THE SPECIFICATIONS AND INFORMATION REGARDING THE PRODUCTS IN THIS MANUAL ARE SUBJECT CHANGE WITHOUT NOTICE. STATEMENTS, INFORMATION, RECOMMENDATIONS IN THIS MANUAL ARE BELIEVED TO BE ACCURATE BUT ARE PRESENTED WITHOUT WARRANTY OF ANY KIND, EXPRESSED OR IMPLIED. USERS MUST TAKE FULL RESPONSIBILITIES FOR THEIR APPLICATION OF THE PRODUCTS.
SIP Operation Manual V2.6 Contents 1. Introduction....................1 Product Overview............................1 Product Features............................2 Hardware Description..........................3 2 ports gateway model: 2S / 2O / 1S1O ..................3 4 ports gateways model: 4S / 4O / 2S2O / 3S1O ................6 8 ports gateways model: 8S / 8O / 6S2O / 4S4O ................
SIP Operation Manual V2.6 RTP Packet Summary..........................54 STUN Inquiry............................. 54 Ping Test ..............................55 Time Settings ............................55 System Operations (Save Settings)......................55 Software Upgrade ............................. 56 Logout ............................... 56 5. IP Sharing Functions...................57 6. Coding Principle ..................60 Instruction..............................60 Dialed Number Processing Flow.......................
SIP Operation Manual V2.6 1. Introduction Product Overview The stand-alone VoIP Gateway carries both voice and facsimile over the IP network. It supports SIP industry standard call control protocol to be compatible with free registration services or VoIP service providers’ systems. It works in two different modes: UA (User Agent) or Server. As a standard user agent, it is compatible to all well-known Soft Switches and SIP proxy servers.
SIP Operation Manual V2.6 Product Features ● SIP (RFC 3261) compliant ● Supports 2 / 4 / 8 simultaneous FAX / Voice calls (port number differs between models) ● 4 Ethernet switch ports with IP sharing functions ● Optional server enables small businesses to build up private VoIP network (SIP model) ●...
SIP Operation Manual V2.6 Hardware Description 2 ports gateway model: 2S / 2O / 1S1O Front Panel WAN, LAN indicators Voice ports indicators Status indicators VoIP Gateway Alarm Power 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.
SIP Operation Manual V2.6 2S Model (2 FXS ports) RESET LAN ports 1 ~ 4 FXS ports 1,2 To reset the (built-in Ethernet switch) (telephone connectors) gateway Connect LAN hosts here Phone sets connection Or to restore to share WAN connection. ports factory settings IP sharing features enabled...
SIP Operation Manual V2.6 1S1O Model (1 FXO and 1 FXS ports) FXS port : 1 RESET LAN ports 1 ~ 4 FXO port : 2 To reset the (built-in Ethernet switch) gateway Connect LAN hosts here FXS connects to phone set; Or to restore to share WAN connection.
SIP Operation Manual V2.6 4 ports gateways model: 4S / 4O / 2S2O / 3S1O Front Panel WAN, LAN indicators Voice ports indicators Status indicators VoIP Gateway Alarm Power 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.
SIP Operation Manual V2.6 4S Model (4 FXS ports) RESET FXS ports 1 ~ 4 LAN ports 1 ~ 4 To reset the (telephone connectors) (built-in Ethernet switch) gateway or to Connecst to phone sets Connect LAN hosts here restore to share WAN connection.
SIP Operation Manual V2.6 2S2O Model (2 FXS and 2 FXO ports) RESET FXS ports 1,2 LAN ports 1 ~ 4 To reset the (built-in Ethernet switch) FXO ports 3,4 gateway Connect LAN hosts here FXS to telephone set; Or to restore to share WAN connection.
SIP Operation Manual V2.6 8 ports gateways model: 8S / 8O / 6S2O / 4S4O Front Panel Voice ports indicators WAN, LAN indicators Status indicators VoIP Gateway Alarm Power 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.
SIP Operation Manual V2.6 8S Model (8 FXS ports) LAN ports 1 ~ 4 RESET FXS ports 1 ~ 8 (built-in Ethernet switch) To reset the Connect LAN hosts here (telephone connectors) gateway to share WAN connection. Connects to phone sets Or to restore IP sharing features enabled factory settings...
SIP Operation Manual V2.6 6S2O Model (6 FXS and 2 FXO ports) LAN ports 1 ~ 4 RESET FXS ports 1 ~ 6 FXO ports 7,8 (built-in Ethernet switch) To reset the Connect LAN hosts here (telephone connectors) (PSTN line connectors) gateway to share WAN connection.
SIP Operation Manual V2.6 2. Installation and Applications Network Interface The network interface is divided into 3 basic modes as described below: Gateway can be assigned with a Public IP Address Gateway can be built under the existing NAT Gateway can be assigned with a Public IP address and serves as an IP sharing router. Gateway Assigned with a Public IP Address The gateway will have a Public IP address for Internet connection regardless of whether it is a static IP address, DHCP (using a Cable Modem), or PPPoE (Dialup / ADSL).
SIP Operation Manual V2.6 Gateway in a NAT network The gateway uses a virtual IP address and the IP sharing function of other systems to connect to the Internet. LAN IP address of IP sharing Please avoid IP address 192.168.0.1-192.168.8.254 (You may need to change the settings of IP sharing or change SIP series Gateway LAN Port IP address) Gateway IP Settings...
SIP Operation Manual V2.6 Gateway assigned with a Public IP Address and serving as an IP sharing device The gateway will have a Public IP address regardless of whether it is a static IP application, DHCP (using a Cable Modem), or PPPoE (To connect to your ADSL account), which can then use the functions of built-in IP sharing function to allow other PCs to be on-line at the same time.
SIP Operation Manual V2.6 Telephone Interface Description (4 ports model used in the example) Example for 4S gateway: 4S gateway connecting directly to phone sets After connecting telephone sets to P1-P4, users can make direct calls,(P1-P4 are FXS interfaces). Each set acts as an independent extension line.
SIP Operation Manual V2.6 Example for 4O gateway: 4O model connecting directly to the Telephone Line of a PSTN P1-P4 is FXO interfaces and can all be connected to a PSTN to serve as a bridge between the PSTN and other VoIP telephones. The system also allows a call to be made from a traditional telephone line to connect with a user behind the Gateway.
SIP Operation Manual V2.6 Example for 2S2O gateway: P1-P2 is FXS interfaces and can be directly connected to a telephone set for direct calls. P3-P4 is FXO interfaces and can be connected to a PSTN to serve as a bridge between the PSTN and other VoIP telephones.
SIP Operation Manual V2.6 3. Setting the Gateway through IVR VoIP transmits voice data (packet) via the Internet to achieve telecommunications. This means that the telecommunication quality is closely related to the whole network environment. If any one of the telecommunicating parties has insufficient bandwidth or frequent packet loss, the telecommunication quality will be poor.
SIP Operation Manual V2.6 Instructions FXS Port: Connected to telephones. To enter IVR mode, enter “ * * password #” after hearing the dial tone. When you hear a second dial tone, the system is in IVR mode, enter the function code.
SIP Operation Manual V2.6 IVR Functions Table: Function Code Description Example 111/101 WAN Port IP address Set/Query Use in conjunction with function code 114, select 1 for a Static IP function. 112/102 WAN Port Subnet Mask Set/Query 113/103 WAN Port Default Gateway Set/Query 114/104 Current Network IP Access Set/Query (1: Static IP, 2.DHCP, 3.PPPoE)
SIP Operation Manual V2.6 Function Code Description Example 215/205 Set/Query Gateway Telephone Number (Representative Number) 216/206 Set/Query the extension number of Line 1. 217/207 Restoring factory default IP address configuration A static IP address for WAN Port IP：192.168.1.2 Mask：255.255.255.0 Gateway：192.168.1.254 Restoring factory default settings Save settings Setting IVR and the language used on the Web GUI...
SIP Operation Manual V2.6 IP Configuration Settings—Setting IP Configuration of WAN Port Static IP Settings NOTE: Complete static IP settings should include a static IP (Option 1 under114), IP address (111), Subnet Mask (112), and Default Gateway (113). Please contact your local Internet Service Provider (ISP) if you have any questions.
SIP Operation Manual V2.6 PPPoE Account Settings After entering IVR mode, dial 121. After hearing “Enter value”, enter the account number, followed by ”#”. Example: If the account is “84943122 @ hinet.net”, please enter 08 04 09 04 03 01 02 02 71 48 49 54 45 60 72 54 45 60 #.
SIP Operation Manual V2.6 PPPoE Character Conversion Table Number Input Key Upper Case Input Key Lower Case Input Key Symbol Input Key Letter Letter • " & < >...
SIP Operation Manual V2.6 4. Setting a Gateway with WEB Browser The gateway allows users to make settings using a web browser. After opening a browser, enter Gateway’s IP address as the website address in order to enter the Web configuration screen as shown in the following diagram.
SIP Operation Manual V2.6 Network Settings The network settings are used to set the gateway’s communication ports, IP configurations, and Phone Books Manager IP etc. Current WAN IP Address: The IP address of WAN port. Listen Port UDP: It is not necessary to change the protocol of the communication port used by VoIP gateway.
SIP Operation Manual V2.6 RTP Starting Port UDP: The initial value of port number for transmitting voice data among Gateway(s). Each line requires 2 ports. It is not necessary to change these. For example, If the starting port is 9000, then Line 1 is using 9000 and 9001, and Line 2 is using 9002 and 9003, and so forth.
SIP Operation Manual V2.6 PPTP※ Select “PPTP” and enter the IP Address, Subnet mask, PPTP Server, PPTP ID and Password. Then click the “Accept” button at the bottom. Save the settings, and then restart the system. The system will take about 40 seconds to restart. If not familiar with the network connection, please contact your local ISP.Domain Name Server BigPond (for Australia use only) Click “BigPond Cable”...
SIP Operation Manual V2.6 Clone MAC Some Internet Service Providers (ISP) assigns the bandwidth via the MAC (Media Access Control) Address. You can click the " Clone" button to copy the MAC address of the Ethernet Card installed in the computer used to configure the device.
SIP Operation Manual V2.6 LAN interface mode※ Router: The system serves as a router. Bridge: The system serves as a bridge between WAN port and LAN port. Network Settings (LAN) Network Settings (LAN): Gateway LAN Port IP address and the subnet mask value. Please note that Gateway is built under NAT: Gateway LAN Port IP address cannot be in the same section as the NAT LAN Port IP address, or else it is unable to make or receive calls.
SIP Operation Manual V2.6 Enable DHCP Server: Enable or Disable DHCP server service of gateway. IP Pool Starting Address: The first IP address to be assigned to DHCP clients. IP Pool ending Address: The last IP address to be assigned to DHCP clients. Lease Time: The valid period of an assigned IP address.
SIP Operation Manual V2.6 QoS Settings WAN QoS QoS (Quality of Service): Sets an external bandwidth to ensure sound quality during transmission (When this function is enabled, the voice packet has the highest priority to ensure telecommunication quality while less bandwidth is assigned for data transmission). Some models of the VoIP gateway without this function can adjust the bandwidth automatically.
SIP Operation Manual V2.6 NAT/DDNS NAT Traversal If a Gateway is set up under an IP sharing setting, you can select either the NAT or STUN protocol. NAT Public IP: The IP address used by the gateway should be a virtual address. Further more, users must set the Virtual Server Mapping in the NAT Server (A virtual server is defined as a Service Port, and all requests to this port will be redirected to this specified the server IP address).
SIP Operation Manual V2.6 Telephony Settings Prefix Number Rules Trunk Dial Out Verify/ Trunk Dial Out Replace: The system will transfer the number for all transit out call through FXO port. For example: If you transit out with 01907123456, the system will trans to 190601 907123456.
SIP Operation Manual V2.6 Trunk Incoming Prompt Voice: Select the greeting (must use the IVR 132 function to record a voice file) when FXO receives an inbound call. Enable: To enable a line; if some lines are not used, disable them (Pause Function) to avoid unnecessary waiting when an incoming call is diverting to this line.
SIP Operation Manual V2.6 Grounding Compensation: If the FXO can’t work correctly, you will hear noises or echoing in the background as you are talking. Enable FAX: Enable this line to detect if there is a FAX tone to transfer the codec. Enable FXO/Trunk Extension Number: Select this function only when FXO receives 2 or more different PBX or PSTN, or under special circumstances.
SIP Operation Manual V2.6 Private Network Users can establish a private network by Phone Book Manager Service. Phone Book Manager Service is different from Proxy. Gateway is able to register to Phone Book Manager Service and SIP Proxy at the same time.
SIP Operation Manual V2.6 The bandwidth of different Codec needing: G.711 G.729a G.723.1 20ms 171.2 59.2 30ms 156.8 44.8 41.6 60ms 142.4 30.4 27.2 Client Settings Register to Phone Book Manager: To register to the Phone Book Manager. If the Gateway is Phone Manager Server, it has to enable this function to communicate with other clients.
SIP Operation Manual V2.6 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 talking gateways. This option tells the gateway to send and receive all SIP packets to the destined outbound proxy server rather than the remote gateway.
SIP Operation Manual V2.6 Proxy Server IP/Domain: Enter the Proxy Server IP address or URL (Uniform Resource Locator). You can set 3 redundant Proxy spread by semicolon. EX: 18.104.22.168;22.214.171.124;proxy.sip.sip Proxy Server Port: Enter the Proxy Server listen port number. (The factory default value is 5060) Proxy Server Realm: Enter the correct registered Proxy Server Realm name to avoid registration failure.
SIP Operation Manual V2.6 Calling Features Unconditional Forward: All incoming calls will be forwarded to the “Forwarding Number” automatically. Busy Forward: Forward the incoming call to “Forwarding Number” when the port is busy. No Answer Forward: Forward the incoming call to “Forwarding Number” after ring timeout expires without answer.
SIP Operation Manual V2.6 Advanced Options Web UI Login ID and Password: Enter login ID and password when you log onto the Web. Web UI auto log out: If a user does not act within the effective time range when logging into a web page, the user will be disconnected from the web page to allow others to login.
SIP Operation Manual V2.6 Line Settings Listening Volume: Adjusts the hearing volume. Speaking Volume: Adjusts the speaking volume. Tone Volume: Adds a new option to make tone volume adjustable. This setting will be applied to all tones generated by the VoIP gateway including Dial Tone, Busy Tone, and so on. Flash Time: FXS: Used to adjust the detecting period of flash signal from the phone set connected to the FXS port.
SIP Operation Manual V2.6 Fax Settings T.38: The T.38 protocol is used for better and faster facsimile transmission. When this function is enabled, the following fax and voice parameter settings will be disabled, so it is recommended to enable this function to gain better fax quality. When this function is enabled, please select UDP, TCP, or AUTO.
SIP Operation Manual V2.6 Local Phone Book This system can set up and store 100 phone numbers into a phone book and provides an IP address query when calling to other Gateway(s). If no Phone books manager is set within a Gateway group, then all Gateway systems have to set up phone data for each VoIP gateway to communicate with each other.
SIP Operation Manual V2.6 CDR Settings The user can set up a CDR Server to record call details for every phone call. The present CDR provides the call detail recording in a text file and if needed. it can be imported to prepare for an analysis report.
SIP Operation Manual V2.6 Transit Call Control Inbound Call Control: To determine when users make a phone call from a PSTN to Gateway FXO whether or not they check the inbound PIN code while using a VoIP －only effective for incoming calls calling from a PSTN trunk.
SIP Operation Manual V2.6 When Gateway A enables Inbound Call Control and Gateway B disables Outbound Call Control, then: If using an extension line at Gateway A to make a local call to a remote end through Gateway B, then dial Gateway B Number + Local Phone Number For example: 5168 35020311# If using a trunk line to dial to Gateway A, and then making a local call to a remote end through Gateway...
SIP Operation Manual V2.6 If Level 1 is set to prohibit dialing any number with prefix 0, then any level below 1 (including Levels 2 to 5) is also prohibited. Since Level 0 is not restricted to any prefix, therefore at level 0 users can dial a number with the prefix 0.
SIP Operation Manual V2.6 Busy Tone Cadence Measurement: Provide a best solution of FXO integrated with PSTN or PBX. FXO will learn the busy tone automatically. BTC Detection Sensitivity: The more sensitivity, the more quickly the system will cut off the call. If the system often cut off an un-finished call, select less sensitivity.
SIP Operation Manual V2.6 Direct Connection to PSTN 36008913 36008914 Connect one of the trunk lines to the Gateway FXO Port (For MODEL 2S2O, please connect to P3). The line of “Dial Number” (36008913) must be hook on. Detect Channel: Enter 3 (The trunk line is connected to P3, and uses P3 for outgoing detection). Phone Number: Enter the number of the FXO line.
SIP Operation Manual V2.6 36008913). After it rings pick up the phone and enter”#”, then hang up. The system will then detect a busy tone automatically. After detecting it will be as below: Connected to a PBX Extension Line If Gateway is connected to a PBX extension line, then the busy tone of both the PBX and the PSTN must be detected.
SIP Operation Manual V2.6 Connect one of the PBX extension lines to Gateway FXO Port (For MODEL 2S2O, connect the line to P3). Detect Channel: Enter 3 (The trunk line is connected to P3, and uses P3 for outgoing detection). Phone Number: Set the number of FXO line –to detect Reorder Tone.
SIP Operation Manual V2.6 System Information RTP Packet Summary Displays the information of the last finished call. It contains peer IP, peer port, packets sent, packet received and packet lost. STUN Inquiry...
SIP Operation Manual V2.6 Ping Test Use “ping” to identify if the remote peer is reachable. Fill in remote IP address and click “Test” will start the test. Time Settings 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.
SIP Operation Manual V2.6 Software Upgrade Gateway provides software upgrade function for a remote end. Software Upgrade Server IP address: Please enter the software server IP address. Software Upgrade Server Port: The default setting is 6001(Do not change this setting). Logout Gateway only allows one user to login at a time, so whenever a change is made, please save the settings, restart the system, or logout to avoid the situation where other users cannot login to change settings.
SIP Operation Manual V2.6 5. IP Sharing Functions All Gateway series have a built-in IP sharing function. The settings and instructions at a PC end are described below: Current Intranet only supports static IP mode, and the settings at the PC end are as follow: Available IP address Range : 192.168.8.1 –...
SIP Operation Manual V2.6 The IP settings on PC are as follows (using Windows 2000 for example) Open Start->Settings->Control Panel Open Network and Dial-up Connection Open Local Area Connection Click Properties...
SIP Operation Manual V2.6 Select TCP/IP, and then click Properties. Select “Use the following IP Address ” and enter IP address, Subnet Mask, and Default Gateway. Please note that an IP address in the same domain cannot be reused. Then, enter the DNS server IP address (varies in different networks.
SIP Operation Manual V2.6 6. Coding Principle Instruction After a phone number is entered, dial # to call out immediately or, wait until the “Inter DTMF Timeout” expires (defined in “Advanced Options”, default=4 seconds). If the phone number fits the setting of Digit Map, the gateway dials out the phone number through the assigned interface automatically.
SIP Operation Manual V2.6 Start Enter a phone number (D#) Is (D#) Dial the number defined in Speed defined in Dial table? SpeedDial table Is (D#) defined in Extension table? Is (D#) defined in Pho ebook table? Is (D#) defined in Phon book Manager? Is (D#) defined in SIP proxy...
SIP Operation Manual V2.6 7. Advanced Feature Static Route※ To build static routes within an internal network. These routes will not apply to the Internet. Route: Enter the IP of the specified network. Route Mask: Enter the subnet mask to be used for the specified network. Next Hop IP: Enter the IP address to the specified network.
SIP Operation Manual V2.6 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.
SIP Operation Manual V2.6 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 limit accessing Internet PC. 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...
SIP Operation Manual V2.6 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 to trigger the special application.
SIP Operation Manual V2.6 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 packet type is more than the setting in one second, then it will be attacked.
SIP Operation Manual V2.6 Notice ※ These functions are only supported for the second hardware. The second hardware can support more function, such as PPTP, LAN Interface Mode, LAN QoS, etc. There is an easiest way to make out the first and second hardware, please check Network Settings of User Interface.