Grandstream Networks GXW410X VOIP GATEWAY User Manual

Grandstream Networks, Inc. VOIP GATEWAY

Contents

Users Manual

Grandstream Networks, Inc.Analog IP Gateway GXW410x4 or 8 FXO Ports w/Video Surveillance GXW410x User Manual www.grandstream.comFirmware Version 1.3.1.6 support@grandstream.com
TABLE OF CONTENTSGXW410X USER MANUALWELCOME .................................................................................................................................................5Gateway GXW410x Overview................................................................................................................................5Safety Compliances.................................................................................................................................................5Warranty.................................................................................................................................................................5PACKAGING ..............................................................................................................................................6CONNECTING THE GXW410X .........................................................................................................................................6APPLICATION DESCRIPTION...................................................................................................................8Functional Diagram of IP-PBX & GXW410x........................................................................................................8GXW400x & GXW410x Scenario/Toll- Free Calling Between Locations.............................................................9FXS and FXO Gateway Configuration Example..................................................................................................10FEATURES...............................................................................................................................................10SOFTWARE FEATURES OVERVIEW.....................................................................................................................................10HARDWARE SPECIFICATION.............................................................................................................................................11CONFIGURATION GUIDE........................................................................................................................13CONFIGURATION WITH WEB BROWSER.............................................................................................................................13Accessing the Web Configuration Menu...............................................................................................................13End-user Configuration........................................................................................................................................13ADVANCED USER SETTINGS............................................................................................................................................17Advanced User Configuration .............................................................................................................................17CONFIGURING THE FXO CHANNELS ................................................................................................................................20PROFILES......................................................................................................................................................................26Saving the Configuration Changes.......................................................................................................................28Rebooting from Remote........................................................................................................................................28VIDEO SURVEILLANCE...........................................................................................................................29VIDEO SURVEILLANCE PROCEDURES.................................................................................................................................29FIRMWARE UPGRADE............................................................................................................................31UPGRADE THROUGH HTTP............................................................................................................................................31UPGRADE THROUGH TFTP.............................................................................................................................................31Download TFTP Server........................................................................................................................................32RESTORE FACTORY DEFAULT SETTING.............................................................................................33EXAMPLES OF GXW410X CONFIGURATIONS......................................................................................34GXW410x User Manual www.grandstream.comFirmware Version 1.3.1.6 support@grandstream.com
TABLE OF FIGURESGXW410X USER MANUALFIGURE 1: DIAGRAM OF GXW410X BACK PANEL................................................................................6FIGURE 2: DIAGRAM OF GXW410X DISPLAY PANEL..........................................................................7FIGURE 7: SCREEN-SHOT OF VIDEO SURVEILLANCE* ....................................................................30APPLICATION ONE: GXW CONNECTED WITH AN IP-PBX OR SIP SERVER....................................34APPLICATION TWO: GXW TO EXTEND A TRADITIONAL PBX SCENARIO.......................................34APPLICATION THREE: GXW CONNECTED WITH AN IP-PBX OR SIP SERVER AND VIDEO SURVEILLANCE.......................................................................................................................................35APPLICATION FOUR: USING A GXW FOR PURE IP- IP COMMUNICATION CONFIGURATION.......36INDEX OF TABLESGXW410X USER MANUALTABLE 1: DEFINITIONS OF THE GXW CONNECTORS..........................................................................6TABLE 2: DEFINITIONS OF THE GXW DISPLAY PANEL.......................................................................7TABLE 4: HARDWARE SPECIFICATION OF GXW410X ......................................................................11TABLE 3: GXW410X SOFTWARE FEATURES......................................................................................12TABLE 6: WEB LOG-IN DEFINITIONS (BASIC SETTINGS PAGE).......................................................14TABLE 7: STATUS PAGE DEFINITIONS...............................................................................................15TABLE 8: ADVANCED CONFIGURATION PAGE DEFINITIONS..........................................................17TABLE 9: FXO LINES CONFIGURATION DEFINITIONS.......................................................................20TABLE 10: FXO LINE TEST TAB DEFINITIONS.....................................................................................23TABLE 11: CHANNELS PAGE DEFINITIONS........................................................................................24TABLE 12: CHANNELS PAGE DEFINITIONS........................................................................................26TABLE 13: PROFILE PAGE DEFINITIONS............................................................................................26 GXW410x User Manual www.grandstream.comFirmware Version 1.3.1.6 support@grandstream.com
GUI INTERFACESGXW410X USER MANUALhttp://www.grandstream.com/support/gxw_series/gxw410x/documents/gxw410x_gui.zip1. SCREENSHOT OF ADVANCED SETTINGS CONFIGURATION PAGE 2. SCREENSHOT OF BASIC SETTINGS CONFIGURATION PAGE3. SCREENSHOT OF CHANNELS CONFIGURATION PAGE4. SCREENSHOT OF FXO LINES CONFIGURATION PAGE5. SCREENSHOT OF PROFILE 1 CONFIGURATION PAGE6. SCREENSHOT OF STATUS CONFIGURATION PAGEGXW410x User Manual www.grandstream.comFirmware Version 1.3.1.6 support@grandstream.com
WELCOME Thank you for purchasing the Grandstream GXW410x IP Analog FXO Gateway. The GXW410x is a cost effective, easy to use and easy to configure IP communications solution for any business. The GXW410x gateway supports popular voice codecs and is designed for full SIP compatibility and interoperability with party SIP providers, thus enabling you to fully leverage the benefits of VoIP technology, integrate a traditional phone system into a VoIP network, and efficiently manage communication costs.This manual will help you learn how to operate and manage your GXW FXO Analog IP Gateway and make the best use of its many upgraded features including simple and quick installation, multi-party conferencing, etc. This IP Analog Gateway is very easy to manage and scalable, specifically designed to be an easy to use and affordable VoIP solution for the small – medium business or enterprise. Enable the video surveillance port to give you peace of mind while you are away from your business.GATEWAY GXW410X OVERVIEWThe GXW410x offers an easy to manage, feature-rich IP communications solution for any small business or businesses with virtual and/or branch locations who want to leverage their broadband network and/or add new IP Technology to their current phone system. The Grandstream enterprise analog VoIP Gateway GXW410x series converts SIP/RTP IP calls to traditional PSTN calls and vice versa. There are two models - the GXW4104 and GXW4108, which feature 4 or 8 FXO ports respectively. The installation is the same for either model.SAFETY COMPLIANCESThe GXW410x is compliant with various safety standards including FCC/CE. Its power adapter is compliant with UL standard. Warning: use only the power adapter included in the GXW410x package. Using an alternative power adapter may permanently damage the unit.WARRANTYGrandstream has a reseller agreement with our reseller customer. End-users should contact the company from whom you purchased the product for replacement, repair or refund.If you purchased the product directly from Grandstream, contact your Grandstream Sales and Service Representative for a RMA (Return Materials Authorization) number. Grandstream reserves the right to remedy warranty policy without prior notification.Caution: Changes or modifications to this product not expressly approved by Grandstream, or operation of this product in any way other than as detailed by this User Manual, could void your manufacturer warranty.•This document contains links to Grandstream GUI Interfaces. Please download the GUI examples http://www.grandstream.com/support/gxw_series/gxw410x/documents/gxw410x_gui.zip for your reference.•This document is subject to change without notice. The latest electronic version of this user manual is available on our website: http://www.grandstream.com/support/gxw_series/gxw410x/documents/gxw410x_usermanual_english.pdf
Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print, for any purpose without the express written permission of Grandstream Networks, Inc. is not permitted.PACKAGING Unpack and check all accessories. Equipment included in the package: 1) One GXW410x Unit 2) One universal power adapter3) One Ethernet cableCONNECTING THE GXW410X FIGURE 1: DIAGRAM OF GXW410X BACK PANELTABLE 1: DEFINITIONS OF THE GXW CONNECTORSLAN (or PC) Connect your PC to this port. It will then be assigned an IP address from your Router/DHCP Server. The GXW410x acts as a switch only.WAN Connect to the internal LAN network or Public Internet.VIDEO IN Connection for Analog based Video Surveillance Camera (RCA)RESET Factory Reset button. Press for 7 seconds to reset factory default settings.POWER IN Power adapter connectionOFF/ON Off/On switchFXO1 - FXO8 FXO ports to be connected to physical PSTN lines from a traditional PSTN PBX or PSTN Central Office.
FIGURE 2: DIAGRAM OF GXW410X DISPLAY PANELTABLE 2: DEFINITIONS OF THE GXW DISPLAY PANELPower LED Indicates Power. Remains ON when Power is connected and unit is turned ON.Ready LED Remains ON after boot-up.LAN LED Indicates LAN (or WAN) port activityPC LED Indicates PC (or LAN) port activityVideo LED Remains solid green on boot-up. If Video IN terminal is connected, indicates video activity.LEDs 1 - 8 Indicate status of the respective FXO Ports on the back panelBusy - ON Available - OFFNOTE: All LEDs display green when ON. The Ready light will only be ON when the network interface is ready and the Web User Interface is accessible. During a firmware upgrade or configuration download the following LED pattern will be observed:Power, Ready, Video and WAN LEDs will be ON. The FXO port LED will keep flashing during download and then stay OFF while the new files are written. The entire process may take between 20 to 30 minutes. The firmware upgrade is complete when you can login into the web configuration pages.
APPLICATION DESCRIPTIONA. IP PBX / SIP Server with GXW410xA SIP proxy server such as Asterisk or a SIP registrar server can be deployed with the GXW410x series gateway. In this environment, the SIP server handles SIP registration and call control and the GXW410x processes media conversion between IP and PSTN calls. There are 2 ways to configure GXW410x when using it with a SIP Server: 1. With SIP accounts configured on Channels page. In this case, the GXW410x gateway acts as an endpoint requesting registration from the SIP Server. Under the Channels page you will need to fill in the information like SIP User ID, Password, etc. Now, when calls are made from an IP phone, the call will be routed to the SIP Server which will forward it to one of the SIP accounts on the GXW410x, which will then forward it to the PSTN line. 2. Without SIP accounts. In this case, you simply have to configure the SIP Server to perform forwarding of the SIP INVITE message with the FXO destination number to the gateway’s IP Address. The GXW410x gateway will receive the digits and immediately forward them on the FXO lines to the destination PSTN. Most of the configuration on the Gateway for this case will remain default, except Stage Method needs to be set to 1, and SIP Server IP Address/DNS name has to be filled.FUNCTIONAL DIAGRAM OF IP-PBX & GXW410XFor incoming calls from the PSTN analog endpoints to the GXW410x gateway, the device will auto forward each call to a configured IP extension. The SIP Server can then route the call based on its own configuration or IVR system.PSTNPSTNCloudCloudAnywhere in the worldGXW-410x4 or 8 PortsFXO LinesPSTN Analog endpointsGrandstream IP PhonesIPPBX or SIP ServerIP/LAN IP/WAN
B. FXS Gateway with GXW410x [No SIP Server required]Alternatively, the GXW410x gateway can be used without a SIP Server. It can be used in conjunction with an FXS Gateway (Ex. GXW400x) and calls can still be originated from the IP network and terminated on the PSTN network and vice versa. All you need to make sure is that both gateways are able to locate each other (i.e. they should be on the same LAN or have public IP addresses).GXW400X & GXW410X SCENARIO/TOLL- FREE CALLING BETWEEN LOCATIONSIf you setup resembles the one in this diagram, you need to configure the SIP Server field to be the IP address of the other gateway (i.e. configure the IP address of GXW400x gateway to be the SIP Server for the GXW410x gateway and vice versa). Please be sure you set SIP Registration to No.EXPECTED CALL FLOW: An analog Phone (connected to the GXW400x gateway) picks up and dials the destination PSTN number. The call gets routed to the GXW410x gateway which dials out the digit string onto the FXO Lines, thus reaching the destination PSTN endpoint. On the reverse, incoming calls from the PSTN endpoints connected to the GXW410x gateway will be routed automatically to the analog endpoints connected to the GXW400x gateway.Branch A - Boston, MA4 employeesBranch B – Denver, CO 4 employeesInternet Internet CloudCloudGXW-400xGXW-410xAnalog PhonesPSTN PSTN CloudCloudFX0 Lines
FXS AND FXO GATEWAY CONFIGURATION EXAMPLEGXW400X GATEWAY GXW410X GATEWAYProfile 1SIP Server - Set it to IP Address of GXW410xSIP Registration - NoOutgoing Call without Registration - YesNAT traversal – NoAdvanced Settings STUN Server - BlankUse Random Port - NoAdvanced SettingsSTUN Server - BlankFXO linesWait for Dial Tone - Y or N (whichever works for your PSTN Service Provider)Stage Method - 1Unconditional Call Forward to VOIP: ch1-8:444; @ch1-8:p1; ch1-8:5060++;Channels1-8 5060 Profile 1Local SIP Listen port (For VOIP to PSTN calls) - 5060++Profile 1SIP Server - Set it to IP Address of GXW400xSIP Registration - NoNAT traversal - NoFor more information regarding this setup, email Grandstream technical support at support@grandstream.com or visit our User and Developer forum at forums.grandstream.com
FEATURESThe GXW410x is a next generation IP voice and video gateway that features full interoperability with leading IP-PBXs, Soft-switches and SIP platforms. The GXW410x series gateways offer superb voice and video quality, traditional telephony functionality, simple configuration, feature-rich functionality and an additional video port that enables the gateway to act as a video surveillance gateway. SOFTWARE FEATURES OVERVIEW•4 and 8 FXO port media gateways •Video surveillance port•External power supply•Two RJ-45 ports (switched or routed)•TFTP and HTTP firmware upgrade support•Multiple SIP accounts, multiple SIP profiles (choice of 3 profiles per account)•Supports Audio Codecs: G711U/A, G723, G729A/B and GSM•Supports Video Codecs: H.264•G.168 – echo cancellation•Flexible DTMF transmission: In Audio, RFC2833, SIP Info or any combination of the 3 •Selectable, multiple LBR coders per channel •T.38 compliantHARDWARE SPECIFICATIONTABLE 4: HARDWARE SPECIFICATION OF GXW410X LAN interface 2xRJ45 10/100MbpsLED 8 LEDs (GREEN)Universal SwitchingPower AdapterInput: 100-240V AC, 50/60Hz, 0.5A MaxOutput: 12V DC, 1.25AUL certified Dimension 225mm (L) x 172mm (W) x 42mm (H)Weight 0.29 lbs (3.5 oz)Temperature 32~104°F 0~40°CHumidity 10% - 90% (non-condensing)Compliance FCC, CE
TABLE 3: GXW410X SOFTWARE FEATURESGXW410x FX0 Analog Gateway SeriesIP settings GXW4104: 4 ports; 4 SIP accounts w/ choice of 3 SIP Server profilesGXW4108: 8 ports; 8 SIP accounts w/ choice of 3 SIP Server profilesRound-robin port scheduling to ensure available lines to access PSTN networksTelephone Interface FXO, RJ11Network Interface Two (2) 10/100 Mbps, RJ45LED Indicators Power, Video, and Line LEDsOn/Off Switch YesVoice over Packet CapabilitiesG.168 compliant Echo Cancellation, Dynamic Jitter Buffer, Modem detection & auto-switch to G.711Voice Compression G.711U, G711A, G.723, G.729A/B, GSMVideo Surveillance Real-time H.264 base CIF resolutionDHCP Server/Client Switch Mode and PPPoEFax over IP T.38 compliant Group 3 Fax Relay up to 14.4kpbs and auto-switch to G.711 for Fax Pass-throughQoS Diffserve, TOS, 802.1 P/Q VLAN taggingIP Transport RTP/RTCP and RTSP PSTN Signaling FXO Loop start, Current Disconnect.DTMF Method Flexible DTMF transmission method, User interface of In-audio, RFC2833, and SIP InfoIP Signaling SIP (RFC 3261)Provisioning TFTP and HTTPMedia SRTPControl TLS and SIPS (pending)Management Syslog support, HTTPS and telnet (pending), remote management using Web browserShort and long haul REN3: Up to150 ft on 24 AWG lineCaller ID Bellcore Type 1 & 2, ETSI, BT, NTT, and DTMF-based CIDPolarity Reversal / Wink Yes (Detection only). The PSTN lines will need to be subscribed to PR service from the Service Provider.EMC GXW410x: EN55022 Class B, CFR Part 15 Class B, EN55024; GXW4104: FCC, CE (in addition)Safety GXW410x: EN60950-1 GXW4108: UL60950-1 (in addition)
CONFIGURATION GUIDECONFIGURATION WITH WEB BROWSERThe GXW410x gateways have an embedded Web server that will allow a user to configure the IP phone through any web browser. Examples of GUI interfaces can be downloaded @ http://www.grandstream.com/support/gxw_series/gxw410x/documents/gxw410x_gui.zipACCESSING THE WEB CONFIGURATION MENU1. Navigate your browser to: http://www.grandstream.com/DOWNLOAD/IPQuery/IPQuery.zip2. Run the Grandstream IPQuery tool, that you just downloaded.3. Click on button in order to begin device detection4. The detected devices will appear in the Output fieldEND-USER CONFIGURATIONWhen the Web Configuration Menu is accessed, the GXW410x will respond with a login screen. There are two default passwords for the login page: User Level: Password: Web pages allowed:End-user Level 123 Only Status and Basic SettingsAdministrator Level admin All pages can be browsed.
After login, the following page is the Basic Configuration page, explained in detail in Table 6: Web Log-in Definition. TABLE 6: WEB LOG-IN DEFINITIONS (BASIC SETTINGS PAGE)Web Access Select HTTP or secure HTTPS protocol for Web AccessWeb Port By default, HTTP uses port 80 and HTTPS uses port 443. This field is for customizable web port.End-user Password This contains the password to access the End-user Web Configuration Menu (Status and Basic Settings). This field is case sensitive with a maximum length of 25 characters.IP Address There are two modes to operate the GXW410x:DHCP mode: all the field values for the Static IP mode are not used (even though they are still saved in the Flash memory.) The GXW410x acquires its IP address from the first DHCP server it discovers from the LAN it is connected. Using the PPPoE feature: set the PPPoE account settings. The GXW410x will establish a PPPoE session if any of the PPPoE fields is set.Static IP mode: configure the IP address, Subnet Mask, Default Router IP address, DNS Server 1 (primary), DNS Server 2 (secondary) fields. DHCP hostname This option specifies the name of the client. This field is optional but may be required by some Internet Service Providers. Default is blank.DHCP domain This option specifies the domain name that client should use when resolving hostnames via the Domain Name System. Default is blank.DHCP vendor class ID Used by clients and servers to exchange vendor-specific information. Default is Grandstream GXW410x.PPPoE account ID PPPoE user name. Necessary if ISP requires you to use a PPPoE (Point to Point Protocol over Ethernet) connection. PPPoE password PPPoE account password.PPPoE Service Name This field is optional. If your ISP uses a service name for the PPPoE connection, enter the service name here. Default is blank.Preferred DNS server This field will let the user enter a preferred DNS server to be used instead of the one acquired by the service provider.Time Zone Controls how the date/time is displayed according to the specified time zone.Allow DHCP Option 2 to override Time Zone SettingsDefault is No. If set to Yes, time zone settings will originate from the DHCP server.
Daylight Savings Time This parameter controls whether the displayed time will be daylight savings time or not. If set to “Yes” and the Optional Rule is empty, then the displayed time will be 1 hour ahead of normal time. The “Automatic Daylight Saving Time Rule” shall have the following syntax: start-time;end-time;savingBoth start-time and end-time have the same syntax:Month ,day ,weekday ,hour ,minutemonth: 1,2,3,..,12 (for Jan, Feb, .., Dec)day: [+|-]1,2,3,..,31weekday: 1, 2, 3, .., 7 (for Mon, Tue, .., Sun), or 0 which means the daylight saving rule is not based on week days but based on the day of the month.hour: hour (0-23),minute: minute (0-59)If “weekday” is 0, it means the date to start or end daylight saving is at exactly the given date. In that case, the “day” value must not be negative. If “weekday” is not zero and “day” is positive, then the daylight saving starts on the first “day”th iteration of the weekday (1st Sunday, 3rd Tuesday etc). If “weekday” us not zero and “day” is negative, then the daylight saving starts on the last “day”th iteration of the weekday (last Sunday, 3rd last Tuesday etc). The saving is in the unit of minutes. The saving time may also be preceded by a negative (-) sign if subtraction is desired instead of addition. The default value for “Automatic Daylight Saving Time Rule” shall be set to“03,11,0,02,00;11,04,0,02,00;60” which is the rule for US.Examples:US where daylight saving time is applicable: 03,11,0,02,00;11,4,0,02,00;60This means the daylight saving time starts from 11th March at 2AM and ends November 4th at 2AM. The saving is 60 minutes (1hour).You may also access the Device Status page which provides details of the GXW product. The Device Status page terms are defined in Table 7: Status Page Definitions. TABLE 7: STATUS PAGE DEFINITIONSHardware Revision Hardware version number: Main Board, Interface BoardMAC Address The device ID in HEX format. This is a very important ID for ISP troubleshooting.IP Address This field shows WAN IP address of GXW410xProduct Model This field contains the product model info (GXW4104 or GXW4108)Software Version Program: This is the main software release. Boot and Loader are not changed often.System Up Time This field shows system up time since the last reboot.Registered This field indicates whether the different SIP Accounts configured under Channels page are successfully registered to the SIP server(s).
FXO Line Connected This field will give the status of each physical FXO Line connected to the Gateway. It will update the status regularly.Yes - Connected and IdleBusy - Connected and BusyNo - Not connectedAdditionally it will also provide real time Caller ID information of Incoming as well as Outgoing calls.PPPoE Link Up This field shows whether the PPPoE connection is running if connected to DSL modem.Detected NAT Type This field shows what kind NAT the GXW410x is connected to via its WAN port. It is based on STUN protocol.
ADVANCED USER SETTINGSADVANCED USER CONFIGURATION The end-user needs to login to the advanced user configuration page the same way as for the basic configuration page. Advanced User configuration includes the end-user configuration and advanced configurations including: SIP configuration, Codec selection, NAT Traversal Setting and other miscellaneous configuration. TABLE 8: ADVANCED CONFIGURATION PAGE DEFINITIONSAdmin PasswordAdministrator password. Only the administrator can configure the “Advanced Settings” page. Password field is purposely left blank for security reasons. The maximum password length is 25 characters.G723 Rate G723 encoding rate (6.3kbps or 5.3kbps)Voice Frames per TxThis field contains the number of voice frames to be transmitted in a single packet. When setting this value, the user should be aware of the requested packet time (used in SDP message) as a result of configuring this parameter. This parameter is associated with the first vocoder in the above vocoder Preference List or the actual used payload type negotiated between the 2 conversation parties at run time. For example if the first vocoder is configured as G723 and the “Voice Frames per TX” is set to be 2, then the “ptime” value in the SDP message of an INVITE request will be 60ms because each G723 voice frame contains 30ms of audio. Similarly, if this field is set to be 2 and if the first vocoder chosen is G729 or G711 or G726, then the “ptime” value in the SDP message of an INVITE request will be 20ms.If the configured voice frames per TX exceeds the maximum allowed value, the gateway will use and save the maximum allowed value for the corresponding first vocoder choice. The maximum value for PCM is 10(x10ms) frames; for G726, it is 20 (x10ms) frames; for G723, it is 32 (x30ms) frames; for G729/G728, 64 (x10ms) and 64 (x2.5ms) frames respectively.Layer 3 QoS This field defines the layer 3 QoS parameter which can be the value used for IP Precedence or Diff-Serv or MPLS. Default value is 48. Its range lies from 0 to 63.Layer 2 QoS This contains the value used for layer 2 VLAN tag. 802.1q / VLAN tag: Default value is 0. Range lies from 0 to 4095.802.1p Priority value: Default value is 0. Range lies from 0 to 7.*** The above 2 settings need to be supported on the network and then configured accordingly on the GXW410x. Incorrect configuration will cause blocked access, which will result in Factory Reset as the only option to renew access.Local RTP port This parameter defines the local RTP-RTCP port pair the GXW410x will listen and transmit. It is the base RTP port for channel 0. When configured, channel 0 will use this port _value for RTP and the port_value+1 for its RTCP; channel 1 will use port_value+2 for RTP and port_value+3 for its RTCP and so on. The default value is 5004.Use Random PortThis parameter, when set to Yes, will force random generation of both the local SIP and RTP ports. This is usually necessary when multiple GXW410xs are behind the same NAT. Keep-alive intervalThis parameter specifies how often the GXW410x sends a blank UDP packet to the SIP server in order to keep the “hole” on the NAT open. Default is 20 seconds.Use NAT IP NAT IP address used in SIP/SDP message. Default is blank.
STUN Server IP address or Domain name of the STUN (Simple Traversal of UDP through NATs) server. Firmware Upgrade & ProvisioningThis radio button will enable GXW410x to download firmware or configuration file through either TFTP or HTTP.Via TFTP Server If selected, the GXW410x will attempt to retrieve new configuration file or new code image from the specified TFTP server at boot time. It will make up to 5 attempts before timeout and then it will start the boot process using the existing code image in the Flash memory. If a TFTP server is configured and a new code image is retrieved, the new downloaded image will be verified and then saved into the Flash memory.Note: Please do NOT interrupt the TFTP upgrade process (especially the power supply) as this will damage the device. Depending on the network environment this process can take up to 25 or 30 minutes.Via HTTP Server The URL for the HTTP server used for firmware upgrade and configuration via HTTP. For example, http://provisioning.mycompany.com:6688/Grandstream/Here “:6688” is the specific TCP port that the HTTP server is listening at, it can be omitted if using default port 80.Note: If Auto Upgrade is set to No, GXW410x will only do HTTP download once at boot up.Firmware Server PathIP address or domain name of firmware server.Config Server PathIP address or domain name of configuration server.Firmware File PrefixDefault is blank. If configured, GXW400X will request firmware file with the prefix. This setting is useful for ITSPs. End-user should keep it blank. Firmware File PostfixDefault is blank. End-user should keep it blank.Config File PrefixDefault is blank. End-user should keep it blank.Config File PostfixDefault is blank. End-user should keep it blank.Allow DHCP Option 66 to override serverDefault value is No. If set to Yes, configuration file will originate from the DHCP server.Automatic UpgradeChoose Yes to enable automatic upgrade and provisioning. In “Check for new firmware every” field, enter the number of minutes to enable GXW410x to check the server for firmware upgrade or configuration. When set to No, GXW410x will only do upgrade once at boot up. Other options are: “ Always check for New Firmware.” “ Check New Firmware only when F/W pre/suffix changes”“ Always skip the Firmware check”Authenticate Conf FileIf set to Yes, configuration file is authenticated before acceptance. This protects the configuration from an unauthorized change.
DTMF Payload TypeThis parameter sets the payload type for DTMF using RFC2833. Default is 101.Syslog Server The IP address or URL of System log server. This feature is especially useful for ITSP (Internet Telephone Service Provider)Syslog Level Select the ATA to report the log level. Default is NONE. The level is one of DEBUG, INFO, WARNING or ERROR. Syslog messages are sent based on the following events:1. product model/version on boot up (INFO level)2. NAT related info (INFO level)3. sent or received SIP message (DEBUG level)4. SIP message summary (INFO level)5. inbound and outbound calls (INFO level)6. registration status change (INFO level)7. negotiated codec (INFO level)8. Ethernet link up (INFO level)9. SLIC chip exception (WARNING and ERROR levels)10. memory exception (ERROR level)The Syslog uses USER facility. In addition to standard Syslog payload, it contains the following components: GS_LOG: [device MAC address][error code] error messageExample: May 19 02:40:38 192.168.1.14 GS_LOG: [00:0b:82:00:a1:be][000] Ethernet link is upNTP server URI or IP address of the NTP (Network Time Protocol) server, which will be used by the phone to synchronize the date and time. Allow DHCP Option 42 to override an NTP serverDefault value is No. If set to Yes, the NTP server will originate from the DHCP server.Enable Video SurveillanceWhen set to Yes, GXW410x will start converting video feed received from analog camera to IP packets. In order to view this video feed, please follow instructions given under Video Surveillance chapter on page 23.RTSP Port By default it is 554.
CONFIGURING THE FXO CHANNELS Configuring the FXO channels on the GXW410x gateway is an easy process. Follow the GUI interfaces. The Device Status page terms are defined in Table 9: FXO Lines Configuration Definitions. An example of the Channel Dialing Configuration is shown in Figure 6. Please note the default is always configured. The user has the option to change the default settings as described in the Table 9.TABLE 9: FXO LINES CONFIGURATION DEFINITIONSSyntax for Channel Specific Settings:Default: ch1-8:X; {all channels 1 to 8 are set to use value X}Additional Examples: ch1,3-6:10;ch2,7-8:12 {channels 1,3,4,5 and 6 are set to use value 10 while channels 2,7 and 8 are set to use value 12.Enable Current DisconnectWhen set to Y, Current Disconnect is enabled. Certain PSTN COs require this to be enabled, in order to realize disconnect signal from PSTN side. Default is Y.If enabled use threshold: Default is 100ms. Range is 40ms to 800ms.Certain PSTN service providers have a threshold time within which the line stabilizes after off-hook. It is entirely dependent on provider, however if you experience PSTN line detection issues, please modify this setting appropriately in 100ms increments.If you are not sure if this option should be enabled please refer to Table 10 (FXO Lines Test Tab Definition). This tool will run an automated test to determine the proper PSTN configuration that the gateway should have to work with your Service Provider or analog PBX.Enable Tone DisconnectDefault is No. If PSTN provider uses call progress tones then it should be set to Yes in order to realize disconnect tone. Please configure accurate Call Progress Tones on Channels web page based on PSTN provider (or traditional PBX) settings.If you are not sure if this option should be enabled or what Call Progress Tones are required please refer to Table 10 (FXO Lines Test Tab Definition). This tool will run an automated test to determine the proper PSTN configuration that the gateway should have to work with your Service Provider or analog PBX.Enable Polarity ReversalDefault is No. This should be set to Yes only if the FXO lines are subscribed to PR service from PSTN Service provider. It is merely a PR detect feature.***Note: If there is no PR service from provider on the FXO line, and this setting is configured to Yes, calls will not be successful.Silence Timeout Terminate call after long silence detected. Default is 60 seconds, max 65536.AC Termination ImpedanceSelects the impedance of the analog line connected to the FXO port on the GXW410x. Here is some basic information which may be helpful for initial configuration:600 Ohm – North America;270 Ohm + (750 Ohm || 150 nF) -- Most of Europe 220 Ohm + (820 Ohm || 120 nF) – Australia, New Zealand220 Ohm + (820 Ohm || 115 nF) – Austria, Bulgaria, Germany, Slovakia, South Africa370 Ohm + (620 Ohm || 310 nF) – UK., IndiaIf this parameter is not configured properly you may experience echo or noise in the line. Please refer to Table 10 (FXO Lines Test Tab Definition). This tool will run an automated test to determine the correct impedance value to match your lines.
Wait for Dial-tone Default is No. When set to Yes, the gateway will recognize dial-tone from the Central Office (CO) before it completes a call. If you can’t make an outbound call, set this is Yes. if this is set to Yes make sure you have configured the dial tone settings correctly in the Channels tab and that there is not any major noise interference in the line.Stage Method Syntax - ch1-8:1; {all channels 1 to 8 are set to value 1 or 2}Stage method can be set to either 1 or 2.Set this parameter to 1 if you need to make a direct PSTN call from a VOIP endpoint. When you set it to 2, you will first dial one of the VOIP channel accounts from the VOIP endpoint, this will result in getting a PSTN line dial-tone to then dial out the destination PSTN number.Most implementations require this setting to be configured to 1. Min. Delay before Dial PSTNDefault is 500ms. This needs to be equal to or greater than the Current Disconnect threshold setting. Once the threshold is reached the gateway can dial out. This parameter should only be used if there are PSTN line detection issues.Unconditional Call Forward to VOIP:This is an extremely important setting to make sure incoming PSTN calls are picked up and forwarded to the correct VOIP destination.User ID - This parameter allows users to configure a User ID or extension number to be automatically dialed upon FXO line off-hook.SIP Server - You also need to specify the Profile of the user id configured above (p1 stands for Profile 1, p2 stands for Profile 2 and so on).SIP Destination Port - Along with the user-id and Profile, you also have the option to choose the destination port where you would like to send the call. By default it should be set to ch1-x:5060; (x can be 4 or 8 depending on number of ports).We can also specify a different destination for each port. For example under User ID we can type in: ch1:104;ch2:227;ch3-5:501;ch6,7:856.Under Sip Server we can type in: ch1:p1;ch2-4:p2;ch5:p3Under Sip Destination Port we can type in: ch1-2:5060;ch2:7080;ch3-8:5066++Number of Rings Before Pickup Default is 4. This is the number of rings the gateway will wait to send the call to the VOIP side in case the Caller ID has yet to be detected. If there's CID information the call will be sent right away. If your lines don't have the CID service set this to 1.
Caller ID Scheme The GXW410x supports 5 different types of schemes: 1. Bellcore (US standard)2. ETSI-FSK during ringing3. ETSI-FSK prior to ringing with DTAS4. ETSI-FSK prior to ringing with LR5. ETSI-FSK prior to ringing with PR6. ETSI-DTMF during ringing7. ETSI-DTMF prior to ringing with DTAS8. ETSI-DTMF prior to ringing with PR9. ETSI-DTMF prior to ringing with PR10. SIN 227 - BT11. NTT (Japanese standard)Please check with your PSTN service provider (or traditional PBX specs) for which caller ID scheme they/it support. If you are not sure about which to use please refer to Table 10 (FXO Lines Test Tab Definition). This tool will run an automated test to determine the proper Caller ID Scheme so the gateway can properly detect the Caller ID.Similarly to the cases explained above we can specify a caller ID scheme for each channel independently.Caller ID Transport typeDefault is “relay via From header”. You may also select : “relay via P_Asserted_Identity header”“Disable” : Caller ID feature will be disabled.“Send anonymous” : All calls forwarded to VOIP end will be sent as anonymous.T.38 Setting This setting allows you to make several options related to facsimile.You can select the method: T.38 or Pass through (G711)You can select the fax transmission rates (2400/4800/7200/9600/12000/14400bps)You can enable or disable ECM (Error Checking Mode)
TABLE 10: FXO LINE TEST TAB DEFINITIONSNote: The user can only test the parameters for only one of the PSTN lines at the same time. In all cases please enter the telephone numbers as if the lines were to dial each other locally.For the AC Impedance Test we only need to select the line to be tested by clicking on the AC impedance box corresponding to that line, telephone numbers are optional. Remember that the AC impedance test is usually used to reduce the echo that might be present in the line.For the CPT test (call progress tones) we will test current disconnect as well. You will need 2 telephone numbers to perform the test. You can only perform the test on one line (row) at the same time and it will be the one that has the box checked for testing. This tested line will use another line connected to the gateway to perform the test by calling into it, this is why you will have to enter the telephone number for a second line to help with the test.For CID detection you will need 2 telephone numbers to perform the test. You can only perform the test on one line (row) at the same time and it will be the one that has the box checked for testing. This tested line will use another line connected to the gateway to perform the test by calling into it, this is why you will have to enter the telephone number for a second line to help with the test.To perform the test, please select the line you want to test and the desired test to be performed. Enter the information for this line as well as a second line if necessary. Then click on the update button and then reboot. Log back in and now you should see the information for the line selected as well as the check box already marked already there. Go ahead and start the test now, please wait a few minutes until the test is done.Notes: It is not required to enter a telephone number when testing for impedance, as the system does not place any actual calls for the test.If you log into the Web Interface while the test is running will not interrupt the process.Line # Enter the telephone (PSTN) number that corresponds to this line. Enter it as if you were going to dial it locallyAC Impedance Select this box if you want to test for impedance on the line that is on the same row as the checked box. Remember that you can only check one item at the same time. CPT Detection Select this box if you want to test for call progress tones and current disconnect threshold on the line that is on the same row as the checked box. Remember that you can only check one item at the same time.External Number Enter an external telephone (PSTN) number to be used as an auxiliary number for the test. This is used only if we do not have at least 2 PSTN lines connected to the gateway. This is only used for CPT and current disconnect threshold testing. In order to use this function you will have to monitor the Syslog output. This is only reserved for very advanced usersExternal Call Timeout This is the time the GXW will wait for the external telephone number to pick up during the test.
Apply test results automaticallyDefault is No. If selected on Yes, then all the results from the test will be applied automatically. If you select No you will have to monitor the Syslog output. This is only reserved for very advanced users.Apply test results to all portsDefault is No. If selected on Yes, then all the test results will be applied to all ports on the gateway. If all the lines belong to the same service provider or PBX it will make sense to apply the results to all ports.Error Timeout This is the time the gateway will wait to exit the test mode, when something unexpected or an error has occurred.TABLE 11: CHANNELS PAGE DEFINITIONSNote – The channels here are basically SIP endpoints that will act as clients registering to the SIP Server configured under the appropriate Profile page.Channels It should be set same as the channel number (i.e 1, 2..4 or 8 depending on number of FXO ports). It is NOT the same as SIP Account ID.SIP User ID This is the SIP account information. Enter the SIP User ID part of the account. Authentication ID SIP service subscriber’s Authenticate ID used for authentication. It can be identical to, or different from SIP User ID.Authentication Password SIP account password needs to be entered here. Note: After entering the password, it will show up as blank but the password still remains active.Profile ID Select the corresponding Profile ID (1/2/3). Profiles are SIP Server configurations.Call Progress Tones Using these settings, user can configure tone frequencies according to user preference. By default, the tones are set to North American frequencies. Frequencies should be configured with known values to avoid uncomfortable high pitch sounds. ON is the period of ringing (ON time in ms) while OFF is the period of silence. In order to set a continuous ring, OFF should be zero. Otherwise it will ring ON ms and a pause of OFF ms and then repeat the pattern.•“Dial tone”•“Ringback tone”•“Busy/Re-order tone”•“Confirmation tone”Please refer the document below to determine your local call progress tones (http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf) or run the FXO Line Test (Table 10).Channel Voice Settings Channel voice settings mentioned below.Tx to PSTN Audio Gain (dB) Allows user to set a value in dB for transmission to PSTN Audio Gain. Default is 1. Range is from -12 to 12dB.Rx from PSTN Audio Gain (dB)Allows user to set a value in dB for receive from PSTN Audio Gain. Default is 0. Range is from -12 to 12dB.Silence Suppression This controls the silence suppression/VAD feature of G723 and G729. If set to “Yes”, when a silence is detected, small quantity of VAD packets (instead of audio packets) will be sent during the period of no talking. If set to “No”, this feature is disabled.
Echo Cancellation When set to Y, Echo cancellation is enabled.Channel specific Setting Channel specific settings mentioned below.DTMF Method This parameter specifies the mechanism to transmit DTMF digits. There are7 modes supported: in audio which means DTMF is combined in audio signal (not very reliable with low bit-rate codec), via RTP (RFC2833), or via SIP INFO. Multiple DTMF transmission schemes can be selected.•1 – in-audio•2 – RFC2833•3 – in-audio and RFC2833•4 – SIP Info•5 – in-audio and RFC2833•6 – SIP Info and RFC28331. 7 – in-audio, RFC2833, and SIP InfoNo Key Entry Timeout Default is 4 seconds. Local SIP Listen Port Default is ch1-8:5060++;. The ++ indicates increments by 2, so port 1 is set at 5060, port at 5062 and so on. This setting can be used with Round Robin and/or Flexible setting below to configure different ports to be placed under different Round Robin groups.SRTP Mode Default is disabled for all ports. The user can select to either enable it but not force it or force it on an individual port basis. When used the communication will be sent using Secure RTP.Round Robin and/or FlexibleDefault is rr:1-8;The syntax is pretty straight-forward here. The rr stands for Round Robin and the numbers stand for the ports that belong to that round robin group.For example: rr:1-8; -> Round robin within the first 8 ports i.e. outgoing calls will be forwarded to the next available port within the group of ports 1 to 8.rr:1,3-6,8;rr:2,7; -> Round robin within port 1,3,4,5,6 and 8; Second round robin group within ports 2 and 7 i.e. outgoing calls to ports 1,3,4,5,6 and 8 will be forwarded to the next available port within this group ONLY. Outgoing calls to port 2 and 7 will be forwarded to the next available port between ports 2 and 7 ONLY. ** In order to terminate a call on FXO port 2 or 7 you will need to change its Local SIP Listen port accordingly.Prefix to specify Port (1 stage dialing method)Default is 99.Syntax to USE this feature: prefix# (that is 99) + ch# (could be anything from 1 to 8) + dialing# will result in this call forwarded to FXO port (ch#) immediately.
TABLE 12: CHANNELS PAGE DEFINITIONSDial Plan The Dial Plan feature implemented is applicable for VOIP to PSTN calls only. You may configure a dial plan based on the following grammar:1. Accept Digits: 1,2,3,4,5,6,7,8,9,0,*,#,A,a,B,b,C,c,D,d2. Grammar: x - any digit from 0-9; xx+ - at least 2 digit number; xx. - at least 2 digit number; ^ - exclude; [3-5] - any digit of 3, 4, or 5; [147] - any digit 1, 4, or 7; <2=011> - replace digit 2 with 011 when dialing WARNING - illegal input will fall back to default Example 1: {[369]11 | 1617xxxxxxx} - Allow 311, 611, 911, and any 10 digit numbers of leading digits 1617. Example 2: {^1900x+ | <=1617>xxxxxxx} - Block any number of leading digits 1900 and add prefix 1617 for any dialed 7 digit numbers. Example 3: {1xxx[2-9]xxxxxx | <2=011>x+} - Allow any length of number with leading digit 2 and 10 digit-numbers of leading digit 1 and leading exchange number between 2 and 9; If leading digit is 2, replace leading digit 2 with 011 before dialing Default: PSTN Outgoing - {x+} Note: If you do not plan to use this feature set to default {x+}Hook flash Duration (X10ms)Default 600ms. This value can accept any value in the 100-2000ms range.Use DTMF Parameter from RFC2833 or SIP InfoDefault Yes, No means to use DTMF parameter settings according to DTMF Digit Length, DTMF Digit Volume and DTMF Dial Pause.DTMF Digit Length Default value is 100ms. Please not that the value will be multiplied by 10msDTMF Digit Volume Default value is -11dB.DTMF Dial Pause Default value is 100ms. Please note that the value will be multiplied by 10ms.PROFILESProfiles are basically IP-PBX / SIP Server configuration templates. If you have more than one IP-PBX system or SIP Server that you would like to use with the GXW410x, then you can configure Profile 2 or 3. Note: Make sure you select the correct profile for each channel under Channels PAGETABLE 13: PROFILE PAGE DEFINITIONSActivate Profile When set to Yes the SIP Profile is activated. Profile Name A name to identify a Profile.
SIP Server SIP Server’s IP address or Domain name provided by VoIP service provider.Outbound Proxy IP address or Domain name of Outbound Proxy, or Media Gateway, or Session Border Controller. Used by GXW410x for firewall or NAT penetration in different network environments. If symmetric NAT is detected, STUN will not work and ONLY outbound proxy can correct the problem.Use DNS SRV: Default is No. If set to Yes the client will use DNS SRV to look up server.User ID is Phone NumberIf the GXW410x has an assigned PSTN telephone number, this field should be set to “Yes”. Otherwise, set it to “No”. If “Yes” is set, a “user=phone” parameter will be attached to the “From” header in SIP request.SIP Registration This parameter controls whether the GXW410x needs to send REGISTER messages to the SIP Server. The default setting is “Yes”. Unregister on RebootDefault is No. If set to yes, the SIP user’s registration information will be cleared on reboot.Register Expiration This parameter allows the user to specify the time frequency (in minutes) for the GXW410x to refresh its registration with the specified registrar. The default interval is 60 minutes (or 1 hour). The maximum interval is 65535 minutes (about 45 days).SIP Registration Failure Retry Wait TimeThis parameter is most used by Service Providers. It prevents message REGISTER overload of SIP Server in case of downtime due to maintenance or power failure. By increasing interval length, common message load is decreased. Interval range is 1 – 3600 seconds.NAT Traversal This parameter defines whether the GXW410x NAT traversal mechanism will be activated or not. If activated (by choosing “Yes”) and a STUN server is also specified, then the GXW410x will behave according to the STUN client specification. Under this mode, the embedded STUN client inside the GXW410x will attempt to detect if and what type of firewall/NAT it is sitting behind through communication with the specified STUN server. If the detected NAT is a Full Cone, Restricted Cone, or a Port-Restricted Cone, the GXW410x will attempt to use its mapped public IP address and port in all of its SIP and SDP messages. If the NAT Traversal field is set to “Yes” with no specified STUN server, the GXW410x will periodically (every 20 seconds or so) send a blank UDP packet (with no payload data) to the SIP server to keep the “hole” on the NAT open.Proxy-Require SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.Early Dial Default is No. Use only if proxy supports 484 response.Session Expiration Grandstream implemented SIP Session Timer. The session timer extension enables SIP sessions to be periodically “refreshed” via a SIP request (UPDATE, or re-INVITE. Once the session interval expires, if there is no refresh via a UPDATE or re-INVITE message, the session will be terminated. Session Expiration is the time (in seconds) at which the session is considered timed out, if no successful session refresh transaction occurs beforehand. The default value is 180 seconds.Min-SE The minimum session expiration (in seconds). The default value is 90 seconds. Caller Request TimerIf selecting “Yes” the phone will use session timer when it makes outbound calls if remote party supports session timer.Callee Request TimerIf selecting “Yes” the phone will use session timer when it receives inbound calls with session timer request.
Force Timer If selecting “Yes” the phone will use session timer even if the remote party does not support this feature. Selecting “No” will allow the phone to enable session timer only when the remote party support this feature. To turn off Session Timer, select “No” for Caller Request Timer, Callee Request Timer, and Force Timer.UAC Specify RefresherAs a Caller, select UAC to use the phone as the refresher, or UAS to use the Callee or proxy server as the refresher. UAS Specify RefresherAs a Callee, select UAC to use caller or proxy server as the refresher, or UAS to use the phone as the refresher.Force INVITE Session Timer can be refreshed using INVITE method or UPDATE method. Select “Yes” to use INVITE method to refresh the session timer. Enable 100rel The use of the PRACK (Provisional Acknowledgement) method enables reliability to be offered to SIP provisional responses (1xx series). This is very important if PSTN inter-networking is to be supported. A user’s request to use reliable provisional responses is invoked by the 100rel tag which is appended to the value of the required header of initial signalling messages.Refer-to uses Target ContactDefault is NO. If set to YES, then for Attended Transfer, the “Refer-To” header uses the transferred target’s Contact header information. INVITE Ring-no-answer TimeoutIn case incoming call has arrived from PSTN to VoIP and INVITE message was generated by GXW device, the call will be disconnected after pre-configured timeout if not answered by VoIP extension.Preferred Vocoder The GXW410x supports up to 5 different Vocoder types including G.711 A-/U-law, GSM, G.723.1, G.729A/B. The user can configure Vocoders in a preference list that will be included with the same preference order in SDP message. The first Vocoder in this list can be entered by choosing the appropriate option in “Choice 1”. Similarly, the last Vocoder in this list can be entered by choosing the appropriate option in “Choice 8”. Special Feature Default is Standard. Choose the selection to meet some special requirements from Soft Switch vendors like Nortel, Broadsoft, etc. SAVING THE CONFIGURATION CHANGESWhen changes are made, press the “Update” button in the Configuration Menu. The GXW410x will display the following screen to confirm that the changes have been saved. To activate changes, reboot or power cycle the GXW410x after all changes are made. REBOOTING FROM REMOTEThe administrator can remotely reboot the unit by pressing the “Reboot” button at the bottom of the configuration menu. The following screen will indicate that rebooting is underway.The user can re-login to the unit after waiting for about 30 seconds.
VIDEO SURVEILLANCEThe GXW410x gateway can be used with an Analog Surveillance CCD Camera to perform video surveillance function. This application should be used in a LAN environment or when both sides have public IP address.VIDEO SURVEILLANCE PROCEDURESGateway side:1. In the ADVANCED SETTING page, find the following field and change from default setting NO to YES, reboot the device. 2. Connect an analog surveillance camera to the VIDEOIN connection at the back panel of the unit.PC side (Monitor Device):1. Download VLC from http://www.videolan.org/vlc/. This is the only player so far that supports RFC 3984. 2. Launch VLC.3. Go to Preferences->Input/Codecs->Demuxers->H264, check “Advanced options” in the bottom. The option “Frames per Second” will show. Change that value to 5 and then save.4. Go to Preferences->Input/Codecs->Access modules->Real RTSP, check “Advanced options” in the bottom. The option “Caching value (ms) will show. Change that value to 1000 and then save. You may change it to a smaller value to reduce the delay.5. If the viewer is under NAT, go to Preferences->Demuxers->Access modules->RTP/RTSP, check “Advanced options” in the bottom. The option “Use RTP over RTSP (TCP)” will show. Check that option box. (Grandstream does NOT recommend this network environment)6. Close the Preferences window and go to File->Open Network Stream:a) Select RTSP as the protocolb) Enter the URL in the format of rtsp://admin: ADMIN_PASSWORD@DEVICE_IP_ADDRESS:DEVICE_RTSP_PORT.Change the blue text according to your configuration:•ADMIN_PASSWORD is the device’s web configuration password for admin. •DEVICE_IP_ADDRESS is the device IP. •DEVICE_RTSP_PORT is the RTSP port setting of the device. If the port uses default value 554, the port portion can be omitted from the URLc) Click OK to start the video.
FIGURE 7: SCREEN-SHOT OF VIDEO SURVEILLANCE* * PC client side running VLC as monitoring station
FIRMWARE UPGRADEOur latest official release can be downloaded from: http://www.grandstream.com/firmware.html. Firmware (or software) upgrades can be done either via TFTP or HTTP. The corresponding configuration settings are on the configuration page. End-users should NOT modify the configuration settings that are useful for ITSPs. To upgrade your unit firmware, follow these steps:1. Under Advanced Settings web page, enter your TFTP or HTTP Server IP address (or FQDN) next to the “Firmware Upgrade: Upgrade Server” field. 2. Select via TFTP or HTTP accordingly.3. If you plan to use Automatic Upgrade, set it to “Yes”, otherwise No (this will make it check for upgrade every time you reboot).UPGRADE THROUGH HTTPTo upgrade firmware via HTTP, the field “Firmware Upgrade and Provisioning: Upgrade Via” needs to be set to HTTP. The “Firmware Server Path” should be set to where the firmware files are located. For example: The user can use the following URL in the Firmware Server Path: firmware.mycompany.com: 6688/Grandstream/1.3.1.6 where firmware.mycompany.com is the FQDN of the HTTP server. It can also be in IP address format. “:6688” is the TCP port the HTTP server listening to, default http server listens to port 80. “/Grandstream/1.3.1.6 is the RELATIVE directory to the root dir on HTTP web server. UPGRADE THROUGH TFTPTo upgrade firmware via TFTP, set the field “Firmware Upgrade and Provisioning: Upgrade Via” to TFTP. The TFTP server can be configured in either IP address format or FQDN. To configure the TFTP server via the Web configuration interface, follow these five steps:1. Open your browser to input the IP address of the GXW410x. 2. Enter the admin password to enter the configuration screen. 3. Enter the TFTP server address or URL in the “Firmware Server Path” field near the bottom of the configuration screen. 4. Once the “Firmware Server Path” is set, update the change by clicking the “Update” button. 5. Reboot or power cycle the unit. If the configured upgrade server is found and a new code image is available, the GXW410x will retrieve the new image files by downloading them into the GXW410 gateway’s SRAM. During this stage, the GXW410x gateway’s LED will blink until the checking/downloading process is completed. Upon verification of checksum, the new code image will be saved into the Flash. If TFTP fails for any reason (e.g., TFTP server is not responding, there are no code image files available for upgrade, or checksum test fails, etc), the GXW410x gateway will stop the TFTP process and simply boot using the existing code image in the flash.
Firmware upgrade process may take as long as 20 minutes over the Internet, or just 20+ seconds if it is performed on a LAN. Grandstream recommends conducting firmware upgrades in a controlled LAN environment if possible. DOWNLOAD TFTP SERVERFor users who do not have a local TFTP server, Grandstream provides a NAT-friendly TFTP server on the public Internet for users to download the latest firmware upgrade automatically. Please check the Services section of Grandstream’s Web site to obtain this TFTP server IP address. Alternatively, user can download and install a free TFTP or HTTP server in his LAN for a firmware upgrade. A free Windows version TFTP server can be downloaded from:http://support.solarwinds.net/updates/New-customerFree.cfm. Directions for Downloading TFTP Server:1. Unzip the file and put all of the files under the root directory of the TFTP server. 2. Put the PC running the TFTP server and the GXW410x in the same LAN segment. 3. Go to File -> Configure -> Security to change the TFTP server's default setting from "Receive Only" to "Transmit Only" for the firmware upgrade. 4. Start the TFTP server, in the phone’s web configuration page.5. Configure the Firmware Server Path with the IP address of the PC.6. Update the change and reboot the unit. You can also download the free HTTP server from http://httpd.apache.org or just use Microsoft IIS web.
RESTORE FACTORY DEFAULT SETTINGWARNING! Restoring the Factory Default Setting will DELETE all configuration information of the phone. Please BACKUP or PRINT out all the settings before you approach to following steps. Grandstream will not take any responsibility if you lose all the parameters of setting and cannot connect to your VoIP service provider.The ONLY way to restore default factory settings is as follows: 1. Unplug the Ethernet cable.2. Locate the needle sized hole on the back panel of the gateway next to the Power connection.3. Enter a thin object in this hole and keep it pressed for about 7 seconds.4. You will see the Network port LEDs (green and orange) go off and on simultaneously; this indicates the reset went through.5. All settings have been erased and the gateway is back to factory settings.
Examples of GXW410x ConfigurationsAPPLICATION ONE: GXW CONNECTED WITH AN IP-PBX OR SIP SERVERScenario: A business with a traditional phone system (with or without broadband access) and an IP PBX or SIP Servers connecting to an Internet Telephone Service Provider (ITSP).APPLICATION TWO: GXW TO EXTEND A TRADITIONAL PBX SCENARIOScenario: a small business with traditional analog PBX lines and broadband access who want to extend their traditional PBX to virtually anywhere in the world, using the Internet. (Any SIP End point, such as Grandstream BugeTone, HandyTone, GXP-2000 or GXV-3000 are needed in this scenario) PSTNPSTNCloudCloudAnywhere in the worldGXW-410x4 or 8 PortsFXO LinesPSTN Analog LinesGrandstream IP PhonesIPPBX or SIP ServerIP/LANIP/WAN
APPLICATION THREE: GXW CONNECTED WITH AN IP-PBX OR SIP SERVER AND VIDEO SURVEILLANCEScenario: The GXW410x offers an additional video surveillance port which can be configured for surveillance. It is the only small business analog gateway that offers this security feature. Company A - Boston, MA6 employeesAny branch, anywhereInternet Internet CloudCloudIPPBX or SIP ServerGrandstream IP Phones PSTNPSTNCloudCloudIP/LANGXW-410xGXW-410xIP/LANFXS | IPPBX | SIP PlatformAny SIP endpointFXO LinesTraditional PBXFXO LinesOptionalAnywhere in the worldBranch A – Boston, MA6 employeesBranch B – Denver, CO4 employeesInterneInternet Cloudt CloudIPPBX or SIP ServerGrandstream IP PhonesPSTN PSTN CloudCloudIP/LANGXW 410xIPPBX or SIP ServerIP/LANGrandstream IP PhonesAnywhere in the worldGXW 410x
APPLICATION FOUR: USING A GXW FOR PURE IP- IP COMMUNICATION CONFIGURATIONScenario Four: The GXW410x offers an IP to IP pure IP Communications System configuration, where all locations use IP phones. Branch A – Boston, MA6 employeesBranch B – Denver, CO 4 employeesInternet Internet CloudCloudIPPBX or SIP ServerGrandstream IP Phones PSTNPSTNCloudCloudIP/LANGXW 410xIPPBX or SIP ServerIP/LANGrandstream IP PhonesAnywhere in the worldGXW 410x
FCC Warning This device complies with part 15 of the FCC Rules. Operation is subject to the following two conditions: (1) This device may not cause harmful interference, and (2) this device must accept any interference received, including interference that may cause undesired operation. Any Changes or modifications not expressly approved by the party responsible for compliance could void the user's authority to operate the equipment. FCC 15.105 Class B (b) For a Class B digital device or peripheral, the instructions furnished the user shall include the following or similar statement, placed in a prominent location in the text of the manual: Note: This equipment has been tested and found to comply with the limits for a Class B digital device, pursuant to part 15 of the FCC Rules. These limits are designed to provide reasonable protection against harmful interference in a residential installation. This equipment generates, uses and can radiate radio frequency energy and, if not installed and used in accordance with the instructions, may cause harmful interference to radio communications. However, there is no guarantee that interference will not occur in a particular installation. If this equipment does cause harmful interference to radio or television reception, which can be determined by turning the equipment off and on, the user is encouraged to try to correct the interference by one or more of the following measures: —Reorient or relocate the receiving antenna. —Increase the separation between the equipment and receiver. —Connect the equipment into an outlet on a circuit different from that to which the receiver is connected. —Consult the dealer or an experienced radio/TV technician for help.

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