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Summary of Contents for XPEECH 2S
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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.
S I P Operation Manual 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.
S I P Operation Manual Hardware Description 2 ports gateway model: 2S / 2O / 1S1O Front Panel WAN, LAN indicators Voice ports indicators Status indicators VoIP Gateway L3 L2 L1 WAN Alarm RUN Power P2 P1 Power Indicator: Green light indicates a normal power supply.
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S I P Operation Manual Model Description 2S: P1-P2 stand for Phone1-Phone2. Connect to your analog telephone. 2O: P1-P2 stand for Line1-Line2. Connect to your original telephone line on the wall jack with RJ-11 cable. 1S1O : P1 stand for Phone1 and P2 stand for Line1. Phone port is connected to your analog telephone, and Line port is connected to your original telephone line on the wall jack with RJ-11 cable.
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S I P Operation Manual 2O Model (2 FXO ports) RESET LAN ports 1 ~ 4 FXO ports 1 ~ 2 To reset the (Built-in Ethernet switch) (PSTN line connectors) gateway Connect LAN hosts here Connect to PSTN lines Or to restore to share WAN connection.
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S I P Operation Manual 4 ports gateway 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.
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S I P Operation Manual Model Description 4S: P1-P4 stand for Phone1-Phone4. Connect to your analog telephone. 4O: P1-P4 stand for Line1-Line4. Connect to your original telephone line on the wall jack with RJ-11 cable. 2S2O: P1-P2 stand for Phone1-Phone2 and P3-P4 stand for Line1-Line2. Phone ports are connected to your analog telephone, and Line ports are connected to your original telephone line on the wall jack with RJ-11 cable.
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S I P Operation Manual 4O Model (4 FXO ports) FXO ports 1 ~ 4 LAN ports 1 ~ 4 RESET To reset the (PSTN line connectors) (built-in Ethernet switch) gateway Connects to PSTN lines Connect LAN hosts here or to restore to share WAN connection.
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S I P Operation Manual 3S1O Model (3 FXS and 1 FXO ports) RESET FXS ports : 1, 2, 3 LAN ports 1 ~ 4 To reset the (built-in Ethernet switch) FXO ports : 4 gateway Connect LAN hosts here FXS to telephone set;...
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S I P Operation Manual 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.
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S I P Operation Manual Model Description 8S: P1-P8 stand for Phone1-Phone8. Connect to your analog telephone. 8O: P1-P8 stand for Line1-Line8. Connect to your original telephone line on the wall jack with RJ-11 cable. 6S2O: P1-P6 stand for Phone1-Phone6 and P7-P8 stand for Line1-Line2. Phone ports are connected to your analog telephone, and Line ports are connected to your original telephone line on the wall jack with RJ-11 cable.
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S I P Operation Manual 8O Model (8 FXO ports) LAN ports 1 ~ 4 RESET FXO ports 1 ~ 8 (built-in Ethernet switch) To reset the Connect LAN hosts here (PSTN line connectors) gateway to share WAN connection. Connects to PSTN lines or to restore IP sharing features enabled factory settings...
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S I P Operation Manual 4S4O Model (4 FXS and 4 FXO ports) LAN ports 1 ~ 4 RESET FXO ports 5~8 FXS ports 1 ~ 4 (built-in Ethernet switch) To reset the Connect LAN hosts here gateway (telephone connectors) (PSTN line connectors) to share WAN connection.
S I P Operation Manual 2. Installation and Applications Network Interface The network interface is divided into 4 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.
S I P Operation Manual 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. Please avoid IP address 192.168.0.1-192.168.8.254 (You may need LAN IP address of IP sharing to change the settings of IP sharing or change SIP series Gateway LAN Port IP address) Set as static IP address, and assign the LAN IP address of the IP...
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S I P Operation Manual 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.
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S I P Operation Manual Gateway assigned with a Public IP Address and serving as a bridge 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 Bridge function to allow a PC to be on-line at the same time.
S I P Operation Manual Telephone Interface Description 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. Integrating the 4S with PBX P1-P4 is FXS interfaces, and some of them can be connected to telephone sets for direct calls.
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S I P Operation Manual 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.
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S I P Operation Manual 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.
S I P Operation Manual 3. Setting the Gateway through IVR VoIP transmits voice data (packets) via the Internet. One effect of this is that telecommunications quality is closely related to the condition and status of the network environment. If any of the parties involved in VoIP communications has insufficient bandwidth or frequent packet loss, the telecommunication quality will be poor.
S I P Operation Manual Instructions FXS Port: Connect FXS port to a telephone. To access IVR mode, you should enter factory default code “** #” . If it requires password, enter ** password #”. Character to number conversion information is provide in the PPPoE Character Conversion Table. After entering the correct IVR password, you will hear an indication tone after which the gateway is in IVR setup mode.
S I P Operation Manual IVR Functions Table: Function Description Example / Notes Code 111/101 WAN Port IP address Set/Query Use function code 114 to select 1 for Static IP connection then 112/102 WAN Port Subnet Mask Set/Query setup the IP address 113/103 WAN Port Default Gateway Set/Query Current Network IP Access Set/Query (1: Static IP,...
S I P Operation Manual IP Configuration Settings of WAN Port Static IP Settings NOTE: Complete static 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.
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S I P Operation Manual ADSL PPPoE Settings NOTE: Before setting PPPoE, you must have PPPoE account (121) and PPPoE password (122) provided by your local Internet Service Provider (ISP). Select a PPPoE After entering IVR mode, dial 114. You will hear “Enter value”. Dial 3 to select PPPoE.
S I P Operation Manual PPPoE Character Conversion Table The table below provides a list of PPPoE conversion codes. The first column in each pair of columns lists the number, letter or symbol that you want to enter. The second column in each pair (“Input Key”) tells you what code to enter for the corresponding number, letter or symbol.
S I P Operation Manual 4. Setting a Gateway with WEB Browser The VoIP gateway allows users to configure its settings using a web interface (Web UI). You can access the Configuration Menu by opening a web-browser (e.g., Internet Explorer or Netscape Navigator) and entering the factory default LAN IP address: 192.168.8.254.
S I P Operation Manual Network Settings (WAN) The network settings are used to set the gateway’s communication ports, and IP configurations, 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.
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S I P Operation Manual RTP Starting Port UDP: The initial value of port number of transmitting voice data among gateway(s). Each line requires 2 ports (RTP/RTCP). It is not necessary to change these, unless it conflicts with ports used by another network device. For example: If the starting port is 9000, then Line 1 is using 9000(RTP) and 9001(RTCP), and Line 2 is using 9002 and 9003, and so forth.
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S I P Operation Manual ADSL PPPoE Settings Select and enter and Re-enter Password to confirm. Then click the PPPoE Account Password button at the bottom. Save the settings, and then restart the system. Accept PPTP Select PPTP and enter IP Address, Subnet mask, PPTP Server, PPTP ID and Password. Then click the Accept button at the bottom.
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S I P Operation Manual Domain Name Server Domain Name Server (DNS): While a gateway is accessing another gateway or a computer with hostname, it will look up the IP address from the DNS provided by your ISP. Normally, the ISP assigns DNS information while negotiating with PPPoE or DHCP.
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S I P Operation Manual Using Phone Book Manager NOTE: This function is only available in Standard version. Please see Private Network if your gateway is Dual Network version. Server Settings Enable Phone Book Manager Server: This allows other gateway users to register the IP address and telephone number in this Phone Book manager.
S I P Operation Manual Network Settings (LAN) LAN interface mode Router: The system serves as a router with NAT. Bridge: The system serves as a bridge between WAN port and LAN port without NAT. (LAN default gateway will still be accessible for configuration). Network Settings (LAN) The gateway LAN Port IP and Subnet mask settings.
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S I P Operation Manual DHCP Settings Enable DHCP Server: Enable or Disable DHCP server service of the 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. IP Pool Uses Other Default Gw: Tick the check box to give DHCP client the other default gateway.
S I P Operation Manual QoS Settings QoS is that according to the actual bandwidth offered by Internet service, set the appropriate value in WAN QoS filed. Reserve bandwidth is recommended to enable the gateway for other transmission application. WAN QoS QoS (Quality of Service): To set true bandwidth of your Internet connection to ensure sound quality during transmission.
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S I P Operation Manual LAN QoS Users can allocate bandwidth for various types of network traffic with each LAN port and apply the LAN QoS control function on incoming and outgoing traffic flows, or both of them. The real throughput of incoming rate is depended on downstream bandwidth of WAN QoS and that of outgoing is depended on upstream bandwidth.
S I P Operation Manual NAT/DDNS NAT Traversal If the 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.
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S I P Operation Manual DDNS These settings are only necessary when the gateway is set up behind an Internet sharing device that uses a dynamic IP address and do not support DDNS. The current system allows users to choose either DynDNS 、 TZO 、 3322.org 、 PeanutHull or a private server.
S I P Operation Manual Caller ID FXS Caller ID Generation: Select this option to enable the caller ID display function on FXS ports. When enabled, the caller’s phone number will be displayed on your phone set when the call comes through. FSK is preferred in some countries. FXO Caller ID Detection: It is to detect Caller ID delivered from PSTN to the FXO port.
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S I P Operation Manual Transit In Caller ID Strip / Replace You can change the information of the calling party while making calls to Internet. Scan code: Defines the rule of the Caller IDs detected by FXO. It can be a prefix or a full number.
S I P Operation Manual Telephony Settings Prefix Number Rules Trunk Dial Out Verify/ Trunk Dial Out Replace: Before FXO dials to PSTN, the gateway will check (verify) the numbers in field and change (replace) them Trunk Dial Out Verify with the numbers in filed.
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Dial-out Prefix: It is the number dialed automatically by the system when the FXO interface diverts a call to the PSTN by VoIP (2S/4S/8S do not support this function). FXO Line Default Dial-Out: Default number that FXO will dial out when it receive an...
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S I P Operation Manual Example: If PBX extension needs to dial “0” to make a PSTN call, and the FXO ports are connected to PBX extension. In this case, the Dial-out prefix should be set to “0”. If the PBX requires some delay time before capturing a line, then the trunk prefix should be set as “0,”...
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S I P Operation Manual Trunk Hunting Order: To set FXO dial-out mode when there is an incoming call dialed FXO representative or unassigned call. First Idle: The gateway will assign each unassigned call from first FXO port. Sequential: The gateway will automatically assign the first unassigned call to the first FXO port.
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S I P Operation Manual Pick up Line by Dialing Extension Number: Allows user to dial the extension number first, after hearing the second dial tone, and dial a PSTN number. If you are registered to a Proxy, it MUST be checked. Wait for Caller ID before FXO / Trunk pick up: To detect caller ID from FXO port.
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S I P Operation Manual FXS Group Hunting/Ring Priority Hunting/Ring: This variable is able to set FXS group hunting using simultaneous ring or sequential ring. Sequential Ring Time: Set the ring time of each port, when sequential ring is chosen. Hunting Priority: This variable can be adjusted using the Up and Down arrows.
S I P Operation Manual 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.
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S I P Operation Manual Each line registers to Proxy independently: As there are various Proxy Server providers, according to RFC standard our company has designed the gateway to be compatible with them. If any registration problem occurs, please consult your Proxy Server Provider. NOTE: When you register with a Proxy, dialing principles may vary with different Proxy Servers, especially when dialing through a remote end FXO port.
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S I P Operation Manual SIP Proxy Server / Soft Switch Settings Enable Support of SIP Proxy Server / Soft Switch: Tick the check box to enable the functions to inter-work with Proxy Server / Soft Switch. When SIP Proxy 1 and 2 are enabled, the gateway will register to SIP Proxy 2 after all lines have failed to register to SIP Proxy 1.
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S I P Operation Manual VoIP failure announcement: As soon as VoIP call or the registration to proxy server is failed, the gateway will drive IVR system to play out failure announcements for the caller. Bind Proxy Interval for NAT: This function is able to keep the binding that exists when the gateway is behind a NAT and Proxy Server is not able to keep the binding.
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S I P Operation Manual P-Asserted (RFC 3325) Settings Enable P-Asserted: It is for caller ID protection. Privacy Type: Privacy type is used to disguise the caller ID when queried via an ITSP/Third-Party Assertion. Other Settings SIP Message Resend Timer Base: This parameter let users define the base of timeout to fit the real network.
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S I P Operation Manual Support URI Percent-Encoding (RFC 3986): It follows RFC 3986 to encode/decode the letters of the basic Latin alphabet, digits, and a few special characters. Compare SIP 'To' Header for Transit Out: When FXO is called and the number of Request line and “To”...
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S I P Operation Manual Example of To Invite Proxy: International Call Prefix Digit: 00 Country Code: 1 Long Distance Call Prefix Digit: 0 Area Code: 567 ENUM Head Exception: 070 Phone Number Dialed By The True Phone Number Description The User Dialed By VoIP gateway Exclude International Call Prefix Digit and...
S I P Operation Manual Private Network This section provides a SIP implement of traditional telephony services. NOTE: This function is only available in Dual Network version. Phone Book Manager Service Server Settings Enable Phone Book Manager Server: This allows other gateway users to register the IP address and telephone number in this Phone Book manager.
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S I P Operation Manual Clients Settings Register to Phone Book Manager: Register to the Phone Book Manager Gateway Name for Phone Book Manager: The alias registered with the Phone Book Manager. Phone Book Manager Login Password: Enter the registered password. If this system is serving as the Phone Book Manager, the set password is also the password used for registering other gateway systems Phone Book Manager IP/Domain: Enter the IP address of the Phone Book Manager.
S I P Operation Manual Calling Features This section provides a SIP implement of traditional telephony services. Do Not Disturb: Tick the check box to reject all incoming calls from VoIP. Allow only to call out. Unconditional Forward: All incoming calls will be forwarded to the Forwarding Number automatically.
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S I P Operation Manual Calling Feature Instructions: Call Hold: The call will be put on hold after the FLASH button is pressed on the phone set. The gateway will play hold music (provided by your VoIP network) to the remote end. Call Transfer: Call will be put on hold after FLASH button pressed on local phone set (gateway plays on-hold music to the remote end).
S I P Operation Manual Advanced Options NOTE: There are two operating levels when entering the Web UI. Logging-in as the Administrator allows you to change all settings. A Web UI user only has access to some settings. Web UI auto log out: If a user does not act within the effective time range when logging into the web interface, the user will be disconnected from the web page to allow others to log-in.
S I P Operation Manual Enable Out-of-Band DTMF: Send DTMF keys (0~9, *, #), follow the RFC2833 rules or via SIP Info. Enable Hook Flash Event: According to RFC2833 or SIP info the gateway will deliver Hook Flash signal to the remote party. Payload Type:Payload type of RFC2833.
S I P Operation Manual PSTN Ring OFF Length: It is used to detect if the PSTN remoter party is on-hook through the ring length from PSTN by FXO port. If the ring length form PSTN is larger than this setting, it is going on-hook by FXO port, and it makes FXO not answering the call.
S I P Operation Manual Fax Settings NOTE: When a fax tone is detected in a call, the gateway will automatically switch from voice mode to fax mode. So fax settings will be temporarily applied to a specific port which detects fax tones, instead of its default voice settings.
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S I P Operation Manual Voice Menu Options Voice Menu Options: Tick the check box to enable or disable IVR function. NOTE: When disabled, call pickup, Automatic Redial and unattend transfer will be disabled.
S I P Operation Manual Digit Map Digit Map now is combined the original feature of Digit Map and Speed Dial. You can use “?” or “%” in the column of Scan Code, VoIP Dial-out and PSTN Dial-out. “?” is a single digit, and “%” is wildcard.
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S I P Operation Manual Enable: Tick the check box to enable detection of this entry. Scan Code: Defines the digits for the gateway to scan while user is dialing. VoIP Dial-out: Defines the dialed number rule for the gateway to call through Internet. PSTN Dial-out: Defines the dialed number rule for the gateway to call through PSTN.
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S I P Operation Manual Method 2- Multi mapping; Fill the prefix code into the column and the format to Scan Code transfer into the column. VoIP Dial-out or PSTN Dial-out Example 1 - Multi mapping, Scan Code: 2??? PSTN Dial-out: 351006??? User Dial Length: 4 Route: PSTN Pick up the handset and dial 2301.
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S I P Operation Manual FXO. You also can use Digit Map Testing to know that the gateway will dial 0423456789 to Internet and 1805423456789 to FXO. Method 3- Substitution; It helps you dial to destination that you can not dial by phone. Destination like: test@1.1.1.1.
S I P Operation Manual Phone Book The gateway 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 Book manager is set within a gateway group, then all gateways have to set up phone data for each gateway to communicate with each others.
S I P Operation Manual CDR Settings The user can set up a CDR Server to record call details for every phone call with TCP protocol. The present CDR provides the call event such as HOOK ON, HOOK OFF, DIALED NUMBER, DATE…recording in a text file and which can be imported to prepare an analysis report.
S I P Operation Manual Transit Call Control This is to control outgoing call and incoming call through FXO. Transit Call Control is effective when it cooperates with Long-Distance Control Table. Long-Distance Exception Table is for an exception and it will not be restricted by Transit Call Control and Long-Distance Control Table. You have to enable both of Inbound/Outbound Call Control PIN Code...
S I P Operation Manual 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 network, as below Descriptions: Digit strings in this table are prefixes that the gateway will check on dialed numbers in transit out calls.
S I P Operation Manual CPT/Cadence Settings CPT/Cadence 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: Provide a solution of FXO integrated with PSTN or PBX.
S I P Operation Manual UDT Detection If the CPT auto detect function is not able to determine whether or not a PSTN-call receiving party has hung up the phone, then the UDT detection function can serve as a back up. To do this, enter the high/low frequency parameters from the CPT table into the UDP table.
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S I P Operation Manual Direct Connection to PSTN 36008913 36008914 Connect one of the trunk lines to the FXO Port (For MODEL 2S2O, please connect to P3). The line of (36008913) must be hook on. Dial Number Detect Channel: Enter 3 (The trunk line is connected to P3, and uses P3 for outgoing detection).
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S I P Operation Manual Detection in Progress: during detection, the following windows will appear. Once detection of a busy tone is in progress, the gateway will dial the number to be tested (in this case 36008913). After it rings pick up the phone and enter “#”, then hang up. The gateway will then detect a busy tone automatically.
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S I P Operation Manual 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. 36008913 Connect one of the PBX extension lines to the FXO Port (For MODEL 2S2O, connect the line to P3).
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S I P Operation Manual Filling in the CPT Table Fill in the table after the detection is completed as below, where the values are the frequency and On-and-Off ratio detected. Please click the button. If connecting gateway to a PBX Accept 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.
S I P Operation Manual System Information This page shows that the status of the 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 gateway is start, etc.
S I P Operation Manual RTP Packet Summary Display the information of the last completed call. This report contains peer IP, peer port, packet sent, packet received and packet lost. Press button to get the latest RTP Packet Summary. Refresh STUN Inquiry Use STUN Inquiry to detect your IP sharing device’s NAT type and communication between a STUN server and client.
S I P Operation Manual This section works with Time server to provide NTP time synchronization. The gateway will get its local time from specific time servers after it connects to Internet. Time Zone: Set the Time Zone where the gateway resides. Time Server #1~#3: Set the Time Server where the gateway should sync up during start up.
S I P Operation Manual Provision Settings This section sets parameters required by Auto Provisioning System. Typically, Provision Server is used to provision, configure, manage and maintain subscribers and network users. This gateway, acts as a part of subscribers, can be controlled by Provision Server. The gateway provides a simply way for users to connect and send request to Provision Server by enabling this setting.
S I P Operation Manual Provision Retry Times/Retry Interval: Adjust Retry times or interval. Suspend Service: When it is clicked, indicating the server has stopped providing provision and VoIP call service. Each FXO/FXS port is not available to make any call. NOTE: Contact your server provider while necessary.
S I P Operation Manual Software Upgrade The gateway provides a software upgrade function for a remote source. Please consult your service provider for information about the following details. Upgrade Server: Choose the server type given by your service provider. Server IP Address: Enter the software server IP address.
S I P Operation Manual 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 –...
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S I P Operation Manual 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...
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S I P Operation Manual Click Properties Select TCP/IP, and then click Properties.
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S I P Operation Manual 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. consult your ISP’s service for information). Click the “OK” button and after completing the settings, users can use both the VoIP and network services concurrently.
S I P Operation Manual 6. Coding Principle Instruction Dial the phone number which you want to call and press # to call out immediately, or wait until the “Inter DTMF Timeout” expires (defined in Advanced Options\Inter Digits Timeout, default=4 seconds). If the phone number fits the setting of the Digit Map, the gateway dials out the phone number through the assigned interface automatically.
S I P Operation Manual Example for Match phone numbers invited by callers The table below is provided as a general reference expresses phone numbers dialed by the gateway instead of real phone numbers that callers dial. Match Scheme Description Ring FXS according to “FXS The same as “FXS Group Hunting / Ring...
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S I P Operation Manual Start Enter a phone number (D#) Is (D#) Dial the number defined in Digit defined in Digit Map table? Map table Is (D#) defined in Extension table? Is (D#) defined in Phone Book table? Is (D#) defined in Phone Book Manager? Is (D#) defined in SIP server?
S I P Operation Manual 7. Advanced Feature Static Route 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: The next hop IP address to the specified network.
S I P Operation Manual Port filtering Port filtering enables you to control all data that can be transmitted over routers. When the port used at the source end is within the defined scope, it will be filtered without transmission. Enable Port Filtering: Tick the check box to enable this function.
S I P Operation Manual MAC Filtering MAC (Media Access Control) address filtering allows you to filter the transmission of data by network card physical address. MAC: Enter a MAC address to prevent the particular device from accessing the Internet. Remark: Remark field for this entry.
S I P Operation Manual URL Filter URL filter is used to deny device on the LAN from accessing specific web sites. The gateway will block any URL that contains the strings listed. Special Applications Provide multiple connections for special applications. Name: The name of the special application.
S I P Operation Manual DoS Prevention Settings Enable DoS Prevention: To prevent DoS attacks from WAN. Enable DoS Prevention on LAN: To prevent DoS attacks from LAN. Enable Source IP Blocking: Block a particular IP. Blocking Time: The time to block the IP.
S I P Operation Manual 8. VPN IPSEC ※ Virtual Privet Network lets two networks communicate securely when the only connection between them is over a third network which they do not trust (ex. Internet). For instance, two VPN enabled devices can establish a secure IP tunnel over Internet that allows traffic from both local network to communicate over this virtual dedicate line.
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S I P Operation Manual IPSEC configurable parameters VPN IPSEC -> Configuration Enable IPSEC: Enable or disable this VPN tunnel. Local Security Group: The IP segment of local LAN port. Local Security Group Mask: The subnet mask of local LAN port. Remote Security Group: The IP segment of remote VPN device LAN port.
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S I P Operation Manual Remote Security Gateway: The IP address or FQDN of remote VPN device. Encryption: Tunnel data encryption. Authentication : Authentication header and checksum. Local ID (optional): ID field to be identified by remote VPN device (ie. “@local-domain.com”).
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S I P Operation Manual In both gateways you will need to enable IPSEC first, fill in gateway1 WAN IP address, gateway1 LAN segment, gateway2 LAN segment, gateway2 WAN IP address, etc. This is a sample of gateway1: fields, put in your local LAN IP segment Local Security Group Local Security Group Mask information.
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S I P Operation Manual This is a sample of gateway2: Save and restart both gateways then these two gateways should be connected with IPSEC VPN.
S I P Operation Manual Notice ※ These functions are only supported for the special hardware. The special hardware supports VPN IPSEC function. General "IMPORTANT SAFETY INSTRUCTIONS - When using your telephone equipment, basic safety precautions should always be followed to reduce the risk of fire, electric shock and injury to persons, including the following: -Do not use this product near water for example, near a bathtub, washbowl, and kitchen sink or laundry tub, in a wet basement or near a swimming pool.
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