Contents 1. Introduction ...........................................................................................................................................4 Product Overview ...................................................................................................................................4 Items that you will need and things you will need to know .........................................................................5 NetComm SmartVoice Gateway Features ...........................
NTP (Network Time Protocol) .................................................................................................................70 System Operations (Save Settings) ........................................................................................................72 Software Upgrade.................................................................................................................................73 Logout ........................................................................
1. Introduction Product Overview The NetComm SmartVoice series of stand-alone VoIP Gateways carry both voice and facsimile over an IP network. They support the SIP industry standard call control protocol to be compatible with free registration services or VoIP service providers’ systems and they work in two different modes: UA (User Agent) or Server. As a standard user agent, the gateways are compatible with all well-known Soft Switches and SIP proxy servers.
Items that you will need and things you will need to know To correctly install and set up your VoIP Gateway you will need to ensure that the required broadband hardware and services are setup and configured in the appropriate way to integrate this equipment. Broadband Service • You will need access to a high speed broadband service via a RJ 45 (Ethernet) connection to a broadband modem. VoIP Service • You will need a VoIP account from your VoIP service provider.
NetComm SmartVoice Gateway Features WAN • One 10/100Mbps auto-negotiation, auto-crossover RJ-45 Ethernet port • Support static IP, PPPoE, Bigpond Cable and DHCP address assignment and dynamic DNS (DDNS) • QoS: IP TOS (Type of Services) and DiffServ (Differentiated Services) for both SIP signaling and RTP • NAT Traversal : Port Forwarding, STUN, UPnP and Outbound Proxy • NTP: (Network Time Protocol RFC 1305), Accepts up to 3 Time Server • Time Zone Support • MAC Address Clone • RTP Packet Summary : packet se
• Speed Dial • Local phone book for peer-to-peer calling • E.
General Specifications • Power Adaptor : AC 100V~240V 50/60Hz input, DC 12V output • Temperature : Operation 0 °C ~ 45 °C • Storage -25 °C ~ 75 °C • Humidity : up to 90% RH, non-condensing • 4-port Model Dimension (W/D/H) 4-port: 202 x 172 x 35 mm • 8-port Model Dimension (W/D/H) 4-port: 287 x 161 x 38 mm • 16-port Model Dimension (W/D/H) 4-port: 442 x 325 x 43 mm • 32-port Model Dimension (W/D/H) 4-port: 442 x 325 x 43 mm • 4-port Model Weight: 0.43Kg • 8-port Model Weight: 1.34Kg • 16-port Model Weight: 4.
SmartVoice Gateway Hardware Description Front Panel (all models) • Power Indicator: Green light indicates a normal power supply. • Run Indicator: Blinking green light indicates normal operation. • Alarm Indicator: When the system starts up, the red light will blink. It also indicates abnormal gateway behaviour. • P1 – PX stands for Port 1 – Port X (or Line 1 – Line X) • WAN stands for the WAN Port Indicator. • L1 – LX stands for the LAN Port Indicator.
Rear Panel (all models) Phone ports are telephone ports (FXS) to be connected to telephone sets or PBX CO ports. Line ports (FXO) are to be connected to PSTN lines (telephone lines from a PSTN network). NOTE: Do not connect FXS ports to each other. Also, do not connect any FXS port directly to a PSTN line or internal PBX. Doing so may damage VoIP gateway. Restore the Gateway To restore the gateway to its factory default setings (IP address, Username and Password only): (1) Disconnect the power plug.
2. Installation and Applications Connecting your SmartVoice Gateway(s) 1. Connect the WAN port of your SmartVoice Gateway to your Broadband Modem using an ethernet cable. 2. Connect the power adapter to your SmartVoice Gateway and plug it into a powerpoint. Please use the power adapter that comes with VoIP gateway. Using an adapter other than the one supplied with the VoIP gateway, may cause problems and will affect the warranty of the product. 3.
Network Connections Possible network connections are divided into 3 basic modes as described below: • Gateway can be assigned with a Public IP Address • Gateway can be installed under the existing NAT • Gateway can be assigned with a Public IP address and serve as an IP sharing router. Gateway Assigned with a Public IP Address The VoIP gateway will have a Public IP address for its Internet connection regardless of whether it is a static IP address, DHCP (using a Cable Modem), or PPPoE (Dialup / ADSL).
Gateway behind a NAT/Firewall Network The VoIP gateway uses a private IP address and the Internet sharing function of other systems to connect to the Internet. The following diagram shows a typical network connection in which broadband devices such as an ADSL router work in router (NAT) mode so that the gateway is behind a NAT/Firewall.
The table shown below lists possible solutions for different network scenarios. LAN IP address of Internet sharing Please avoid IP address 192.168.8.1-192.168.8.
Gateway Assigned with a Public IP Address and Serving as an Internet Sharing Device The SmartVoice gateway will have a Public IP address, regardless of whether it is the public IP from a static setting, DHCP (using a Cable Modem), or PPPoE (to connect to your ADSL account), and can use the built-in Internet sharing functions to allow other PCs attached to SmartVoice gateway LAN ports to share the Internet connection. The network connection is shown in following diagram.
Telephone Connection Example of SmartVoice Gateway using FXS interfaces SmartVoice Gateway connecting directly to phone sets After connecting telephone sets to Phone1-Phone4, users can make direct calls (Phone1-Phone4 are FXS interfaces). Each set acts as an independent extension line. Integrating the SmartVoice Gateway with PBX Phone1-Phone4 are FXS interfaces, and some of them can be connected to telephone sets for direct calls.
Example of SmartVoice Gateway using FXO interfaces SmartVoice Gateway connecting directly to the Telephone Line of a PSTN Line1-Line4 are FXO interfaces and can all be connected to a PSTN to serve as a bridge between the PSTN and VoIP network. Integrating the SmartVoice Gateway with PBX Line1-Line4 are FXO interfaces and can be connected with PBX extension lines (this is exclusively for an analog interface, and is not applicable for a digital interface).
Example of SmartVoice Gateway using a combination of FXS and FXO ports Phone1-Phone2 are FXS interfaces and can be directly connected to telephone sets for direct calls. Line1-Line2 are FXO interfaces and can be connected to PSTN lines to serve as a bridge between the PSTN and other VoIP telephones. The system also allows a call to be made from a traditional telephone line to connect with a user behind the SmartVoice Gateway.
3. Configuring the Gateway via WEB Browser The gateway also allows users to change gateway settings using a web browser. After opening a browser window, enter the gateway’s IP address as the website address in order to enter the Web configuration screen as shown in the following diagram. (IE Browser used for example: Enter http://192.168.1.2.) The factory default WAN IP address for Gateway is 192. 168. 1. 2. You can also enter ”101” from the handset to inquire about the current WAN Port IP address.
Network Settings The network settings are used to set the gateway’s communication ports, IP configurations, DNS and DHCP server etc. • Current WAN IP Address: The IP address of the WAN port. • Listen Port UDP: It is not necessary to change the protocol of the communication port used by the gateway, unless it conflicts with ports used by another device in your network. • RTP Starting Port UDP: The initial value of the port number for transmitting voice data among gateway(s). Each line requires 2 ports.
Setting Dynamic IP (DHCP) Click “DHCP” to obtain a Dynamic IP address, and then click the “Accept” button at the bottom of the screen. To save the settings click System Operation to select “Save Settings”, “Restart”, and then click the ”Accept” button. Wait for a while (about 40 seconds), and the system will obtain the related IP value from the DHCP Server. NOTE: After the system has obtained a new IP address, if using WAN Port to enter the Web Configuration Screen, a new IP address has to be used.
BigPond Click “BigPond Cable”. Enter User Name and Password and then click the “Accept” button at the bottom. DNS Settings Domain Name Server (DNS): While a gateway is accessing another gateway or computer with a hostname, it will look up the IP address from the DNS provided by the ISP. The ISP whilst negotiating with PPPoE or DHCP usually assigns the DNS information.
LAN interface mode • Router: The system serves as a router. • Bridge: The system serves as a bridge between WAN port and LAN port. (LAN default gateway will still be accessible for configuration). Network Settings (LAN) • Network Settings (LAN): The gateway’s LAN IP address and subnet mask value. Please note that the gateway is built under NAT: The gateway’s LAN IP address cannot be in the same section as the Internet sharing devices, or else it is unable to make or receive calls.
QoS Settings WAN QoS • QoS (Quality of Service): Sets true bandwidth of your Internet connection to ensure sound quality during transmission. (When this function is enabled, the voice packet has the highest priority to ensure telecommunication quality while less bandwidth is assigned for data transmission). Some models of the VoIP gateway without this function can adjust the bandwidth automatically.
NAT/DDNS (NAT Traversal) If a gateway is set up behind an Internet sharing device, you can select either the NAT or STUN protocol. • NAT Public IP: The IP address used by the gateway should be a private address. Furthermore, users must set the Virtual Server Mapping in the Internet sharing device (For example, a virtual server is usually defined as a Service Port, and all requests to this port will be redirected to this specified server’s private IP address).
DDNS These settings are only necessary when the gateway is set up behind an Internet sharing device that uses a dynamic IP address and does not support DDNS. You need to apply for an account with a DDNS service from www.dyndns.org before you type in the following information. • Server address: Sets up the IP address or URL (Uniform Resource Locator) of the DDNS Server. • Hostname: The URL of the system (or NAT) – apply from domain name registration providers (e.g. www.dyndns.org).
Telephony Settings Prefix Number Rules • Trunk Dial Out Verify/ Trunk Dial Out Replace: The system will transfer the number for all transit out calls through the FXO port. For example: If you transit out with 01907123456, the system will replace it with 190601 907123456. If you transit out with 008621123456 the system will replace it with 190200 8621123456. • Trunk Dial Out Deny: The system will deny the call with the leading number filled in this column.
• FXO Hunting Default Dial-Out: This will take effect as FXO Hunting VoIP call in option is set to Default Dial-Out. • FXO Line VoIP call in options: The usage of this option is as same as FXO Hunting VoIP call in option. The Default Dial-Out refers to individual lines rather than a hunting group. The individual FXO line default dial out number can be set in Line configuration section as below.
• Group Hunting: Select group hunting when there is an incoming call and the gateway will automatically assign an unassigned call according to the Hunting Priority. If Line 2 does not want to be set as an assigned line to receive any inbound calls, the function can be disabled. Users can also use the Up or Down key to adjust hunting priority (No setting is required for the FXO interface). • Enable FAX: Enable this line to detect if there is a FAX tone to transfer the codec.
SIP Settings • All Call through OutBound Proxy: An outbound proxy server handles SIP call signaling as a standard SIP proxy server would. Furthermore, it receives and transmits phone conversation traffic (media) in between two communication parties. This option tells the gateway to send and receive all SIP packets to the destined outbound proxy server rather than the remote gateway. This helps VoIP calls to pass through any NAT protected network without additional settings or techniques.
E.164 • International Call Prefix Digit: Enter the International call prefix. • Country Code: Select the desired country code. • Long Distance Call Prefix Digit: The long-distance prefix digit for making a long-distance call. • Area Code: Please enter the area code. • E.164 Numbering: To invite Proxy to follow the E.164 rule. It depends on the Proxy. NOTE: All settings in this section are specific to your VoIP network. Please ask your VoIP service provider whether they require these settings.
• Proxy Server Realm: This is used for gateway SIP account authentication in a SIP server. In most cases, the gateway can automatically detect your SIP server realm. So you can leave this option as blank. However, if your SIP server requires you to use a specific realm you can manually type it in here. If you fail to make a call, please contact your VoIP service provider. • TTL: Enter the desired time interval at which the gateway will report to your Proxy Server.
FXS/ FXO Representative number registers to Proxy: Assuming that your registered ID and password are individual, the settings should be as above: • FXS Representative Number: Register all FXS ports as a hunting group. • FXO Representative Number: Register all FXO ports as a hunting group. All the grouped FXO ports will be hunted automatically. It is available when FXO registers to Proxy. • Register: Register to Proxy if ticked.
Private Network Users can establish a private network by using the Phone Book Manager Service. The Phone Book Manager Service is different from using the Proxy. The gateway is able to register with the Phone Book Manager Service and the SIP Proxy at the same time. Server Settings • Enable Phone Book Manager Server: It allows other Gateway users to register the IP address and Gateway Number in this Phone book manager server. It is recommended to use a static IP for the Phone Book Manager Server.
NOTE: When the gateway is configured as Phone Book Manager server, it can also be configured as a client at the same time by enabling Register to Phone Book Manager option. This means the gateway also shares its Private Gateway Number to other clients which register on it. Private Network Numbers • Gateway Number: A self-defined phone number for the gateway that the gateway will use as its phone number to register on a Phone Book Manager server.
Calling Features • Do Not Disturb: It will only be able to call out when it is enabled. All incoming calls will be restricted. • Unconditional Forward: All incoming calls will be forwarded to the “Forwarding Number” automatically. If it forwards to FXO, it only make FXO hook off, not make FXO dial out. • Busy Forward: Forward the incoming call to “Forwarding Number” when the port is busy. • No Answer Forward: Forward the incoming call to “Forwarding Number” after ring timeout expires without answer.
Advanced Options NOTE: There are two levels to enter Web. Administrator is able to change all settings. Web UI only changes some settings. • Administrator’s name and Password: Enter administrator name and password, which has the highest level of control of the gateway. • Web UI Login ID and Web UI/IVR Password: Enter login ID and password when you log into the Web interface/IVR of the gateway as a normal user.
• Payload Type: payload type of RFC2833. • Uses Second CPT for VoIP Call: This function is usually applied when the user selects VoIP as the primary path for outgoing calls and PSTN as the backup. By enabling this function, the gateway will generate a different set of tones to inform the user that VoIP is in service. Should VoIP fail and fallback to PSTN, the user will hear PSTN tones instead of the second set CPT. (for CPT related settings, please refer to Trunk Management -> CPT/Cadence Settings).
Line Settings • Listening Volume: Adjusts the hearing volume. • Speaking Volume: Adjusts the speaking volume. • Tone Volume: This setting will be applied to all tones generated by the VoIP gateway including Dial Tone, Busy Tone, and so on. • Flash Time: FXS: Used to adjust the detection period of flash signal from the phone set connected to the FXS port. For example, if pressing the FLASH key disconnects a call, increase the “Flash Detect Time” to fix this issue.
Codec Settings • Preferred Codec Type: Since different voice codecs have different compression ratios, so the sound quality and occupied bandwidths will also be different. It is recommended that you use the default provided (G.729) because it is popular and will provide better sound quality. • Jitter Buffer: Adjusts the jitter to receive a packet. If the jitter range is too wide, it will delay voice transmission.
Fax Settings • T.38: The T.38 protocol is used for better and faster facsimile transmission. So it is recommended to enable this function to gain better fax quality. When this function is enabled, please select UDP or TCP. If selecting TCP and some gateways cannot use the Fax function, please select UDP instead. NOTE: When a FAX tone is detected in a call, the gateway will automatically switch from voice mode to fax mode.
Drop Inactive Call • Silence Detection Threshold: The volume below the threshold is used as a standard to determine whether or not to hang up the phone. • Drop Silent Call Timeout: If the detected volume is below the threshold and the time exceeds the silence detection interval, the system will hang up the phone automatically to avoid keeping the line engaged. NOTE: Please be careful with these setting. Improper values might cause unexpected automatic disconnection of a call.
Digit Map There are 50 sets of leading digit entries to choose voice routing interface – Auto select, PSTN or VoIP. • Default Call Route: The default call route can be Auto, VoIP, PSTN and Deny. Auto (VoIP first): The call route is VoIP first, and the next is PSTN. VoIP: The call route is VoIP only. PSTN: The call route is PSTN only. Deny: The call will be deny if the dial-out number is not in the table. • Enable: Enable detection of this entry.
Phone Book The system can set up and store 100 phone numbers in the phone book and provide an IP address query when calling other Gateway(s). If no Phone book manager is set within a Gateway group, then all Gateway systems have to set up phone data for each VoIP gateway to communicate with each other. • Gateway Name: Enter the Gateway’s code or an easy-to-remember name. • Gateway Number: Enter the desired Gateway number.
Speed Dial This system can set up 100 numbers for speed dialing. Setting methods are as follows: Method 1- Single mapping: Fill a short code into the “Speed Dial Code” column, and enter the desired phone number into the “Number To Dial” column. For example, pick up the handset and dial 55# and the system will dial 32568791. Method 2- Multi mapping: Fill the prefix code into the “Speed Dial Code” column and the format to transfer into the “Number To Dial” column.
Caller Filter This function is used to allow or deny the SIP Invite message from the Proxy list ONLY. • Filter IP Address: Enter the IP address that you would like to allow/deny. • Subnet mask: Enter the subnet mask that is applied to the IP that entered in the Filter IP address field.
Transit Call Control If you wish to restrict a general user (one who is not required to enter the PIN code) to local calls only and prohibit him/her from making long-distance calls started with a prefix “0”, do the following steps: 1. Enable the Outbound Call Control function, 2. Set the PIN code for Outbound Level 5 to blank, 3. Set the Long-Distance Control Table to correspond with the Outbound Level 5 to prohibit making any call with the prefix “0”.
Dialing principle to enable PIN code *Inbound Call Control PIN Code * Outbound Call Control PIN Code * Phone number Using * to the separate PIN code and the phone number based on actual settings. The PIN can also be omitted.
Long-Distance Control Table This table controls the level of authority of an outgoing (transit out) call that is dialed through FXO and diverted to PSTN, as below: Descriptions: • Digit strings in this table are prefixes that the gateway will check on dialed numbers in transit out calls. • This table is used to prohibit dialing any numbers started with specified prefixes.
CPT/Cadence Settings CPT/Cadence setting parameters serve as the basis of an FXO interface to determine whether or not a PSTN-call receiving party has hung up the phone. If the following parameters differ from the parameters of the actual assigned lines, it could cause the FXO to continue to engage a line. Busy Tone Cadence Measurement • Busy Tone Cadence Measurement and auto learning: Provides a solution of FXO integrated with PSTN or PBX. FXO will learn the busy tone automatically.
NOTE: To cope with different local PSTN and different PBX models, the system provides auto busy tone cadence learning to prevent the FXO from engaging a line. However, if the line of the receiving party is engaged and his/her PSTN uses a voice prompt to replace the traditional beep sound, then the system would not be able to detect a busy tone. Silence detection should then be used to determine whether or not to end the call. CPT Auto Detect • 2 PSTN phone numbers or 2 PBX extension lines are needed.
Direct Connection to PSTN Note: The above diagrams show a 4-port SmartVoice gateway for illustration purposes. An 8, 16 or 32 port SmartVoice gateway could also be used depending upon your network requirements. • Connect one of the trunk lines to the Gateway FXO Port. • The line of “Dial Number” (36008913) must be on-hook. • Detect Channel: Enter 3 (if the trunk line is connected to P3, and uses P3 for outgoing detection). • Phone Number: Enter the number of the FXO line.
Once detection of a busy tone is in progress, the system will dial the number to be tested (in this case 36008913). After it rings pick up the phone and enter”#”, then hang up. The system will then detect a busy tone automatically.
Connected to a PBX Extension Line If the gateway is connected to a PBX extension line, then the busy tone of both the PBX and the PSTN must be detected. Connecting to a PBX extension line and detecting the busy tone of the PBX 307 36008913 301 Note: The above diagrams show a 4-port SmartVoice gateway for illustration purposes. An 8, 16 or 32 port SmartVoice gateway could also be used depending upon your network requirements. • Connect one of the PBX extension lines to the Gateway FXO Port.
Connected to a PBX extension line to detect the busy tone of PSTN 307 36008913 36008914 301 Note: The above diagrams show a 4-port SmartVoice gateway for illustration purposes. An 8, 16 or 32 port SmartVoice gateway could also be used depending upon your network requirements. • Connect one of the PBX extension lines to the gateway’s FXO Port. • Detect Channel: Enter 3 (if the trunk line is connected to P3, and uses P3 for outgoing detection).
Filling in the CPT Table Fill in the table after the detection is completed as below, where the values are the frequency and Onand-Off ratio detected. Please click the “Accept” button. If connecting the Gateway to a PBX extension line, please do not set the detected busy tone of the PBX and the PSTN in the same set, otherwise the value detected the first time will be overwritten. Save Settings Save the settings and restart the system after the test is completed. Then click the “Accept” button.
Current Status and System Information These two pages show that the status of VoIP Gateway. There are Port Status, Server Registration Status, WAN Port Information, LAN Port Information and Hardware. • Port Status: It includes if each port registers to Proxy successfully, the lasted dialed number, how many calls each port had since the system is start, etc. • Server Registration Status: It shows the registration status of DDNS, Phone Book Manager, STUN and UPnP.
RTP Packet Summary Displays the information of the last finished call. It contains peer IP, peer port, packets sent, packet received and packet lost. Press the Refresh button to get the latest RTP Packet Summary. Voice quality can be degraded significantly if packet lost exceeds 5%. To avoid high packet loss, you need either to increase your broadband bandwidth or configure your QoS in your broadband router.
STUN Inquiry Using “STUN Inquiry” to detect your IP sharing device’s NAT type and communication between STUN server and client (built-in SmartVoice gateway). NOTE: If detected NAT type is “Symmetric NAT”, then the gateway is not able to traverse the NAT of your Internet sharing devices. This is not a flaw of the gateway’s design, but nature of STUN protocol design.
Ping Test Use “ping” to identify if the remote peer is reachable. Fill in remote IP address and click “Test” will start the test.
Port Filtering Port filtering enables you to control all data that can be transmitted in routers; principles of filtering--When the port used at the source end is within the limited scope, it will be filtered without transmission. • Enable port filtering: whether to enable this function or not. • Port Range: Set the range of port to be filtered, suppose it is 80 and when use protocol is Both or TCP, all computers will be unable to use the services of http (port 80)— will be unable to browse normal Web pages.
IP Filtering IP Filtering is to limit internal users from accessing the Internet. • IP: Input the IP address that you want to filter; the limited IP address will be unable to transmit the data to the Internet. • TCP/UDP: Choose to either filter TCP or UDP, or choose to filter both. • Remark: Remark filed, you can write comments by yourself.
MAC Filtering MAC (Media Access Control) address filtering is to filter the transmission of data by network card physical address. MAC: input MAC that will be limited to accessing Internet.
Virtual Server Enabling the users on Internet to access the WWW, FTP and other services under your NAT. When remote user are accessing Web or FTP servers through WAN end IP address, it will be routed to the server at the internal LAN end and be routed to the server at the internal LAN end as appropriate in accordance with the externally required services. • WAN Port Range: Input the port on the WAN side. • TCP/UDP: Select the communication protocols used by the server—TCP or UDP.
DMZ Lets the server on the LAN to be directly exposed to the Internet for accessing data. Either this function or the virtual server can be selected for use.
URL Filter URL filter is used to deny device from LAN accessing specific web sites. The system will block the URL that contains the string.
Special Applications Provide multiple connections for special applications. • Name: The name of the special application. • Incoming Type: The protocol used by the special application. • Incoming Port range: Port range on the WAN side that will be used to access the application. • Trigger Type: The protocol used to trigger the application. • Trigger Port Range: Port range used to trigger the application.
DoS Prevention Settings • Enable DoS Prevention: To prevent DoS from WAN. • Enable DoS Prevention on LAN: To prevent DoS from LAN. • Packet/Second: If the same type of packets received in one second is more than the specified value, then it will be treated as attacks. And VoIP gateway will block the IP if you checked “Enable Source IP Blocking”.
• Enable Source IP Blocking: Block the IP. • Blocking Time: The time to block the IP.
NTP (Network Time Protocol) Time Zone: Set the Time Zone where VoIP gateway resides. Time Server #1~#3: Set the Time Server where VoIP gateway should sync up during start up. (NTP protocol) Backup and Restore (Configuration) You can backup settings to a file and restore settings from that file. You also can restore all settings back to default by selecting Restore Default Configurations and click Restore.
Provision Settings Options in this section are only required for VoIP networks in which provisioning system has implemented. Fill in the parameters needed of Provision Server from your service provider.
System Operations (Save Settings) Some settings are effective by Restart. Remember to save all settings by Save Settings before to restart. • Save Settings: Save settings after completing. The new settings will take effect after the system is restarted. Please select “Save Settings”. • Restart: If it is necessary to restart the system, please select “Restart” and click the “Accept” button.
Software Upgrade Gateway provides software upgrade function from a remote end. Please consult your service provider for all following details. • Upgrade Server: Choose the server type. • Software Upgrade Server IP: Enter the software server IP address. • Software Upgrade Server Port: Enter the port that server uses. TFTP is 69, FTP is 21 and SmartVoice gateway upgrade server is 6001. • User Name/ Password: The account to access FTP server. • Directory: The path of TFTP or FTP.
Logout Gateway only allows one user to login at a time, so whenever a change is made, please save the settings, restart the system, or logout to avoid the situation where other users cannot login to change settings.
4. Configuring the Gateway via IVR Preparation • Install the Gateway according to the instructions as described in Chapter 2. • If a static IP is used, confirm the desired IP settings of the WAN Port (IP address, Subnet Mask, and Default gateway). Please contact your local Internet Service Provider (ISP) if you have any questions. • If using dialup ADSL (PPPoE) for network connection, confirm your broadband account name and password.
IVR (Interactive Voice Response) The gateway provides a convenient IVR function. Users only need to pick up a handset and enter the function code for the query and setting. A PC is not required. NOTE: After finishing the settings, make sure the new settings are saved. This is so that the new settings will take effect after the system is restarted. Instructions FXS Port: 1. Connected the Gateway to the telephones as required. 2. To access the gateway via IVR, pick up the handset and press ***[password]#.
IVR Functions Table Function Code Description Example 111/101 WAN Port IP address Set/Query 112/102 WAN Port Subnet Mask Set/Query Use in conjunction with function code 114, select 1 for a Static IP function. 113/103 WAN Port Default Gateway Set/Query 114/104 Current Network IP Access Set/Query (1: Static IP, 2.DHCP, 3.
Function Code Description 215/205 Set/Query Gateway Telephone Number (Representative Number) 216/206 Set/Query the extension number of Line 1. 217/207 Set/Query the gateway web configuration interface port number 109 Restoring factory default IP address configuration Example A static IP address for WAN Port IP: 192.168.1.2 Mask: 255.255.255.0 Gateway: 192.168.1.
IP Configuration Settings (WAN port) Static IP Settings NOTE: Complete static public/private IP settings should include a static IP (Option 1 under 114), IP address (111), Subnet Mask (112), and Default Gateway (113). Please contact your local Internet Service Provider (ISP) if you have any questions. Function Command Select a Static IP • After entering IVR mode, dial 114. • After hearing “Enter value”, dial 1 (to select static IP) • After entering IVR mode, dial 111.
Select PPPoE • After entering IVR mode, dial 114. • After hearing “Enter value”, dial 3 (to select PPPoE). PPPoE Account Settings • After entering IVR mode, dial 121. • After hearing “Enter value”, enter the account number, followed by ”#”. Example: If the account is “0284943122@isp.com.au”, please enter 0208 04 09 04 03 01 02 02 71 49 59 56 72 43 55 53 #. Please note that it is necessary to enter two digits for each character/number; for example, enter “01” for “1” and “11” for “A”.
PPPoE Character Conversion Table Number Input Key Input Key 00 Upper Case Letter A Input Key Symbol Input Key 11 Lower Case Letter A 0 41 @ 71 1 01 B 12 B 42 • 72 2 02 C 13 C 43 ! 73 3 03 D 14 D 44 “ 74 4 04 E 15 E 45 $ 75 5 05 F 16 F 46 % 76 6 06 G 17 G 47 & 77 7 07 H 18 H 48 ‘ 78 8 08 I 19 I 49 ( 79 9 09 J 20 J 50 ) 80 K 21 K 51 + 81 L 22 L 52 , 82 M 23 M 53 - 83 N 24 N 54 / 84 O 25 O 55 : 85
5. Dialing and dialed number routing principle Instruction • After a phone number is entered, dial # to call out immediately or, wait until the “Inter DTMF Timeout” expires (defined in “Advanced Options”, default=4 seconds). • If the phone number fits the setting of Digit Map, the gateway dials out the phone number through the assigned interface automatically. • The phone number should have at least 2 digits (not including * and #).
Start Enter a phone number (D#) Dial the number defined in SpeedDial table Yes Is (D#) defined in Speed Dial table? No Is (D#) defined in Extension table? Yes No Is (D#) defined in Phonebook table? Yes No Is (D#) defined in Phonebook Manager? Yes No Is (D#) defined in SIP proxy server? Yes No Dial (D#) through the first available FXO port to PSTN Yes Does this gateway has an FXO port? Dial out as defined in the first match case through the gateway No End SmartVoice Gateway(s) User Guid
Appendix A: Glossary 10BASE-T A designation for the type of wiring used by Ethernet networks with a data rate of 10 Mbps. Also known as Category 3 (CAT 3) wiring. See also data rate, Ethernet. 100BASE-T A designation for the type of wiring used by Ethernet networks with a data rate of 100 Mbps. Also known as Category 5 (CAT 5) wiring. See also data rate, Ethernet. ADSL Asymmetric Digital Subscriber Line. The most commonly deployed type of DSL for home users.
DHCP relay Dynamic Host Configuration Protocol relay. A DHCP relay is a computer that forwards DHCP data between computers that request IP addresses and the DHCP server that assigns the addresses. Each of the My ADSL Modem’s interfaces can be configured as a DHCP relay. See DHCP. DHCP server Dynamic Host Configuration Protocol server. A DHCP server is a computer that is responsible for assigning IP addresses to the computers on a LAN. See DHCP.
hop When you send data through the Internet, it is sent first from your computer to a router, and then from one router to another until it finally reaches a router that is directly connected to the recipient. Each individual “leg” of the data’s journey is called a hop. hop count The number of hops that data has taken on its route to its destination. Alternatively, the maximum number of hops that a packet is allowed to take before being discarded , See also TTL.
Microfilter In splitterless deployments, a microfilter is a device that removes the data frequencies in the DSL signal, so that telephone users do not experience interference (noise) from the data signals. Microfilter types include in-line (installs between phone and jack) and wall-mount (telephone jack with built-in microfilter). See also splitterless.
protocol A set of rules governing the transmission of data. In order for a data transmission to work, both ends of the connection have to follow the rules of the protocol. remote In a physically separate location. For example, an employee away on travel who logs in to the company’s intranet is a remote user. RIP Routing Information Protocol The original TCP/IP routing protocol. There are two versions of RIP: version and version II.
TFTP Trivial File Transfer Protocol. A protocol for file transfers, TFTP is easier to use than File Transfer Protocol (FTP) but not as capable or secure. TTL Time To Live A field in an IP packet that limits the life span of that packet. Originally meant as a time duration, the TTL is usually represented instead as a maximum hop count; each router that receives a packet decrements this field by one. When the TTL reaches zero, the packet is discarded.
Appendix B: Cable Information This cable information is provided for your reference only. Please ensure you only connect the appropriate cable into the correct socket on either this product or your computer. If you are unsure about which cable to use or which socket to connect it to, please refer to the hardware installation section in this manual. If you are still not sure about cable connections, please contact a professional computer technician or NetComm for further advice.
Straight and crossover cable configuration There are two types of the wiring: Straight-Through Cables and Crossover Cables. Category 5 UTP/STP cable has eight wires inside the sheath. The wires form four pairs. Straight-Through Cables has same pinouts at both ends while Crossover Cables has a different pin arrangement at each end. In a straight-through cable, wires 1,2,3,4,5,6,7 and 8 at one end of the cable are still wires 1~8 at the other end.
Appendix C: Registration and Warranty Information All NetComm Limited (“NetComm”) products have a standard 12 month warranty from date of purchase against defects in manufacturing and that the products will operate in accordance with the specifications outlined in the User Guide. However some products have an extended warranty option (please refer to your packaging).
Customer Information ACA (Australian Communications Authority) requires you to be aware of the following information and warnings: (1) This unit shall be connected to the Telecommunication Network through a line cord which meets the requirements of the ACA TS008 Standard. (2) This equipment has been tested and found to comply with the Standards for C-Tick and or A-Tick as set by the ACA. These standards are designed to provide reasonable protection against harmful interference in a residential installation.
Product Warranty The warranty is granted on the following conditions: 1. This warranty extends to the original purchaser (you) and is not transferable; 2. This warranty shall not apply to software programs, batteries, power supplies, cables or other accessories supplied in or with the product; 3. The customer complies with all of the terms of any relevant agreement with NetComm and any other reasonable requirements of NetComm including producing such evidence of purchase as NetComm may require; 4.
