4-/8-/16-/24-/32-Port SIP VoIP Gateway VGW-x20FS Series
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.
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. Do not dispose of WEEE as unsorted municipal waste and have to collect such WEEE separately. Trademarks The PLANET logo is a trademark of PLANET Technology.
-/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.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Contents Preface ................................................................................................................................................................... III Welcome ...........................................................................................................................III About this manual .............................................................................................................
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 ...............................................
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 .................................
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 ................................................................................................
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 1.1 Overview 1 Introduction of VGW-X20FS SERIES 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.
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 In-band/out of band DTMF (RFC 4733, RFC 2833 and SIP INFO) Echo cancellation exceeding ITU-T G.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 1.3 Function Specifications 1.3.
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 Supplementary Service Unconditional Call Forward Warm/Immediately Hotline Call Hold Do-not-disturb 3-way Conferencing Message Waiting Indicator Hunting Group Web ACL Telnet ACL Action URL Software Features PPPoE/IPv4
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 1.3.
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 Supplementary Service Unconditional Call Forward Warm/Immediately Hotline Call Hold Do-not-disturb 3-way Conferencing Message Waiting Indicator Hunting Group Web ACL Telnet ACL Action URL Software Features PPPoE/IPv4/IPv6 Digitmap
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 1.3.
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 Call Forward on No Reply Supplementary Service Unconditional Call Forward Warm/Immediately Hotline Call Hold Do-not-disturb 3-way Conferencing Message Waiting Indicator Hunting Group Web ACL Telnet ACL Action URL Sof
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 VGW-420FS 4 4 VGW-820FS 8 8 VGW-1620FS 16 16 VGW-2420FS 24 24 VGW-3220FS 32 32 1.4.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Port Name Connector Description Power Jack Power Jack To connect DC 12V power supply WAN/LAN Port RJ45 To connect to the IP network over a DSL modem or router or a LAN switch FXS Ports 0-3 RJ11 FXS ports to connect standard analog phone or fax machine or a PBX 1.4.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Port Name Connector Description Power Jack Power Jack To connect DC 12V power supply WAN/LAN 0-2 Port RJ45 To connect to the IP network over a DSL modem or router or a LAN switch FXS Ports 0-7 RJ11 FXS ports to connect standard analog phone or fax machine or a PBX 1.4.
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 LAN Port 0-3 RJ45 To connect to the IP network over a DSL modem or router or a LAN switch FXS Ports 0-15 RJ11 FXS ports to connect standard analog phone or fax machine or a PBX Console Port RJ48 Console port is used to carry out maintenance-related configurations 1.4.
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 LAN Ports 0-3 RJ45 To connect to the IP network over a DSL modem or router or a LAN switch FXS Ports 0-24 RJ11 FXS ports to connect standard analog phone or fax machine or a PBX Console Port RJ48 Console port is used to carry out maintenance-related configurations 1.4.
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 LAN Ports 0-3 RJ45 To connect to the IP network over a DSL modem or router or a LAN switch FXS Ports 0-31 RJ11 FXS ports to connect standard analog phone or fax machine or a PBX Console Port RJ48 Console port is used to carry out maintenance-related configurations 1.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 2 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.
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” instead. 2.
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. A presses hook flash again, then the calls between A and B and between A and C are placed on hold.
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# *50# *87* Dial *51# to enable ‘call waiting’ feature Dial *50# to disable ‘call waiting’ feature Dial *87* to blind transfer a call *72* *73# Dial *72* to enable ‘unconditional call forwarding’ feature Dial *73# to disable ‘unconditional call forward’ feature *90* Dial *90* to enable ‘busy call forwarding’ feature *91# *92* Dial *91# to disable ‘busy
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 2.9 Local IVR Operation 2.9.1 Inquire IP address 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. 2.9.2 Factory Reset Pick up the phone, and then dial *166*000000#. After hearing a voice prompt of ‘setting successfully’, hang up the phone and the gateway is reset to factory defaults. 2.9.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3 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. Step 2：Start a web browser. To use the user interface, you need a PC with Internet Explorer (version 8 or higher), Firefox, or Safari (for Mac).
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.3 State and Statistics 3.3.1 System Information 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. This ID is used for warranty and cloud server authentication.
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.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.3.2 Registration Information 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. 3.3.3 TCP/UDP Statistics Figure 3.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.3.5 CDR Statistics 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. The CDR function can be enabled on this interface. 3.
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.
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. When the gateway works in the route mode, log in the gateway’s web configuration interface via the LAN port.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.5.2 VLAN（Virtual Local Area Network） 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. Different kinds of VLANs have different messages. 802.1Q The IEEE 802.
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， data messages, voice messages and management messages are subject to the network setting, 802.1Q VLAN1 ID and 802.1P Priority of VLAN1. 802.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.5.4 DMZ Host (Route Mode for VGW-420FS/820FS) 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. Figure 3.
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. 3.5.6 Static Route (Route Mode for VGW-420FS/820FS) 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.
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. 2）Usually, SIP server does not participate in media processing. Under SIP network, media always use end-to-end negotiating.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Explanation of SIP parameters: Primary SIP Server Address Primary SIP Server port The IP address or domain name of the primary SIP server is provided by VoIP service provider. The Service port of the primary SIP server is 5060 by default. It is used to avoid excessively frequent registrations. Registration Expires When the time that is set expires, terminals will send register request to the primary SIP server. The time is 1800s by default.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.7 Port Figure 3.
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 SIP Display Name Primary/Secondary SIP User ID Primary/Secondary Authenticate ID Primary/Secondary Authenticate password Offhook Auto-dial Auto-dial Delay Time DND Caller ID Number for CFU Number for CFB Number for CFNRy Primary /Secondary SIP account description.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.8 Advanced 3.8.1 FXS/FXO Parameters FXS parameters include: timeout Call Progress Tone, Timeout for Dialing, Send Polarity Reversal, etc. Figure 3.
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. Default value is 4 seconds. Timeout for answer (Outgoing call) Timeout for making outgoing calls through a phone.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.8.2 Media Parameter 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: Use Random Port If this parameter is enabled, the gateway will choose a port at random as the start port for RTP.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.8.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Figure 3.10-3 SIP Parameter Configuration Interface Explanation of SIP parameters: Indicator) You will be notified when ‘voicemail message waiting indicator’ is enabled. MWI Subscription Expires MWI subscription expiry time; default value is 3600s. Voicemail User ID The user ID for access to voicemail box RFC3407 Support Whether to enable RFC3407 support.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Send BYE when Recv REFER Response (unattended) Send New REGISTER when Recv 423 Response If this parameter is enabled, the third party will send BYE to release session after receiving REFER during a blind transfer. If this parameter is enabled, the value of ‘expires’ header will be automatically updated and REGISTER will be re-sent after receiving of 423 response.
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）Check feature code in Elastix and change it if necessary.
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. After that, user will hear a voice message.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.8.4 Fax Parameter Figure 3.10-4 Configuration Interface for Fax Parameter Explanation of fax parameters: Fax Mode Include “a=X-fax” Attribute There are four fax modes: T.38, T.30 (Pass-through), Modem and Adaptive. Include “a=fax” Attribute If this parameter is enabled, “a=fax” attribute will be carried in SDP. Include “a=X-modem” Attribute If this parameter is enabled, “a=X-modem” attribute will be carried in SDP.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.8.5 Digit Map Figure 3.10-5 Digit Map Digit Map Syntax Supported Digit objects T DTMF Range [] 0-9 Timer A digit, a timer, or one of the symbols of A, B, C, D, #, or *. One or more DTMF symbols enclosed in the [], but only one DTMF symbol can be selected. Range () One or more expressions enclosed the (), but only one can be selected. Separator | Separated expressions or DTMF symbols.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.8.6 Feature Codes Please make reference to 2.7 Description of Feature Codes and the following table.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.8.7 System Parameter 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.
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’. NTP 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; Default port is 123.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.9 Call & Routing 3.9.1 Wildcard Group Figure 3.11-1 Wildcard Group 3.9.2 Port Group On the Port Group interface, user can group several ports together and then set a strategy for port selection of the group.
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 Primary/Secondary Display Name Via:SIP/2.0/UDPpc33.atlanta.com;branch=z9hG4bK776asdhds Max-Forwards: 70 To: Bob From: Alice
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.9.3 IP Trunk 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 The no. of the IP trunk ranging from 0 to 127.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.9.5 IP -> Tel Routing 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; ‘any’ means any IP addresses The prefix of the caller number, which helps match routing exactly. its length is less than or equal to the Caller Prefix caller number.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.9.6 Tel-IP/Tel Routing Figure 3.11-6 Configuration Interface for Tel-IP/Tel Routing Explanation of related parameters: Index The index of this Tel IP/Tel routing, from 0 to 127. Each index cannot be used repeatedly. Routing priority: 0 is the highest priority. Description It is used to identify the routing Calls From Tel IP calls are from a port or a port group The prefix of the caller number, which helps match routing exactly.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.9.7 IP – IP Routing Figure 3.11-7 Configuration Interface for IP->IP Routing Explanation of related parameters: Index The index of this IP IP routing, from 0 to 127. Each index cannot be used repeatedly. Routing priority: 0 is the highest priority. Description It is used to identify the routing Calls From Calls are from IP trunk. The prefix of the caller number, which helps match routing exactly.
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. 3.10.1 IP -> Tel Callee Figure 3.12-1 Add IP -> IP Callee The index of this manipulation, from 0 to 127. Each index cannot be used repeatedly.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.10.2 Tel -> IP/Tel Caller Figure 3.12-2 Add Tel -> IP Caller Configuration parameters are the same as those of ‘IP->Tel Callee’.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.10.3 Tel-IP/Tel Callee Figure 3.12-3 Add Tel-IP Callee Configuration parameters are the same as those of ‘Tel->IP Caller’.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.11 Routing rule examples 3.11.1 Route any calls from any IP to specific port 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.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.11.2 Route any calls from any IP to specified port group Create port group Before we can route calls to a port group, create the port group first as shown below. On the Call & Routing Port Group, click Add to create a new port group. Port 0 to port 2 are assigned to port group 7. Route any calls to the port group On the Call & Routing IP-Tel Routing interface, click Add to create a new routing rule.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.11.3 Route any calls from any port to specific SIP IP trunk 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. Create Tel -> IP routing rule On the Call & Routing Tel-IP Routing interface, click “Add” to create a new Tel IP routing rule.
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.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.12 Maintenance 3.12.1 TR069 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 3.12.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Figure 3.14-2 SNMP Parameters User configuration is only available on SNMP v3.
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.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.12.3 Syslog 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. There are 5 levels of syslog, including NONE, DEBUG, NOTICE, WARNING and ERROR.
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. 3.12.4 Provision 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.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.12.5 Cloud Server User can register the gateway to cloud server, and then the gateway will be managed by cloud server. Figure 3.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.13 Security 3.13.1 WEB ACL 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 3.13.2 Telnet ACL ACL (Access Control List) for Web is used to configure IP addresses (users) that are allowed to access the Telnet page of the gateway. The IP address list can’t be null once ACL is enabled.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.13.3 Passwords 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.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway 3.14 Tools 3.14.1 Firmware upload Firmware upload steps: Step 1. Check the current firmware version on the System Information page Figure 3.16-1 Firmware Version 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 18 is hardware version, xx.xx is version number Step 3.
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 3.14.2 Data Backup The process data backup: 1) Click “Data Backup” 2) Click “Backup” to backup data to PC. Figure 3.16-5 Data Backup 3.14.3 Data Restore The processes of data restore: Click ‘Data Restore’; Browse file, select data file. Click ‘Restore”’and then import successfully; the device will restart automatically. Figure 3.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Figure 3.16-7 Ping Test 3.14.5 Tracert Test 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.
4-/8-/16-/24-/32-Port SIP Internet Telephony Gateway Figure 3.16-8 Tracert Test 3.14.6 Outward Test 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.
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 3.14.7 Network Capture Network capture is a very important diagnostic tool for maintenance. It can be used to capture data packages of the available network ports. Default Setting is PCM capture PCM capture helps to analysis voice stream between analog phone and DSP chipset.
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’; x is serial number of capture and will be added 1 next time.
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: Select DSP only on Network Capture page Click Start to enable DSP capture Dial out through gateway, start talking a short time and then hang up the call.
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. 3.14.8 Factory Reset Click ‘Apply’ to restore the factory settings. Factory Reset 3.14.9 Device Restart After saving all the configurations or changes to the equipment, user can restart the VGW-X20FS SERIES gateway for the changes to take effect.
