Planet Networking & Communication VGW 20FS Series User Manual

Planet Networking & Communication VGW 20FS Series User Manual

4-/8-/16-/24-/32-port sip voip gateway
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4-/8-/16-/24-/32-Port SIP VoIP
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VGW-x20FS Series

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Summary of Contents for Planet Networking & Communication VGW 20FS Series

  • Page 1 4-/8-/16-/24-/32-Port SIP VoIP Gateway VGW-x20FS Series...
  • Page 2 4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Copyright Copyright (C) 2020 PLANET Technology Corp. All rights reserved. The products and programs described in this User’s Manual are licensed products of PLANET Technology. This User’s Manual contains proprietary information protected by copyright, and this User’s Manual and all accompanying hardware, software, and documentation are copyrighted.
  • Page 3 4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway WEEE Warning To avoid the potential effects on the environment and human health as a result of the presence of hazardous substances in electrical and electronic equipment, end users of electrical and electronic equipment should understand the meaning of the crossed-out wheeled bin symbol.
  • Page 4: About This Manual

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Preface Welcome Thanks for choosing VGW-X20FS SERIES VoIP Gateway. We hope you will make optimum use of this flexible, feature-rich VoIP-to-FXS gateway. Please read this document carefully before installing the gateway. About this manual This manual provides information about the introduction of the gateway, and about how to install, configure or use the gateway.
  • Page 5: Table Of Contents

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Contents Preface ................................... III Welcome ...........................III About this manual ......................III Intended audience......................III 1 Introduction of VGW-X20FS SERIES ........................1 1.1 Overview ........................1 1.2 Product Features ......................2 1.3 Function Specifications ....................3 1.3.1 VGW-420FS / VGW-820FS ....................3 1.3.2 VGW-1620FS ........................
  • Page 6 4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 2.5.2 Attended Transfer ......................16 2.6 Three-way Calling ...................... 17 2.7 Description of Feature Codes ..................17 2.8 Sending and Receiving Fax ..................18 2.8.1 T. 38 and Pass-through ....................18 2.9 Local IVR Operation ....................19 2.9.1 Inquire IP address ......................
  • Page 7 4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.5.7 ARP ..........................30 3.6 SIP Server ........................31 3.7 Port ..........................33 3.8 Advanced ........................35 3.8.1 FXS/FXO Parameters ....................35 3.8.2 Media Parameter......................37 3.8.3 SIP Parameters ......................38 3.8.4 Fax Parameter ......................43 3.8.5 Digit Map ........................
  • Page 8 4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.11.1 Route any calls from any IP to specific port ............... 57 3.11.2 Route any calls from any IP to specified port group ..........58 3.11.3 Route any calls from any port to specific SIP IP trunk ..........59 3.12 Maintenance ......................
  • Page 9: Introduction Of Vgw-X20Fs Series

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Introduction of VGW-X20FS SERIES 1.1 Overview High Quality yet Affordable for All Businesses PLANET VGW-x20FS series enterprise-class 4-/8-/16-/24-/32-port SIP VoIP Gateway provides added flexibility during migration to Unified Communications by supporting the traditional analog devices. These devices include analog phones, fax machines, modems, voicemail systems and speakerphones.
  • Page 10: Product Features

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 1.2 Product Features SIP Applications  IETF SIP RFC 3261 based on UDP/TCP/TLS  4-/8-/16-/24-/32-line FXS connects to analog phone set or PABX  Fax over T.38 and Pass-through ITU-T G.711 A-law, G.711 μ-law, G.723.1 and G.729 voice coding ...
  • Page 11: Function Specifications

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 1.3 Function Specifications VGW-420FS / VGW-820FS 1.3.1 Product VGW-420FS VGW-820FS Hardware 1 x 10/100BASE-TX RJ45 port 1 x 10/100BASE-TX RJ45 port 3 x 10/100BASE-TX RJ45 port 4 x RJ11 connection (4 x Foreign 8 x RJ11 connection (8 x Foreign Voice eXchange Station) eXchange Station)
  • Page 12 4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway  Outbound Proxy  DNS SRV/A Query/NATPR Query  SIP Trunk  Early Media/Early Answer  NAT:STUN, Static/Dynamic NAT  Call Waiting  Blind Transfer  Attend Transfer  Call Forward on Busy  Call Forward on No Reply ...
  • Page 13: Vgw-1620Fs

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway VGW-1620FS 1.3.2 Product VGW-1620FS Hardware 4 x 10/100BASE-TX RJ45 port Voice 16 x RJ11 connection (32 x Foreign eXchange Station) Console 1 x RS232, 115200bps 2700g Weight Dimensions (W x D x H) 440 x 230 x 44 mm Power Requirements 100-240VAC, 50-60 Hz Power Consumption...
  • Page 14 4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway  DNS SRV/A Query/NATPR Query  SIP Trunk  Early Media/Early Answer  NAT:STUN, Static/Dynamic NAT  Call Waiting  Blind Transfer  Attend Transfer  Call Forward on Busy  Call Forward on No Reply ...
  • Page 15: Vgw-2420Fs / Vgw-3220Fs

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway VGW-2420FS / VGW-3220FS 1.3.3 Product VGW-2420FS VGW-3220FS Hardware 4 x 10/100BASE-TX RJ45 port 24 x RJ11 connection (24 x Foreign 32 x RJ11 connection (32 x Foreign eXchange Station) eXchange Station) Voice 1 x RJ21, 50 PIN 2 x RJ21, 50 PIN Console 1 x RS232, 115200bps...
  • Page 16 4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway  Primary/Backup SIP Server  Outbound Proxy  DNS SRV/A Query/NATPR Query  SIP Trunk  Early Media/Early Answer  NAT:STUN, Static/Dynamic NAT  Call Waiting  Blind Transfer  Attend Transfer  Call Forward on Busy ...
  • Page 17: Ports And Connectors

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 1.4 Ports and Connectors The FXS analog gateway is available in the following configurations: Model Voice Channels FXS Ports Physical Port Labels VGW-420FS VGW-820FS VGW-1620FS 0-15 VGW-2420FS 0-23 VGW-3220FS 0-31 VGW-420FS 1.4.1...
  • Page 18: Vgw-820Fs

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Port Name Connector Description Power Jack Power Jack To connect DC 12V power supply To connect to the IP network over a DSL modem or router or WAN/LAN Port RJ45 a LAN switch FXS ports to connect standard analog phone or fax machine FXS Ports 0-3 RJ11 or a PBX...
  • Page 19: Vgw-1620Fs

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Port Name Connector Description Power Jack Power Jack To connect DC 12V power supply To connect to the IP network over a DSL modem or router or a LAN WAN/LAN 0-2 Port RJ45 switch FXS ports to connect standard analog phone or fax machine or a FXS Ports 0-7 RJ11 VGW-1620FS...
  • Page 20: Vgw-2420Fs

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Port Name Connector Description Power Jack Power Jack To connect 100-240V AC 50-60HZ power supply To connect to the IP network over a DSL modem or router or a LAN LAN Port 0-3 RJ45 switch FXS ports to connect standard analog phone or fax machine or a FXS Ports 0-15 RJ11...
  • Page 21: Vgw-3220Fs

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Port Name Connector Description Power Jack Power Jack To connect 100-240V AC 50-60HZ power supply To connect to the IP network over a DSL modem or router or a LAN LAN Ports 0-3 RJ45 switch FXS ports to connect standard analog phone or fax machine or a FXS Ports 0-24 RJ11...
  • Page 22: Functions And Features

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Port Name Connector Description Power Jack Power Jack To connect 100-240V AC 50-60HZ power supply To connect to the IP network over a DSL modem or router or a LAN Ports 0-3 RJ45 LAN switch FXS ports to connect standard analog phone or fax machine or a FXS Ports 0-31 RJ11...
  • Page 23: Basic Operations

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Basic Operations 2.1 Methods to Number Dialing Dial mobile phone or extension number Dial the number directly and wait for 3 seconds (Default “No dial timeout”); Dial the number directly and press #. 2.2 Direct IP Calls The VGW-x20FS series gateway allows users to directly call through IP address.
  • Page 24: Call Holding

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 2.3 Call Holding Place a call on hold by pressing the “flash” button on the analog phone (if the phone has the button). Press the “flash” button again to release the previously held caller and resume conversation. If no “flash” button is available, use “hook flash”...
  • Page 25: Three-Way Calling

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 2.6 Three-way Calling Three-way calling A calls B,B picks up the phone, then A and B enters into conversation; A presses the hook flash, and the call between A and B is placed on hold. Then C calls A and A answers the call.
  • Page 26: Sending And Receiving Fax

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Dial *# to place a call on hold *47* Dial *47* to establish a call through IP address *51# Dial *51# to enable ‘call waiting’ feature *50# Dial *50# to disable ‘call waiting’ feature *87* Dial *87* to blind transfer a call *72* Dial *72* to enable ‘unconditional call forwarding’...
  • Page 27: Local Ivr Operation

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 2.9 Local IVR Operation Inquire IP address 2.9.1 Connect analog phone to FXS ports of the VGW-X20FS SERIES gateway, then pick up the phone. After dialing tone, dial *158# to inquire the IP address of LAN port and dial *159# to inquire the IP address of WAN port.
  • Page 28: Configurations On Web Interface

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Configurations on Web Interface 3.1 Logging in Web Interface The VGW series is easy to install by following the steps below. Step 1:Connect a computer to a LAN port on the VGW series. Your PC must be set to 192.168.0.X, the same domain as that of the VGW series.
  • Page 29: State And Statistics

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.3 State and Statistics System Information 3.3.1 On the System Information interface, you can view the information of device ID, MAC address, network mode, IP addresses, version information, sever register status and so on. Figure 3.5-1 System Information Explanation of items on System Information interface Device ID A unique ID of each device.
  • Page 30 4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Subnet Mask: The netmask of the router connected to the VGW-X20FS SERIES;  Default Gateway: The IP address of the router connected to the VGW-X20FS SERIES;  PPPoE: PPPoE is an acronym for point-to-point protocol over Ethernet, which relies on two widely accepted standards: PPP and Ethernet.
  • Page 31: Registration Information

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Registration Information 3.3.2 Figure 3.5-2 Port and Port Group Registration Information Primary/Secondary User status: Registered: The port is registered to SIP server successfully; Unregistered: The port fails to be registered to SIP server. TCP/UDP Statistics 3.3.3 Figure 3.5-3 TCP/UDP Statistics Information The above interface shows the statistical number of sending or receiving packets over TCP, and the...
  • Page 32: Cdr Statistics

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway CDR Statistics 3.3.5 CDR (Call Detail Record) is a data record produced by a telephone exchange or a telecommunication device, which contains the details of a telephone call that passes through the device. On the Status & Statistic  CDR interface, details of all calls through the ports of the VGW-X20FS SERIES are displayed.
  • Page 33 4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway PPPoE: PPPoE is an acronym for point-to-point protocol over Ethernet, which relies on two widely accepted standards: PPP and Ethernet. PPPoE is a specification for connecting the users on an Ethernet to the Internet through a common broadband medium, such as a single DSL line, wireless device or cable modem.
  • Page 34 4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Figure 3.7-2 Bridge Mode If DHCP is selected to obtain IP address, please ensure DHCP server in the network works normally. When the gateway works in the route mode, the IP address of LAN port and that of WAN port cannot be in the same network segment, otherwise the gateway can’t work normally.
  • Page 35: Vlan(Virtual Local Area Network

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway VLAN(Virtual Local Area Network) 3.5.2 In order to control the impacts brought by broadcast storms, user can divide VLANs into three groups, namely VLAN1, VLAN2 and VLAN3. There are three kinds of VLANs, including data VLAN, voice VLAN and management VLAN.
  • Page 36: Dhcp Server (Route Mode For Vgw-420Fs/820Fs)

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Explanations of the parameters in VLAN interface: VLAN1/VLAN2/VLAN3 The gateway supports three VLANs at most. Please enable VLAN according to actual needs. If the checkboxes on the right of data, voice and management of VLAN1 are selected, it means Data/Voice/Management,...
  • Page 37: Dmz Host (Route Mode For Vgw-420Fs/820Fs)

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway DMZ Host (Route Mode for VGW-420FS/820FS) 3.5.4 If the DMZ service is enabled, the devices in the wide-area network are allowed to have direct access to the devices in the DMZ (demilitarized zone). In this way, devices in the wide-area network can visit the devices which are in the local area network and meanwhile the devices in the local area network are protected.
  • Page 38: Static Route (Route Mode For Vgw-420Fs/820Fs)

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway When both DMZ Host and forwarding rule are configured, the configuration of forwarding rule is prior to that of DMZ Host. Static Route (Route Mode for VGW-420FS/820FS) 3.5.6 Static route determines the routing rule during the handling of messages by the gateway. Most of time, user does not need to configure static route.
  • Page 39: Sip Server

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.6 SIP Server Introduction of SIP Server: 1)SIP server is the main component of VoIP network and is responsible for establishing all the SIP calls. SIP server is also called SIP proxy server or register server. Both IPPBX and softswitch can act as the role of SIP server.
  • Page 40 4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Explanation of SIP parameters: The IP address or domain name of the primary SIP server is provided by Primary SIP Server Address VoIP service provider. Primary SIP Server port The Service port of the primary SIP server is 5060 by default. It is used to avoid excessively frequent registrations.
  • Page 41: Port

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.7 Port Figure 3.9-1 Port Configuration Interface...
  • Page 42 4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Explanation of port parameters: Port Port number Disable port Whether to disable port temporarily Registration Whether to enable registration for the port Primary/Secondary Primary /Secondary SIP account description. It is used to identify the SIP account Display Name Primary/Secondary User account information provided by VoIP service provider (ITSP).
  • Page 43: Advanced

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.8 Advanced FXS/FXO Parameters 3.8.1 FXS parameters include: timeout Call Progress Tone, Timeout for Dialing, Send Polarity Reversal, etc. Figure 3.10-1 Configuration Interface for FXS Parameters...
  • Page 44 4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Explanation of FXS parameters: With the help of dialing timeout, you can limit the time between two digits while users are typing the digits of a number through an extension. If the timeout Timeout for dialing expires, the gateway will consider the dialing has finished and will try to send message to SIP server.
  • Page 45: Media Parameter

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Media Parameter 3.8.2 Media parameters mainly include: RTP start port, DTMF parameter, Preferred Vocoder, etc. Figure 3.10-2 Configuration Interface for Media Parameters Explanation of media parameters: If this parameter is enabled, the gateway will choose a port at random as the start Use Random Port port for RTP.
  • Page 46: Sip Parameters

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway SIP Parameters 3.8.3...
  • Page 47 4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Figure 3.10-3 SIP Parameter Configuration Interface Explanation of SIP parameters: SUBSCRIBE for MWI (Message Waiting You will be notified when ‘voicemail message waiting indicator’ is enabled. Indicator) MWI subscription expiry time; default value is 3600s. MWI Subscription Expires The user ID for access to voicemail box Voicemail User ID...
  • Page 48 4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Send BYE when Recv REFER Response If this parameter is enabled, the third party will send BYE to release (unattended) session after receiving REFER during a blind transfer. If this parameter is enabled, the value of ‘expires’ header will be Send New REGISTER when Recv 423 automatically updated and REGISTER will be re-sent after receiving of Response...
  • Page 49 4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Voicemail instructions: How the voicemail works in the VGW-X20FS SERIES gateway together with Elastix. 1)After the gateway is registered to Elastix server, enable the voicemail function in Elastix for the corresponding extension number and then set password shown below: Elastix Voicemail Configuration Interface 2)...
  • Page 50 4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3)Set ringing time in Elastix. Elastix will prompt user to leave a message after the corresponding extension rings 15 seconds (by default). Then the Elastix sever will record the message. Related setting is shown as follows: Voicemail Setting 4)Dial *200# on the extension which is connected to VGW-X20FS SERIES, then dial voicemail user ID and enter password for authentication.
  • Page 51: Fax Parameter

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Fax Parameter 3.8.4 Figure 3.10-4 Configuration Interface for Fax Parameter Explanation of fax parameters: Fax Mode There are four fax modes: T.38, T.30 (Pass-through), Modem and Adaptive. Include “a=X-fax” Attribute If this parameter is enabled, “a=X-fax” attribute will be carried in SDP. Include “a=fax”...
  • Page 52: Digit Map

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Digit Map 3.8.5 Figure 3.10-5 Digit Map Digit Map Syntax Supported Digit objects Timer DTMF A digit, a timer, or one of the symbols of A, B, C, D, #, or *. Range One or more DTMF symbols enclosed in the [], but only one DTMF symbol can be selected.
  • Page 53: Feature Codes

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Feature Codes 3.8.6 Please make reference to 2.7 Description of Feature Codes and the following table. Inquiry LAN port IP address Dial*158# to obtain device WAN port IP address Inquiry WAN port IP address Dial*159# to obtain device WAN port IP address (For VGW-420FS/VGW-820FS only) Inquiry Phone Number Dial*114# to obtain port account...
  • Page 54: System Parameter

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway System Parameter 3.8.7 System parameters include: STUN, NTP, Provision, EB parameter and Telnet. 1) STUN: STUN (Simple Traversal of UDP over NATs) is a lightweight protocol that allows applications to discover the presence and types of NATs and firewalls between them and the public Internet. It also provides the ability for applications to determine the IP addresses allocated to them by the NAT.
  • Page 55: Action Url

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Explanation of related parameters: Hint Language IVR language of the gateway NAT Traversal User can choose ‘Disable’, ‘ STUN’, ‘static NAT’ and ‘dynamic NAT’. To Enable or disable NTP Primary NTP server address The IP address of primary NTP server; default IP address is us.pool.ntp.org. Primary NTP server port The service port of primary NTP server;...
  • Page 56: Call & Routing

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.9 Call & Routing Wildcard Group 3.9.1 Figure 3.11-1 Wildcard Group Port Group 3.9.2 On the Port Group interface, user can group several ports together and then set a strategy for port selection of the group. Parameters of port group include registration, primary display name, primary SIP user ID, primary authentication ID and password, secondary display name, secondary SIP user ID, secondary authentication ID and password, off-hook auto dial, auto dial delay time, port select and so Figure 3.11-2 Configuration Interface for Port group...
  • Page 57 4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Explanation of related parameters Index The No. of the port group; it uniquely identifies a route, ranging from 0 to 7. Description Port group display is used in SIP message like the examples below: INVITE sip:bob@biloxi.com SIP/2.0 Via:SIP/2.0/UDPpc33.atlanta.com;branch=z9hG4bK776asdhds Max-Forwards: 70 Primary/Secondary Display Name To: Bob <sip:bob@biloxi.com>...
  • Page 58: Ip Trunk

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway IP Trunk 3.9.3 A peer-to-peer VoIP call occurs when two VoIP phones communicate directly over IP network without IP PBXs between them. IP trunk helps establish peer-to-peer call between gateway and VoIP phones. IP trunk will be used in routing configuration. Figure 3.11-3 IP Trunk Configuration Interface Explanation of related parameters: Index...
  • Page 59: Ip -> Tel Routing

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway IP -> Tel Routing 3.9.5 Figure 3.11-5 Configuration Interface for IP-Tel Routing Explanation of related parameters: Index IP Routing priority: from 0 to127; 0 is the highest priority. Description It is used to identify the IP  routing Calls from IP Trunk or SIP Server;...
  • Page 60: Tel-Ip/Tel Routing

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Tel-IP/Tel Routing 3.9.6 Figure 3.11-6 Configuration Interface for Tel-IP/Tel Routing Explanation of related parameters: The index of this Tel IP/Tel routing, from 0 to 127. Each index cannot be used repeatedly. Routing Index priority: 0 is the highest priority. Description It is used to identify the routing Calls From...
  • Page 61: Ip - Ip Routing

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway IP – IP Routing 3.9.7 Figure 3.11-7 Configuration Interface for IP->IP Routing Explanation of related parameters: The index of this IP IP routing, from 0 to 127. Each index cannot be used repeatedly. Routing priority: Index 0 is the highest priority.
  • Page 62: Manipulation Configuration

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.10 Manipulation Configuration Number manipulation refers to the change of a called number or a caller number during calling process when the called number or the caller number matches the preset rules. IP -> Tel Callee 3.10.1 Figure 3.12-1 Add IP ->...
  • Page 63: Tel -> Ip/Tel Caller

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Tel -> IP/Tel Caller 3.10.2 Figure 3.12-2 Add Tel -> IP Caller Configuration parameters are the same as those of ‘IP->Tel Callee’.
  • Page 64: Tel-Ip/Tel Callee

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Tel-IP/Tel Callee 3.10.3 Figure 3.12-3 Add Tel-IP Callee Configuration parameters are the same as those of ‘Tel->IP Caller’.
  • Page 65: Routing Rule Examples

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.11 Routing rule examples Route any calls from any IP to specific port 3.11.1 After entering the Web interface, click Call & Routing  IP-Tel Routing in the navigation tree on the left, and then click Add to create a new routing rule. In the example above, all calls will be routed to port 0 when the routing rule is matched.
  • Page 66: Route Any Calls From Any Ip To Specified Port Group

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Route any calls from any IP to specified port group 3.11.2 Create port group Before we can route calls to a port group, create the port group first as shown below. On the Call & Routing ...
  • Page 67: Route Any Calls From Any Port To Specific Sip Ip Trunk

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Route any calls from any port to specific SIP IP trunk 3.11.3 Create IP Trunk on the Call & Routing  IP Trunk interface: After IP Trunk is created, check the following configuration: As shown above, the IP trunk is created, and the remote end IP address is 172.16.125.125, the SIP port is 5060.
  • Page 68 4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway All Tel calls from any caller number to any called number will be routed to IP trunk 127.
  • Page 69: Maintenance

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.12 Maintenance TR069 3.12.1 ACS URL (auto-configuration server URL address) is provided by service provider. The ACS URL generally starts with http:// or https:// Username and password are used for ACS authentication. Figure 3.14-1 TR069 Parameters SNMP (Simple Network Management Protocol) 3.12.2 SNMP Parameters:...
  • Page 70 4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Figure 3.14-2 SNMP Parameters User configuration is only available on SNMP v3.
  • Page 71 4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Group configuration Group: community group name which consist of character string. Community: let community join the community group which configured above Trap configuration Trap configuration is enabled to configure Trap Server IP and port. This setting is available for SNMP v2c and v1.
  • Page 72: Syslog

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Syslog 3.12.3 Syslog is a standard for network device data logging. It allows separation of the software that generates messages from the system that stores them and the software that reports and analyzes them. It also provides devices which would otherwise be unable to communicate a means to notify administrators of problems or performance.
  • Page 73: Provision

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Figure 3.14-3 Syslog Parameter Enable send CDR, and then send communication information to syslog server. Provision 3.12.4 Provision is used to make the VGW-X20FS SERIES automatically upgrade with the latest firmware stored on an http server an ftp server or a tftp server. Figure 3.14-4 Provision Provisioning server URL and supporting HTTP, TFTP, FTP Check Interval...
  • Page 74: Cloud Server

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Cloud Server 3.12.5 User can register the gateway to cloud server, and then the gateway will be managed by cloud server. Figure 3.14-5 Cloud Server Explanation of related parameters Server Address The IP address or domain of the cloud server port The listening port of the cloud server Password...
  • Page 75: Security

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.13 Security WEB ACL 3.13.1 ACL (Access Control List) for Web is used to configure IP addresses (users) that are allowed to access the Web page of the gateway. The IP address list can’t be null once ACL is enabled. Figure 3.15-1 ACL for WEB Telnet ACL 3.13.2...
  • Page 76: Passwords

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Passwords 3.13.3 On the following interface user can configure or modify the username and password for access to the Web interface and the Telnet interface. Both the username and password of Web and Telnet are ‘admin’ and ‘admin’. Figure 3.15-3 Password Modification...
  • Page 77: Tools

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.14 Tools Firmware upload 3.14.1 Firmware upload steps: Step 1. Check the current firmware version on the System Information page Firmware Version Figure 3.16-1 Step 2. Prepare firmware package. The most important is that the package must match with the existing version. Package version consists of the following parts: 1.18.xx.xx 01/02 is vendor name...
  • Page 78: Data Backup

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Step 5. Reboot gateway. Refer to web page Maintenance-> Device Restart Figure 3.16-4 Restart Gateway Data Backup 3.14.2 process data backup: Click “Data Backup” Click “Backup” to backup data to PC. Figure 3.16-5 Data Backup Data Restore 3.14.3 The processes of data restore:...
  • Page 79: Tracert Test

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Figure 3.16-7 Ping Test Tracert Test 3.14.5 Tracert is a trace router used to track routing. Tracert sends a sequence of Internet Control Message Protocol (ICMP) echo request packets addressed to a destination host. Determining the intermediate routers traversed involves adjusting the time-to-live (TTL), aka hop limit, Internet Protocol parameter.
  • Page 80: Outward Test

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Figure 3.16-8 Tracert Test Outward Test 3.14.6 Outward test enables user to diagnose the physical phone lines which follow GR909 standards. To start outward test, select the ports to be tested and click ‘start’. Testing will take a few minutes. Figure 3.16-9 Outward Test Test results OK: The analog phone set and phone line are working well...
  • Page 81: Network Capture

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Failed: Analog phone could not be connected to FXS port or there’s something wrong with the phone set Network Capture 3.14.7 Network capture is a very important diagnostic tool for maintenance. It can be used to capture data packages of the available network ports.
  • Page 82 4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway  Click “Start’ to enable syslog capture  Dialing out through gateway, start talking a short while and then hanging up the call.  Click ‘Stop’ to disable syslog capture  Save the capture to local computer The capture is named as ‘capture(x).pcap’;...
  • Page 83 4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Getting started with DSP capture DSP capture helps to analyze voice stream inside the DSP chipset. The DSP chipset handles RTP from IP network as well as voice stream from analog phone. To enable DSP capture: ...
  • Page 84: Factory Reset

    4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Configurable capture options Getting started with custom capture This menu provides more options to capture specific packets according to actual needs. Factory Reset 3.14.8 Click ‘Apply’ to restore the factory settings. Factory Reset Device Restart 3.14.9 After saving all the configurations or changes to the equipment, user can restart the VGW-X20FS SERIES gateway for the changes to take effect.

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