2~24-Port H.323/SIP VoIP Gateway VIP-281/480/880/1680/2480 series User’s manual Version 2.1.
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TABLE OF CONTENTS Chapter 1 Introduction.......................................................................... 6 Overview............................................................................................................................6 Package Content ...............................................................................................................8 Physical Details .................................................................................................................
Management....................................................................................................................53 Save Configuration ...................................................................................................54 Access Control..........................................................................................................54 Set To Default Configuration....................................................................................
Chapter 1 Introduction 1 Overview With years of Internet telephony and router manufacturing experience, PLANET proudly introduces the newest member of the PLANET VoIP gateway family: the VIP-GW, VIP-281 / VIP-480 / VIP-880 / VIP-1680 / VIP-2480 series. The PLANET VoIP Gateway is fully both SIP and H.
¾ VIP-281FS equips two FXS interfaces telephone set or FAX machine connection (FXS). 4-Port Model ¾ VIP-480 equips two FXO and two FXS interfaces to have the great flexibility of PBX connection (FXO), and telephone set or FAX machine connection (FXS). ¾ VIP-480FS equips four FXS interfaces telephone set or FAX machine connection (FXS). ¾ VIP-480FO equips four FXO interfaces to have the great flexibility of PBX connection (FXO).
• Point-to-Point Protocol over Ethernet (PPPoE) Client Support: If you are a DSL user, the router has a built-in PPPoE client for establishing a DSL link connection with the ISP. There is no need to install a further PPPoE driver on your computers. • Smart QoS The smart QoS provide stable voice quality while user access internet from private LAN to internet at thesame time.
¾ RS-232 cable x 1 (8 / 16 / 24-port model) ¾ 25 port Telephone Cable x 1 (16 / 24-port model) ¾ Rack mount brackets x 2 (16 / 24-port model) Physical Details The following figure illustrates the front/rear panel of VIP-GW series: Front Panel of VIP-281/VIP-281FS Rear Panel of VIP-281/VIP-281FS Front Panel of VIP-480/VIP-480FS/VIP-480FO Rear Panel of VIP-480/VIP-480FS/VIP-480FO Front Panel of VIP-880/VIP-880FS/VIP-880FO 9
Rear Panel of VIP-880/VIP-880FS/VIP-880FO Front Panel of VIP-1680/VIP-1680FS/VIP-1680FO/VIP-1680FD Front Panel of VIP-2480/VIP-2480FS/VIP-2480FO/VIP-2480FD Rear Panel of VIP-1680/VIP-1680FO/VIP-2480/VIP-2480FO Front Panel LED Indicators & Rear Panels Front Panel LED PWR State Descriptions On GW is powered ON Off GW is powered Off CPU (VIP-880 / VIP-1680 / VIP-2480 series) Flashing The system is running WAN Port ON GW network connection established Flashing Data traffic on cable network Off W
NOTE: System initialization will turn some LEDs ON for a few seconds. ÍNote The Default LAN IP is http://192.168.0.1. Press RESET button on rear panel over 5 seconds will reset the VoIP Gateway to this default LAN/WAN IP address and Username/Password function. Rear Panel WAN LAN (VIP-880/VIP-1680/ VIP-2480 series) LAN 1 ~ LAN 4 Descriptions The WAN port supports auto negotiating Fast Ethernet 10/100Base-T networks. This port allows your voice gateway to be connected to an Internet Access device, e.g.
(CO line), and the FXS interface is designed for connecting to analog telephone sets or fax machines. If the telephone cable connects to VIP-16/2480 series, the FXS interfaces are odd ports i.e. 1, 3, 5, 7, 9, 11, 13, 15, 17, 19, 21, 23, and the FXO interfaces are even ports, i.e. 2, 4, 6, 8, 10, 12, 14, 16, 18, 20, 22, 24.
Chapter 2 Preparations & Installation 2 Physical Installation Requirement This chapter illustrates basic installation of VIP-GW series • Network cables. Use standard 10/100Base-TX network (UTP) cables with RJ45 connectors. • TCP/IP protocol must be installed on all PCs.
ÍNote Please locate your PC in the same network segment (192.168.0.x) of VIP-GW. If you’re not familiar with TCP/IP, please refer to related chapter on user’s manual CD or consult your network administrator for proper network configurations. LAN/WAN Interface quick configurations Nature of PLANET VIP-GW is an IP Sharing (NAT) device, it comes with two default IP addresses, and default LAN side IP address is “192.168.0.1”, default WAN side IP address is “172.16.0.1”.
Parameter Description LAN IP address of VIP-GW IP address Default: 192.168.0.1 LAN IP address of VIP-GW Subnet Mask L Hint Default: 255.255.255.0 It is suggested to keep the DHCP server related parameters in default state to keep machine in best performance. After confirming the modification you’ve done, Please click on the Apply button to macke the changes effective, and click “Save Configuration” to save configuration.
Chapter 3 Network Service Configurations 3 Configuring and monitoring your VoIP Gateway from web browser The VIP-GW integrates a web-based graphical user interface that can cover most configurations and machine status monitoring. Via standard, web browser, you can configure and check machine status from anywhere around the world. Overview on the web interface of VoIP Gateway With web graphical user interface, you may have: More comprehensive setting feels than traditional command line interface.
VIP-GW main page Wizard Setup for Quick Start Wizard Setup After finishing the authentication, the Main menu will display 3 parts of configuration, please click “Wizard Setup” to enter quick start: 1. WAN Port Type Setup (Setup First) For most users, Internet access is the primary application. The VIP-GW support the WAN interface for Internet access and remote access. The following sections will explain more details of WAN Port Internet access and broadband access setup.
Default Gateway check with your ISP provider ADSL Dial-Up User (PPPoE Enable) Some ISPs provide DSL-based service and use PPPoE to establish communication link with end-users. If you are connected to the Internet through a DSL line, check with your ISP to see if they use PPPoE. If they do, you need to select this item.
2. Configuring NAT or Bridge setting: Bridge Mode: When working on Bride Mode, the VoIP gateway will use only the LAN setting IP, The VIP-GW will use the same LAN IP setting as WAN IP. That means, when Bride mode enable, the WAN connection setting will be ignored. NAT mode: LAN IP Network Configuration IP Address Subnet Mask Private IP address for connecting to a local private network (Default: 192.168.0.1) Netmask for the local private network (Default: 255.255.255.
3. VoIP Call Protocol Setup STEP 1 : Configure VoIP Call Signal Protocols : User could select either H.323 or SIP Protocol, and click “select” STEP 2 : Configure the numbering with phone/line ports Phone Number The representation number is the phone number of the telephone that is connected to phone port Line ports are connected to the extension ports of the PBX system or the PSTN line. They have a common Line Hunting Group Number.
STEP 4: Outgoing Dialing Plan The purpose of “Outgoing Direct Call” setting is to let user create a proprietary dialing plan when this Gateway is not registered to any H.323 Gatekeeper or any SIP proxy server. This setting can also assign some dialing plan to local ports (including prefix strip, prefix addition). Through this setting, user can directly map a number to a specific gateway (IP address).
Chapter 4 System Configurations 4 Advance Setup of Network Setup In Advanced Setup, VIP-GW provides user two major parts function to configure: One is “Network Setup”, the other one is “VoIP Call Setup” Network Setup Label WAN Setting LAN Setting Virtual Server Dynamic DNS Network Parameters Sets/changes the WAN port type like “Fixed IP”, “DHCP Client” or ”PPPoE”. Modifies the IP address of the LAN port and setting DHCP server parameters.
Static IP You are a leased line user with a fixed IP address; fill out the following items with the information provided by your ISP IP Address Kindly please check with your ISP provider Netmask Kindly please check with your ISP provider Default Gateway Kindly please check with your ISP provider PPPoE for ADSL Some ISPs provide DSL-based service and use PPPoE to establish communication link with end-users.
L Note WAN port display the IP address, Subnet Mask and default gateway IP address if DHCP client is successful LAN Setting There are two kinds of network feature to configure: Bridge Mode and NAT Mode: Bridge Mode Select this VIP-Gw as Bridge. (WAN Port and LAN Port use the same IP address) NAT Mode Each of the VIP-GW has two Ethernet interfaces, one is for connecting to local network users, and the other is for connecting to an external broadband device (i.e. DSL modem/router or Cable modem).
DHCP Server Configuration DHCP stands for Dynamic Host Configuration Protocol. It can automatically dispatch related IP settings to any local user configured as a DHCP client. The DHCP server supports up to 253 users (PCs) on Yes: Enables the DHCP server. No: Disables the DHCP server. Start IP Address Sets the start IP address of the IP address pool. End IP Address Sets the end of IP address in the IP address pool. DNS Server IP Address DNS stands for Domain Name System. Every Internet host.
Public Port Specifies which port should be redirected to the internal host. Private IP Specifies the private IP address of the internal host offering the service. Private Port Specifies the private port number of the service offered by the internal host. Apply Click here to add the port-mapping entry and enable the service. Dynamic DNS DDNS is a service that maps Internet domain names to IP addresses.
User Name Input your DDNS User Name Password Input your DDNS Password Domain Name Input you set from your DDNS DNS Server IP Input your DNS Server IP Netwrok Management Network Parameter allows you to modify the access port of gateway.
Advance Setup of VoIP Setup In Advanced Setup, VIP-GW provides user two major parts function to configure: One is “Network Setup”, the other one is “VoIP Call Setup” VoIP Setup Label The PLANET series gateway support 2~24 phone/line for SIP and VoIP Basic H.323 VoIP call applications. You can configure these ports from this menu. Dialing Plan Users could apply any dial policy by setting Dial Plan including outgoing dial plan and incoming dial plan.
Configure the numbering with FXS / FXO ports. (Depending on GW model number: if user uses the model number is VIP-1680, this VIP-1680 has 16 voice channels for setting, and the VIP-2480 had 24 voice channels for setting) FXS Number The representation number is the phone number of the telephone that is connected to FXS port. FXO ports are connected to the extension ports of the PBX system or the PSTN line. They have a common Line Hunting Group Number.
H.323 Parameters Label H.323 ID Sets the unique name of this Gateway, that is communicated as part of H.323 messaging. Primary Gatekeeper IP There are two gatekeeper address fields, one is primary, Address the other secondary. If this gateway does not want to register to any gatekeeper, just set value 0 to the primary gatekeeper address. If the primary gatekeeper address is Secondary Gatekeeper IP Address not 0, the gateway will register to the primary gatekeeper.
In H.323 standard the RAS default port number is 1719. The VoIP gateway provides user to change RAS port RAS Port Adjustment number to meet the network environment.(Some area carrier blocks or forbidden the default port number) In H.323 standard the default Q.931 port number is 1720. The VoIP gateway provides user to change Q.931 port to Q.931 Port Adjustment meet the network environment. (Some area carrier blocks or forbidden the default port number) H.323 Call Pass through NAT H.
“Add Digit Number” is the digits that will be added to the beginning of the dialed number. “Destination IP Address / Domain Name” is the IP address / Domain Name of the destination gateway that owns this phone number. Scenario description: Normally dial 001x leading call out, call to destination IP address: 172.16.0.100 002x leading call out, call to destination domain name: h323gw.test.
“Length of Number“ has two text fields need filled: “Min Length” and “Max Length” is the min/max allowed length you can dial. “Delete Length” is the number of digits that will be stripped from beginning of the dialed number. “Add Digit Number” is the digits that will be added to the beginning of the dialed number. “Destination telephone port” is “FXS/FXO port number” ; this is for local dial plan setting phone number.
Scenario description: Hunting for FXO port (VIP-480FO) Port 1: FXO was connected to PSTN. Port 2: FXO was connected to PSTN. Port 3: FXO was connected to PSTN. Port 4: FXO was connected to PSTN. H.323 number “123” call incoming, the port 1 will be off-hook and hear the dial tone from PSTN. If port 1 is busy, the port 2 will be will be off-hook and hear the dial tone from PSTN. If port 1 and port 2 are busy, the port 3 will be off-hook and hear the dial tone from PSTN.
Scenario description: Termination call to FXO for one-shoot call Port 1: FXO was connected to PSTN (area code is 81xxxxxxxx). H.323 leading number “081x” incoming, and delete the first one digit “0”, and call to PSTN number. Note: “081x” will be registered to H.323 Gatekeeper if “Register to GK” was enabled, show as below: Scenario description: Termination call to FXO Port 1: FXS Port 1: FXO was connected to PSTN (area code is 92xxxxxxxx).
H.323 VoIP Advance Configurtion Smart-QoS If this function is enabled, when VoIP call is occurred, the other data will be automatically reduced traffic which across the internet in order to guarantee the voice bandwidth. After the VoIP call is connected, when you dial a digit, this digit is sent to the other side by DTMF tone. There are two methods of sending the DTMF tone. The first is “in band”, that is, sending DTMF Relay for H.323 the DTMF tone in the voice packet.
The T.38 is a “Real Time Group 3 FAX communication over IP network” format. That’s meaning it’s a protocol for Fax over IP. You have to enable this function. This command configures the number of seconds that the gateway should be considered active by the H.323 Gatekeeper. H.323 RRQ TTL The gateway transmits this value in the RRQ message to the gatekeeper.The default value is “0”. H.323 Registration type There are 2 choices for this setting. “Gateway” means it will act as the VIP-GW.
Ring Frequency You can configure how long the Ring Frequency do you want to use. FXO Battery Reverse Enable battery reverse to detect polarity from PSTN line. The PSTN line can send H.323 case: Sending the Q.931 connect signal to caller when detecting polarity reverse from PSTN line. When user calls the PSTN line which was connected with the FXO port, there are three answer mode for user to configure. 1. Ringing Answer Mode (Default Setting): FXO answer the call once the ring coming from PSTN line. 2.
Note: This case can avoid the local PSTN charge when the FXS port still ring. 3. Non Answer Mode: FXO will NOT answer the call in any time. Note: Some ITSP only let the FXO for termination function, they do not user use the FXO port for origination Scenario description: H.323 call connecting answer mode Case B: “Hot Line Number” was assigned and the Hot line number belongs to remote H.323 device. Note: The remote H.323 device need Disable the “Auto Answer” 1.
H.323 Netwrok Advance Configuration Smart-QoS If this function is enabled, when VoIP call is occurred, the other data will be automatically reduced traffic which across the internet in order to guarantee the voice bandwidth. Bandwidth control You can configure your bandwidth what the Max byte of download G.723/G.729 and upload of ADSL modem rate. Bandwidth IP TOS Enable / Disable Type of Service in IP packets.
For example: Port 1 and port 2 is hunting for the port 1 SIP account. If the port 1 is incoming call, the other one SIP call from internet will ring port 2. SIP Proxy Server Setting Domain/Realm Enter the SIP realm in this field Enter the SIP service IP address or domain name in this field SIP Proxy Server (the domain name that comes after the @ symbol i n a full SIP URI).
Dialing Plan to SIP protocol The “Dialing plan” needs setting when the user uses the method of Peer-to-Peer or registering SIP proxy server mode. The SIP dialing plan has two kinds of directions: Outgoing (call out) and incoming (call in).
Scenario description: Normally dial 2290x leading call out, call to destination domain name: sipgw.test.com 221 leading call out, call to destination IP address: 172.16.0.100 Scenario description: Speed dial If user dials “101”, the gateway automatically dials “1234567890” to destination IP address: 172.16.0.101 If user dials “202”, the gateway automatically dials “0987654321” to destination IP address: 172.16.0.
Scenario description: Hunting for FXS port (VIP-400FS) Port 1: FXS Port 2: FXS Port 3: FXS Port 4: FXS H.323 number “123”call incoming, the port 1 will be ringing. If port 1 is busy, the port will be ringing. If port 1 and port 2 are busy, the port 3 will be ringing. If port 1, port 2 and port 3 are busy, the port 4 will be ringing.
Note: “123” will be NOT register to SIP proxy server when gateway is registering SIP proxy server mode Scenario description: Termination call to FXO for one-shoot call Port 1: FXO was connected to PSTN (area code is 81xxxxxxxx). SIP leading number “081x” incoming, and delete the first one digit “0”, and call to PSTN number. Note: “081x” will be NOT register to SIP proxy server when gateway is registering SIP proxy server mode.
T.30/T.38 real-time FAX compliant Voice/FAX auto-switch. FAX Mode Option The T.38 is a “Real Time Group 3 FAX communication over IP network” format. That’s meaning it’s a protocol for FAX over IP. You have to enable this function. SIP Telephone Advance Configuration If this function is enabled, when silence is occurred for a Silence Compression period of time, no data will be sent across the network during this period in order to save bandwidth. Disable / Enable dialing complete tone.
generate flash to FXS. Flash Generation: Let you change flash generation time (milliseconds) for PBX detection. Ring Frequency You can configure how long the Ring Frequency do you want to use. Enable battery reverse to detect polarity from PSTN line. FXO Battery Reverse The PSTN line can send SIP case: Sending the 200 OK connect signal to caller when detecting polarity reverse from PSTN Line.
the Phone (connected to the FXS port) was picked up by user. (Note: This case can avoid the Local PSTN charge when the FXS port still ring.) 6. Non Answer Mode: FXO will NOT answer the call in any time. (Note: Some ITSP only let the FXO for termination function, they do not user use the FXO port for origination) Scenario description: SIP call connecting answer mode Case B: “Hot Line Number” was assigned and the hot line number belongs to SIP device. 1.
SIP Netwrok Advance Configuration Smart-QoS If this function is enabled, when VoIP call is occurred, the other data will be automatically reduced traffic which across the internet in order to guarantee the voice bandwidth. Bandwidth control G.723/G.729 Bandwidth IP TOS You can configure your bandwidth what the Max byte of download and upload of ADSL modem rate. Enable / Disable Type of Service in IP packets.
Example: ITSP_A and ITSP_B are configured in this setting. - Step 2: Configure SIP Account You can find Set Register SIP Server Plan in Advance Setting – VoIP Basic Set Register SIP Server Plan Account Input SIP account(Username) Password Input Password that ITSP support. Expires This field sets how long an entry remains registered with the SIP register server. The register server can use a different time period.
- Step 3: Configure Least Cost Route You can find LCR Outgoing Dial Rule in Advance Setting – Dialing Plan LCR Outgoing Dial Rule Outgoing Number It’s is the leading digits of the call out dialing number Length Min Min allowed length you can dial. Length Maxi MAX allowed length you can dial. Delete Length the number of digits that will be stripped from beginning of the dialed number. Add Digits is the digits that will be added to the beginning of the dialed number.
According the rules, the gateway can achieve the goal: Least Cost Route by making calls through relatively cheaper ITSP providers. Port Status Port Status Display: This selection will display concurrent call status of this gateway. The status information of each voice channel includes codec, dialing number and destination IP address. The status is refreshed every 3 seconds.
Chapter 5 System Administrations 5 Management Management Label You can save configuration and restart the gateway with the default Save Configuration configuration or with the current running configuration. Access Control Users can sets/changes the administrator password... Set to Default You can restart the VIP-GW with the default configuration. Backup/Restore User can backup the configuration file of VPI-GW to PC or restore Configuration the configuration file from PC.
Save Configuration This page allows you to click “Save Configuration and Reboot” to save configuration and begin to restart. Access Control Changing the Administrator/Guest Password For security reasons, we strongly recommend that you set an administrator/password for the router. On first setup the router requires no password. If you don’t set a password the router is open and can be logged into and settings changed by any user from the local network or the Internet.
Backup/Restore Configuration to a File User can backup the configuration to a File at Microsoft Operation System. And also restore the configuration file to the VIP-GW from PC. System Information Display Function Click System Information Display to open the Online Status page. In the example, on the foll owing page, both PPPoE connections is up on the WAN interface, H323/SIP Status, MAC addr ess, Register Status.., etc.
SNTP Setting Function Click SNTP setting to open the Online Status page. In the example, on the following page: Use SNTP Setting— when checked, gateway uses a Simple Network Time Protocol (SNTP) to set the date and time. The gateway synchronizes the gateway’s time after you select the time zone. Use SNTP Setting; select the time zone which gateway was at. Syslog setting Use Syslog server to record your VIP-GW log file. To set the Syslog server IP address for this function.
Capture packetackets Function Use “Capturer Packets” to record VIP-GW packets. Users can start and stop the capture then save the file to PC. Use the Ethereal Tool (www.ethereal.com) to analyze the packets.
Appendix A Voice communications The chapter shows you the concept and command to help you configure your PLANET VIP-GW through sample configuration. And provide several ways to make calls to desired destination in VIP-GW. In this section, we’ll lead you step by step to establish your first voice communication via web browsers operations. Concepts: Voice port There are two type of the voice port, FXO (Foreign exchange Office) and FXS.
FXS (Foreign exchange Station) port The FXS port allows the connection to an end node, like telephone, fax machine, or out-line of PBX system. FXS port is as like your local phone service provider who provides a number to you. It is easy to tell that after you have connected an end-device to FXS port and you will hear the dial-tone from FXS port once the hand set off-hook. FXS 222 412-1111 0 Caution The FXS port is with voltage and current. DO NOT connects the port to any PBX extension line or PSTN line.
H.323 VoIP Call: Peer-To-Peer Mode Scenario 1: Gateway 1 to Gateway 2 PLAR connection H.323 Call (Peer-To-Peer Mode) Outgoing Dial plan Outgoing Dial plan No: 8x | Digit: 3~3 |Des: GW2 IP address No: 9x | Digit: 3~3 | Des: GW1 IP address x: wild card Des: Destination IP WAN Digit: Digit Length min~ max Gateway#2 Gateway#1 801 901 801 901 Scenario 2: Gateway 1 (with PBX) to Gateway 2 PLAR connection H.
Scenario 3: Gateway 1 (with PBX/PSTN) to Gateway 2 PLAR connection Call Method: Two-Stages-Dialing H.
Scenario 5: Gateway 2 to Gateway 1 (Remote Call PSTN number) PLAR connection Call Method: One-Shot-Dialing H.
H.323 VoIP Call: Gatekeeper Mode Scenario 7: Gateway 1 to Gateway 2 PLAR connection H.323 Call (GK Mode) Register Number List GW1: 801 GW2: 901 GK WAN Gateway#1 Gateway#2 801 801 901 901 Scenario 8: Gateway 2 to Gateway 1 (Call PBX extension number) PLAR connection Call Method: Two-Stages-Dialing H.
Scenario 9: Gateway 2 to Gateway 1 (Remote Call PSTN number with PBX) PLAR connection Call Method: Two-Stages-Dialing H.323 Call (GK Mode) with PBX: Remote Call PSTN number Method 1: Two-Stages-Dialing Register Number List GW1: 801 GW2: 901,609 GK Gateway#1 WAN Gateway#2 Extension 609 02-12345678 801 901 Trunk-Line 801 Extension 601 PSTN 609, 0, 23221344 Scenario 10: Gateway 2 to Gateway 1 (Remote Call PSTN number with PBX) PLAR connection Call Method: One-Shot-Dialing H.
Scenario 11: Gateway 2 to Gateway 1 (Remote Call PSTN number) PLAR connection Call Method: One-Shot-Dialing H.
SIP VoIP Call: Peer-To-Peer Mode Scenario 13: Gateway 1 to Gateway 2 PLAR connection SIP Call (Peer-To-Peer Mode) Outgoing Dial plan Outgoing Dial plan No: 8x | Digit: 3~3, Des | GW1 IP address No: 9x | Digit: 3~3, Des | GW1 IP address x: wild card Des: Destination IP WAN Digit: Digit Length min~ max Gateway#1 Gateway#2 801 801 901 901 Scenario 14: Gateway 2 to Gateway 1 (Call PBX extension number) PLAR connection Call Method: Two-Stages-Dialing SIP Call (Peer-To-Peer Mode) with PBX: Call PBX Ex
Scenario 15: Gateway 2 to Gateway 1 (Remote Call PSTN number with PBX) PLAR connection Call Method: Two-Stages-Dialing SIP Call (Peer-To-Peer Mode) with PBX: Remote Call PSTN number Method 1: Two-Stages-Dialing Outgoing Dial plan Outgoing Dial plan No: 8x | Digit: 3~3, Des | GW2 IP address No: 9x | Digit: 3~3, Des | GW1 IP address No: 6x | Digit: 3~3, Des | GW1 IP address x: wild card Gateway#1 Des: Destination IP WAN Gateway#2 Digit: Digit Length min~ max Extension 609 02-12345678 Trunk-Line 80
Scenario 17: Gateway 2 to Gateway 1 (Remote Call PSTN number) PLAR connection Call Method: One-Shot-Dialing SIP Call (Peer-To-Peer Mode) : Remote Call PSTN number Method: One-Shot-Dialing Outgoing Dial plan Outgoing Dial plan No: 8x | Digit: 3~3, Des | GW2 IP address No: 9x | Digit: 3~3 | Des: GW1 IP address Incoming Dial Plan No: 6x | Digit: 3~3 | Des: GW1 IP address No: 02x | Digit: 3~10 | Strip:2 | FXO port No: 02x| Digit: 3~10 | Des: GW 1 IP address x: wild card Gateway#1 WAN Gateway#2 Des
SIP VoIP Call: SIP Proxy Server Scenario 19: Gateway 1 to Gateway 2 PLAR connection SIP Call (Register to SIP Proxy Server Mode) Register Number List GW1: 801 SIP Proxy Server GW2: 901 WAN Gateway#2 Gateway#1 801 901 801 901 Scenario 20: Gateway 2 to Gateway 1 (Call PBX extension number) PLAR connection Call Method: Two-Stages-Dialing SIP Call (SIP Proxy Server Mode) with PBX: Call PBX Extension Method 1: Two-Stage-Dialing Register Number List GW1: 801 GW2: 901,609 SIP Proxy Server Gateway#1 WAN
Scenario 21: Gateway 2 to Gateway 1 (Remote Call PSTN number with PBX) PLAR connection Call Method: Two-Stages-Dialing SIP Call (SIP Proxy Server Mode) with PBX: Remote Call PSTN number Method: Two-Stages-Dialing Register Number List GW1: 801 SIP Proxy Server GW2: 901,609 Gateway#1 WAN Gateway#2 Extension 609 Trunk-Line 801 901 801 02-12345678 PSTN Extension 601 609, 0, 12345678 Scenario 22: Gateway 2 to Gateway 1 (Remote Call PSTN number) PLAR connection Call Method: Two-Stages-Dialing SIP Call
Scenario 23: Gateway 2 to Gateway 1 (PSTN Call PSTN number) PLAR connection Call Method: Two-Stages-Dialing SIP Call (SIP Proxy Server Mode) : PSTN Call PSTN number Method: Two-Stages-Dialing Register Number List GW1: 801,804 GW2: 901,904 SIP Proxy Server Gateway#1 Gateway#2 904 804 FXO FXO 02-87654321 02-12345678 WAN 901 04-56785678 801 87654321, 804, PSTN PSTN 56785678,904, 12345678 04-43221344 71
Appendix B FAQ Q: What is the default administrator password to login to the gateway? A: By default, your default username is “admin”; default password is “123” to login to the router. For security, you should modify the password to protect your gateway against hacker attacks. Note: Default guest login username/password: guest/guest Q: I forgot the administrator password. What should I do? A: Press the Reset button on the rear panel for over 5 seconds to reset all settings to default values.
FAQ 1: Firmware upgrade Requirement and Process 1. Environment Requirement a) A PC with FTP Server (Server-U software) b) A PC or Notebook witch connected to LAN port of gateway. c) Put the image (firmware) named “FW-VIP880_vxxx.bin ” at the assigned folder in FTP Server. For example: “FW-VIP880_v282.bin” is version 2.8.2L Note: Free FTP server: 172.16.0.101 username: xxxx, password: xxxx Environment Architecture (Gateway and FTP server are in Internet): 2.
FAQ 2: Busy Tone Learning STEP 1: Let the FXO port connect to PBX ext. STEP 2: To dial to the FXO port from PBX another ext. STEP 3: Hear dial tone, please dial FXS port number. When FXS port ring, please hang up the phone. STEP 4: At this moment, hear busy tone and also press “y” to learning.
FAQ 3: FXO Ringer Voltage Threshold / Ringer Voltage Filter Setting VIP-Gateway provides ring detector in FXO device avoiding can not answer and always OFF-HOOK status. This ring detector provides two functions to meet the various PBX’s extension port: 1. FXO ringer voltage threshold 2.
2. Voice Detection: Voice Detection-based answer supervision is a feature where the Gateway can be configured to “listen” on the line for different tones and voice. The Gateway sends a “connect” signals out or “disconnect” signaling using internet. H.323 scenario description: Loop Start Reverse Battery Æ PSTN line was set polarity reverse a) The gateway can send the “Q.931 connect” H.323 signals to Billing System of ITSP, after the user pick up the Phone and detect the PSTN line answer voltage.
Scenario description: Voice Detection based on answer supervision PSTN Line was not support Polarity Reverse: a) The gateway can send the 200 OK SIP signals to Billing system of ITSP, after the user pick up the Phone and detect the voice. b) The gateway can send the 200 BYE SIP signals to Billing system of ITSP, after the user hang up the phone and detect the hang up voice. c) This type of answer supervision is not 100% accurate.
Note: Some ITSP only let the FXO for termination function, they do not user use the FXO port for origination SIP Call Connecting Answer Mode Scenario B description: Hot Line Number” was assigned and the hot line number belongs to SIP device. a) When the call com from PSTN to FXO, FXO start the Hot line dialing to remote SIP gateway b) The phone of remote SIP gateway start ring. c) When the phone was picked up, the remote SIP Gateway sends “SIP 200 OK” signal to FXO port.
H.323 Call Connecting Answer Mode Scenario B description: Hot Line Number” was assigned and the hot line number belongs to remote H.323 device. Note: The remote H.323 device need disable the “Auto Answer” a) When the call com from PSTN to FXO, FXO start the Hot line dialing to remote H.323 gateway b) The phone of remote H.323 gateway start ring. c) When the phone was picked up, the remote H.323 Gateway send “Q.931 connects” signal to FXO port. Once FXO port receives the “Q.
FAQ 6: Peer to Peer call: FXO to FXO Scenario description: User (500) on site A (VIP-880FO) wishes to have telephone calls to extension (600) on Site B (VIP-880FO B). User (500) on site B (VIP-880FO B) can connect to ser A in the same way. IP address of VIP-880FO_A is: 172.16.0.1 VIP-880FO_A number and dial plan setting: Each port number is 100,200,300,400,500,600,700,800 IP address of VIP-880FO_B is: 172.16.0.
The dial plan of VIP-880FO_B dial plan setting: that means call 0xxx leading number go to IP address 172.16.0.1 gateway (VIP-880FO_A). Usage: The ext.509 dial to ext 501 (connect to FXO port 1) will hear the dial tone, and then dial 0100 go to IP address gateway 172.16.0.2 (VIP-880FO_B), and hear the dial tone again, then dial 609 ext, the ext.609 will ring.
IP address of VIP-880FO is: 172.16.0.1 VIP-880FO number and dial plan setting: Each port number is 100,200,300,400,500,600,700,800 IP address of VIP-880FS is: 172.16.0.2 VIP-880FS number and dial plan setting: Each port number is 100,200,300,400,500,600,700,800 The dial plan of VIP-880FO dial plan setting: that means call 0xxx leading number go to IP address 172.16.0.2 gateway (VIP-880FS). The dial plan of VIP-880FS dial plan setting: that means call 0xxx leading number go to IP address 172.16.0.
Usage: The ext.509 dial to ext 501 (connect to FXO) will hear the dial tone, and then dial 0100 go to IP address gateway 172.16.0.2 (VIP-880FS), and phone of port 1(100) will ring. FAQ 8: Peer to Peer call for one shoot dialing: FXO to FXO Scenario description: User (500) on site A (VIP-880FO) wishes to have telephone calls to extension (600) on Site B (VIP-880FO B). User (500) on site B (VIP-880FO B) can connect to ser A in the sam way. IP address of VIP-880FO_A is: 172.16.0.
The dial plan of VIP-880FO_A: that means call 0xxx leading number will go to IP address 172.16.0.2 gateway (VIP-880FO_B). VIP-880FO_B number and dial plan setting: The dial plan of VIP-880FO_B dial plan setting: that means incoming call 60x leading number will hunt port 1 to port 8 The dial plan of VIP-880FO_B: That means call 0xxx leading number will go to IP address 172.16.0.1 gateway (VIP-880FO_A). Usage: The ext.
FAQ 9: Peer to Peer call: Hotline setting Hot line Basic Concept: Any number set in Hot line field will be dialed by VoIP call automatically. For FXS port case: When user picks up the phone, the gateway will dial the hot line number to internet by VoIP call. For FXO port case: When the FXO off-hook (PSTN call coming or PBX extension ring in), the gateway will dial the hot line number to internet by VoIP call. Scenario description: Peer to Peer direct call via SIP or H.
Scenario description: Register to SIP proxy server/H.323 Gatekeeper direct call STEP 1: Let your VIP-GW register to SIP proxy or H.323 Gatekeeper server STEP 2: To set hot line number in Hot Line Option STEP 3: When users pick up the phone (port1), the gateway will dial the “911” to SIP proxy or H.323 Gatekeeper server (ITSP) Scenario description: Register to SIP Proxy / H.323 Gatekeeper server and Peer to Peer direct call first STEP 1: Let your VIP-GW register to SIP proxy or H.
STEP 4: When users pick up the phone (port 1), the gateway will dial the “911” to the gateway which’s IP address 172.16.0.119. Note: This call will not call to SIP Proxy or H.323 Gatekeeper server because of direct call first. FAQ 10: SIP speed call setting Speed calls Concept: Cut your phone number down to fewer digit dialing! Life is moving fast - you've got to dial fast. Now you can with Speed Dial. Dial the people you call most with just dialing fewer digits instead of dialing the full phone number.
Scenario description C: User wants to dial 888 instead of 810-861234567890 Note: The destination IP address is the domain name of sip proxy server 88
Appendix C VIP-281 series Specifications Product Model Hardware WAN LAN Voice Protocols and Standard Standard Voice codec Fax support Voice Standard Protocols Advanced Function 2-Port H.323/SIP VoIP Gateway VIP-281 VIP-281FS 1 x 10/100Mbps RJ-45 port 4 x 10/100Mbps RJ-45 port 2 x RJ-11 connection ( 1 x FXS, 1 x FXO) 2 x RJ-11 connection ( 2 x FXS) H.
VIP-480 series Specifications Product Model Hardware WAN LAN Voice Protocols and Standard Standard Voice codec Fax support Voice Standard Protocols Advanced Function 4-Port H.323/SIP VoIP Gateway VIP-480 VIP-480FS VIP-480FO VIP-480FD 1 x 10/100Mbps RJ-45 port 4 x 10/100Mbps RJ-45 port 4 x RJ-11 4 x RJ-11 connection connection ( 2 x FXS, 2 x FXO) ( 4 x FXS) 4 x RJ-11 connection ( 4 x FXO) 4 x RJ-11 connection ( 4 x FXO with Caller ID) H.
VIP-880 series Specifications Product Model Hardware WAN LAN Voice Protocols and Standard Standard Voice codec Fax support Voice Standard Protocols Advanced Function Network and Configuration Access Mode Management LED Indications Dimension (W x D x H) Operating Environment Power Requirement EMC/EMI 8-Port H.323/SIP VoIP Gateway VIP-880 VIP-880FO 1 x 10/100Mbps RJ-45 port 1 x 10/100Mbps RJ-45 port 8 x RJ-11 connection ( 4 x FXS, 4 x FXO) 8 x RJ-11 connection ( 8 x FXO) H.
VIP-1680 series Specifications Product Model Hardware WAN LAN Voice Protocols and Standard Standard Voice codec Fax support Voice Standard Protocols Advanced Function Network and Configuration Access Mode Management LED Indications Dimension (W x D x H) Operating Environment Power Requirement EMC/EMI 16-Port H.
VIP-2480 series Specifications Product Model Hardware WAN LAN Voice Protocols and Standard Standard Voice codec Fax support Voice Standard Protocols Advanced Function Network and Configuration Access Mode Management LED Indications Dimension (W x D x H) Operating Environment Power Requirement EMC/EMI 24-Port H.
