you key-in by phone-set. fwd.pulver.com is the registered proxy of this gateway.
Change Service Port 1:FTP; 2:HTTP 3:Telnet (Port: 0-65535)
Change WEB 6 digits
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16
40
41
42
46
47
95
97
98
Password
Change FTP
Password
Listen for the IP
Address
Listen for the Subnet
Mask
Listen for the Default
Gateway
Listen for WEB, FTP,
Telnet Port
Listen for Current
Public Address
6 digits
(ending ”#” is not required)
(ending ”#” is not required)
(ending ”#” is not required)
1:FTP; 2:HTTP 3:Telnet
(ending ”#” is not required)
Region ID
Reset unit to Factory
Default values
2 digits
1: reset all; 2: keep IP
System Warm Restart 1: do it
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15.2. Appendix B: Console Command
User Exec commands
Enable
Exit
Help
Show
Turn on privileged commands
Exit from the EXEC
Description of the interactive help system
Show running system information show
Dns Show the IP address of domain name server ethernet history
FastEthernet port status and configuration
Display the session command history
Ip running-config version
Display IP configuration
Show current operating configuration
System hardware and software status
Privileged Mode
Configure Enter configuration mode
Disable
Exit
Help
Ping
Probe-hook
Probe-remove
Reload
Restart
Show
Global Mode
Dbflush
Dns
End
Exit
Help
Ip
Log
No pppoe regional_id service_port
Turn off privileged commands
Exit from the EXEC
Description of the interactive help system
Send echo request to destination probe busytone cadence stop probe busytone cadence
Halt and perform cold start
Halt and perform warm start
Show running system information
DataBase flush
Set the IP address of domain name server
Exit from configure mode to privileged mode
Exit from configure mode
Description of the interactive help system
Global IP configuration subcommands
Control log output
Negate a command or set its defaults
PPPoE configuration subcommands
Set regional id
Set service port number
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15.3. Specifications
Voice Interface
FXS interface
FXO interface
Connectors
Voice compression
Silence suppression
Echo cancellation
Jitter buffer
Gain control
Transport protocols
Call control protocol
Network Interface
Number of ports
Loop start, 2 wire
Feeding Voltage: 20V
Feeding Current: 30 mA
Loop start, 2 wire
RJ-11 Connectors (3304/3306)
IDC Connectors (3310/3318)
G.711/G.723/G.729AB
VAD, CNG
G.165/G.168 16ms
Adaptive jitter buffer management
In/Out +/-6db
RTP, RTCP
Pure SIP
Two Ethernet ports
VG3300
series user guide
General Spec
Dimension
Power
Power consumption
Working environment
EMI
PTT
Safety
VG3306: 172mm x 177mm x 35 mm
VG3310: 440mm x 44mm x 254 mm
VG3318: 440mm x 66mm x 254 mm
Voltage: 100-240 VAC, Frequency: 50/60 Hz
VG3306: 12W
VG3310/3716: 70W
Operating temperature: 0 to 50℃
Storage temperature: -10 to 70℃
FCC part 15 Class B . CE Mark
FCC part 68 , NALTE , iDA , JATE cUL , CCIB , CB
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15.4. Mapping table of characters used in PPPoE
Character Digits to key-in Character Digits to key-in
0 30 X 58
1 31 Y 59
2 32 Z 5*0
3 33 a 61
4 34 b 62
5 35 c 63
6 36 d 64
7 37 e 65
8 38 f 66
9 39 g 67
@ 40 h 68
A 41 i 69
B 42 j 6*0
C 43 k 6*1
D 44 l 6*2
E 45 m 6*3
F 46 n 6*4
G 47 o 6*5
H 48 p 70
I 49 q 71
J 4*0 r 72
K 4*1 s 73
L 4*2 t 74
M 4*3 u 75
N 4*4 u 76
O 4*5 w 77
P 50 x 78
Q 51 y 79
R 52 z 7*0
S 53 = 3*3
T 54 . 2*4
U 55
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15.5. Region ID
Country Region ID Country Region ID Country Region ID
Argentina 01 France 12 Singapore 36
Australia 02 Germany 13 Slovenia 38
Portugal 04 India 18 Spain 40
Brazil 05 Italy 22 Switzerland 42
Canada 06 Japan 23 Taiwan 43
China 07 Korea 24 Thailand 44
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15.5. Region ID
Country Region ID Country Region ID Country Region ID
Argentina 01 France 12 Singapore 36
Australia 02 Germany 13 Slovenia 38
Portugal 04 India 18 Spain 40
Brazil 05 Italy 22 Switzerland 42
Canada 06 Japan 23 Taiwan 43
China 07 Korea 24 Thailand 44
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