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GNU GPL INFORMATION The firmware for the HT813 contains third-party software licensed under the GNU General Public License (GPL). Grandstream uses software under the specific terms of the GPL. Please see the GNU General Public License (GPL) for the exact terms and conditions of the license. Grandstream GNU GPL related source code can be downloaded from Grandstream web site from: http://www.grandstream.com/support/faq/gnu-general-public-license/gnu-gpl-information-download P a g e...
DOCUMENT PURPOSE This document describes the basic concept and tasks necessary to use and configure your HT813. It covers also the topic of connecting and configuring the HT813, making basic operations and the call features. to download the latest “HT813 User Guide”. Please visit http://www.grandstream.com/support This guide covers following topics:...
CHANGE LOG This section documents significant changes from previous versions of admin guide for HT813. Only major new features or major document updates are listed here. Minor updates for corrections or editing are not documented here. Firmware Version 1.0.0.8 This is the initial version for HT813.
GUI INTERFACE EXAMPLES http://www.grandstream.com/sites/default/files/Resources/HT813_web_gui.zip 1. Screenshot of Login Page 2. Screenshots of Status Page 3. Screenshots of Basic Settings Page 4. Screenshots of Advanced Settings Page 5. Screenshots of FXS Port Page 6. Screenshots of FXO Port Page P a g e HT813 Administration Guide Version 1.0.0.8...
WELCOME Thank you for purchasing Grandstream’s HT813, It is the first ATA in the HandyTone 81x series offering functions as a true 3-in-1 gateway for PSTN network, analog telephone FXS interface and IP network. It enables remote call origination and termination from/to PSTN. The HT813 is specifically designed to be an easy to use and affordable VoIP solution for both the residential user and the remote user.
PRODUCT OVERVIEW The HT813 is an analog telephone adapter (ATA) featuring 1 analog telephone FXS port and 1 PSTN line FXO port. The integration of FXO and FXS ports enables remote call origination and termination to and from the PSTN line. The 1 FXS port allows for extension of a VoIP service to 1 analog phone. HT813’s ultra-compact size, voice quality, advanced VoIP functionality, security protection and auto provisioning options enable users to take advantage of VoIP on analog phones and enables service providers to offer high quality IP service.
HT813 Technical Specifications The following table resumes all the technical specifications including the protocols/standards supported, voice codecs, telephony features, languages and upgrade/provisioning settings for the HT813. Table 2: HT813 Technical Specifications Interfaces Telephone Interfaces One (1) RJ11 FXS port, One (1) RJ11 FXO PSTN line port with lifeline support Network Interface Two (2) 10/100/1000 Mbps ports (RJ45) with integrated NAT router LED Indicators...
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Physical Universal Power Input: 100-240VAC, 50/60Hz Supply Output: 12V/0.5A Operational: 32° – 104°F or 0º – 40ºC Environmental Storage: 14° – 140°F or -10º – 60ºC Humidity: 10 – 90% Non-condensing Dimensions and 130.5 x 90.5 x 29 mm (L x W x D) Weight Weight: 0.142Kg Compliance...
GETTING STARTED This chapter provides basic installation instructions including the list of the packaging contents and also information for obtaining the best performance with the HT813. Equipment Packaging The HT813 ATA package contains: Figure 1: HT813 Package Contents Note: Check the package before installation. If you find anything missing, contact your system administrator HT813 Ports Description The following figure describes the different ports on the back panel of the HT813.
Table 3: HT813 Connectors Definitions Connects the analog phone / fax machine to the ATA using an RJ-11 telephone cable. FXO telephone port (PSTN Port) 1x PSTN pass-through and life line port. Connects the ATA to your router, switch or modem using an Ethernet RJ45 network cable.
Insert a standard RJ11 telephone cable into FXS port and connect the other end of the telephone cable to a standard touch-tone analog telephone. Connect a computer or switch to the LAN port of the HT813 using an Ethernet Cable. Insert the power adapter into the HT813 and connect it to a wall outlet and make sure to respect the technical specifications of the power adapter used.
Figure 4: HT813 LEDs Pattern Table 4: HT813 LEDs Pattern Description LED Lights Status Power LED The Power LED lights up when the HT813 is powered on and it flashes when the HT813 is booting up. WAN LED The WAN LED lights up when the HT813 is connected to your network through the WAN port.
CONFIGURATION GUIDE The HT813 can be configured via one of two ways: The IVR voice prompt menu. The Web GUI embedded on the HT813 using PC's web browser. Obtain HT813 IP Address via Connected Analogue Phone HT813 is by default configured to obtain the IP address from DHCP server where the unit is located. To know which IP address is assigned to your HT813, you should access to the “Interactive Voice Response Menu”...
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“IP Address “ + IP address The current WAN IP address is announced If using “Static IP Mode”, enter 12-digit new IP address. You need to reboot your HT813 for the new IP address to take Effect. “Subnet “ + IP address Same as menu 02 “Gateway “...
“Direct IP Calling” Enter the target IP address to make a direct IP call, after dial tone. (See “Make a Direct IP Call”.) Voice Mail Access to your voice mails messages. “RESET” Press “9” to reboot the device Enter MAC address to restore factory default setting (See RESTORE FACTORY DEFAULT SETTINGS section) “Invalid Entry”...
Accessing the Web UI Via WAN port 1. You may check your HT813 IP address using the IVR on the connected phone. Please see Obtain the HT813 IP address via the connected analogue phone 2. Open the web browser on your computer. 3.
reboot button on the web page Saving the Configuration Changes After users make changes to the configuration, pressing Update button will save but not apply the changes until Apply button is clicked. Users can instead directly press Apply button. When a reboot is required to apply changes, the web page will prompt the user to reboot by offering a reboot button on the web page.
Changing Viewer Password 1. Access your HT813 web UI by entering its IP address in your favorite browser. 2. Enter your admin password (default: admin). 3. Press Login to access your settings. 4. Go to Basic Settings New Viewer Password and enter the new viewer password. 5.
Figure 8: Web HTTP(S) Port Web Configuration Pages Definitions This section describes the options in the HT813 Web UI. As mentioned, you can log in as an administrator or an end user. STATUS: Displays the system info, network status, account status, and line options. ...
WAN IPv4 Address Displays assigned IPv4 address. WAN IPv6 Address Displays assigned IPv6 address. Product Model Displays product model info. Default is HT813. Hardware Version Displays the hardware revision information and the part number. Software Version Program: Specifies Program version. This is the main firmware release number, which is always used for identifying the software system of the HT813.
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New Viewer Configures viewer level password. Case sensitive and max. Length of 25 characters Password Re-enter the viewer password to confirm change viewer password on web GUI to avoid Confirm Viewer Password typo or mistakes. Web/SSH Access Web Session Configure timer to logout web session during idle. Timeout Default is 10 min.
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Black List for WAN Allows users to configure the black List for WAN Side to ban WAN side web access. Side Multiple IPs are supported and need to be separated by “space”. Example: 192.168.5.222 192.168.5.223 192.168.7.0/24 Internet Protocol Selects one of the following IP protocol modes: ...
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DHCP vendor class Exchanges vendor class ID by clients and servers to convey particular configuration or other identification information about a client. Default is HT8XX. Defines the PPPoE username. Necessary if ISP requires you to use a PPPoE (Point PPPoE account ID to Point Protocol over Ethernet) connection.
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NAT TCP Timeout NAT TCP idle timeout in seconds. Connection will be closed after preconfigured, timeout if not refreshed. Range: 0 - 3600 NAT UDP Timeout NAT TCP idle timeout in seconds. Connection will be closed after preconfigured, timeout if not refreshed. Range: 0 – 3600, default is 300 Uplink Bandwidth Specifies the maximum uplink bandwidth permitted by the device.
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LAN DHCP Start IP Default value is 100. The last segment of IP address assigned to the HT813 in the LAN Network. Default configuration assigns IP address (to local network devices) starting from 192.168.2.100. LAN DHCP End IP Default value is 199. This parameter allows a user to limit the number of local network devices connected to the internal router.
Unconditional Call VoIP calls will be forwarded to the specified PSTN number. Forward to PSTN Specify PSTN number. Unconditional Call Incoming PSTN calls will be forwarded to the VoIP number. Forward to VoIP Specify “User ID”, “SIP Server” and “SIP Destination Port” Advanced Settings Page Definitions Table 8: Advanced Settings Advanced Settings...
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Use STUN to detect network Uses STUN keep-alive to detect WAN side network problems. If keep-alive request does not yield any response for configured number of times connectivity (minimum 3), the device will restart the TCP/IP stack. If the STUN server does not respond when the device boots up, the feature is disabled.
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Config File Postfix Checks if configuration files are with matching postfix before downloading them. It allows user to store different configuration files in one directory on the provisioning server. Obtains configuration and upgrade server’s information using options 66 Allow DHCP Option 66 to Override Server from DHCP server.
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This feature allows users to validate server certificates with our trusted list of Validate Server TLS connections. Default is enabled. Certificates The device needs to reboot after changing the setting. SIP TLS Certificate Specifies SSL certificate used for SIP over TLS is in X.509 format. The HT813 has built-in private key and SSL certificate.
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SNMP Port Listening Port of SNMP daemon (Default 161). SNMP Trap IP Address IP address of trap destination. Port of Trap port Port of Trap destination (Default 162) SNMP Trap Version Choose between (Version 1, Version 2c, or Version 3). SNMP Trap Interval Time interval between traps (Default is 5).
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Select the Privacy Protocol: “None” or “AES/AES128” or “DES”. SNMPv3 Trap Privacy Protocol Enter the Trap Authentication Key. SNMPv3 Trap Authentication SNMPv3 Trap Privacy Key Enter the Trap Privacy Key. Enable RADIUS Web Access Default is No. Control Action upon RADIUS Auth Choose action upon RADIUS server error.
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Call Progress Tones: Using these settings, users can configure tone frequencies and cadence according to their preference. By default, they are set to North American Dial Tone Ringback Tone frequencies. Busy Tone Configure these settings with known values to avoid uncomfortable high pitch ...
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NTP Server Defines the URL or IP address of the NTP server. The ATA may obtain the date and time from the server. The default setting is “pool.ntp.org”. Defines whether DHCP Option 42 should override NTP server or not. When Allow DHCP Option 42 to enabled, DHCP Option 42 will override the NTP server if it’s set up on the LAN.
Download Device Allows user to download and save a text file containing all the P values of each Configuration setting as configured at that point on the unit. For Security Reasons, Passwords won’t be Downloaded. Allows user to download and save an XML file containing all the P values of Download Device XML Configuration each setting as configured at that point on the unit.
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Configures the backup outbound proxy to be used when the “Outbound Proxy” Backup Outbound registration fails. By default, this field is left empty. Proxy If the user configures this option to “Yes”, when registration expires, the device will re- Prefer Primary Outbound Proxy register via primary outbound proxy.
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When this option is set to “Yes”, when the HT is registered on second SRV and makes DNS SRV use an outbound call, it will try the second SRV (registered IP) first. By default, this option Registered IP is disabled and the DNS SRV will use first SRV instead of the registered IP. Indicates E.164 number in “From”...
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Enable SIP Enables SIP OPTIONS to track account registration status so the phone adapter will send periodic OPTIONS message to server to track the connection status with the OPTIONS Keep Alive server. Default setting is No. Configures the time interval when the phone adapter sends OPTIONS message to SIP SIP OPTIONS Keep server.
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Includes “SIP Instance ID” attribute to “Contact” header in REGISTER request as Support SIP Instance ID defined in IETF SIP outbound draft. Default is No. Validate Incoming Validates incoming messages. Default is No. Messages Checks SIP User ID in the Request URI of incoming INVITE; if it doesn't match the Check SIP User ID for Incoming HT813 SIP User ID, the call will be rejected.
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Specifies which address (LAN or WAN address) the device will detect to use it in SIP SIP REGISTER Register Contact Header. When set to LAN, Contact header will include local IP from Contact Header ATA in REGISTER messages, while if set to WAN, host/port/contact will be updated Uses from SIP 401/403/404/407 Via header "received"/"rport"...
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Send Hook Flash Default is No. If set to yes, flash will be sent as DTMF event. Event Overrides the default settings for call control when both channels are in use. Flash Digit Control Enables do not disturb, call forward and other call features via the local feature codes Enable Call on the base.
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Disable Call Disables displaying caller ID when receiving a second incoming call. Default is No. Waiting Caller ID Disables playing call waiting tone during active call when receiving a second incoming Disable Call Waiting Tone call. The CWCID will still be displayed. Default is No. Disable Connected Disables displaying the number of the person answering the phone.
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Dial Plan Prefix Adds specified prefix to dialed number. Treats “#” as the “Send” (or “Dial”) key. If set to “No”, this “#” key can be included as part of the dialed number. Use # as Dial Key Default is Yes. P a g e HT813 Administration Guide Version 1.0.0.8...
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Dial Plan Rules: 1. Accept Digits: 1,2,3,4,5,6,7,8,9,0 , *, #, A,a,B,b,C,c,D,d 2. Grammar: x - any digit from 0-9; xx+ - at least 2 digits number; xx – exactly 2 digits number; ^ - exclude; . – wildcard, matches one or more characters [3-5] - any digit of 3, 4, or 5;...
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Sends SUBSCRIBE periodically (depends on “Register Expiration” parameter) for SUBSCRIBE for message waiting indication. Default is No. Sets “From”, “Privacy” and “P_Asserted_Identity” headers in outgoing INVITE message Send Anonymous to “anonymous”, blocking caller ID. Default is No. Rejects incoming calls with anonymous caller ID with “486 Busy here” message. Anonymous Call Rejection Default is No.
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Uses INVITE message to refresh the session timer. Force INVITE Default is No. Appends “100rel” attribute to the value of the required header of the initial signaling Enable 100rel messages. Default is No. Adds “Authentication” header with blank “nonce” attribute in the initial SIP REGISTER Add Auth Header on Initial request.
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Re-INVITE after Fax Allows the unit to send out the re-INVITE for T.38 or Fax Pass Through if a fax tone is Tone Detected detected. Default is Enabled Selects jitter buffer type (Fixed or Adaptive) based on network conditions. Jitter Buffer Type ...
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NTT JAPAN DTMF Denmark prior to ringing with no DTAS no LR DTMF Denmark prior to ringing with LR DTMF Sweden/Finland prior to ringing with LR DTMF Brazil DTMF-FSK Brazil Defines the start and stop tones as delimiters for the caller ID. DTMF Caller ID Start Tone and Stop Tone can be set to “Default”, “A”, “B”, “C”, “D”...
Disable Line Echo Disables the LEC will per call base. Recommended for FAX/Data calls. Canceller (LEC) Default is No. Disables the NEC will per call base. Recommended for FAX/Data calls. Disable Network Echo Suppressor Default is No. Outgoing Call Defines the call duration limit for the outgoing calls, Default is 0 (No limit). Duration Limit Ring Frequency Configures ringing frequency for your phone.
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Selects transport protocol for SIP packets; UDP or TCP or TLS. Please make sure your SIP Server or network environment supports SIP over the selected transport method. SIP Transport Default is UDP. Indicates type of NAT for each account. This parameter configures whether the NAT NAT Traversal traversal mechanism is activated.
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Controls whether the HT813 needs to send REGISTER messages to the proxy server. SIP Registration Default setting is Yes. Controls whether to clear SIP user’s information by sending un-register request to the Unregister on proxy server. The un-registration is performed by sending a REGISTER message with Reboot Contact set to * and Expires=0 parameters to the SIP server.
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Defines the local RTP-RTCP port pair the HT813 will listen and transmit. It is the HT813 Local RTP Port RTP port for channel 0. The default value for FXS port is 5012. Controls whether to use configured or random SIP ports. This is usually necessary when Use Random SIP Port multiple HT813 are behind the same NAT.
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Specifies if the P-Preferred-Identity Header will be presented in the SIP INVITE message. If set to “default", the P-Preferred-Identity Header will be omitted in SIP Use P-Preferred- INVITE message when Telkom SA or CBCO is active. If set to "Yes", the P-Preferred- Identity Header Identity Header will always be presented.
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Generate When enabled the RFC2833 events are generated until key is released. Continuous Default is No. RFC2833 Events Flash Digit Control Overrides the default settings for call control when both channels are in use. Determines a SIP Extension to notify the SIP server that the HT813 is behind a Proxy-Require NAT/Firewall.
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Dial Plan Rules: 1. Accept Digits: 1,2,3,4,5,6,7,8,9,0 , *, #, A,a,B,b,C,c,D,d 2. Grammar: x - any digit from 0-9; xx+ - at least 2 digits number; xx – exactly 2 digits number; ^ - exclude; . – wildcard, matches one or more characters [3-5] - any digit of 3, 4, or 5;...
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Sends SUBSCRIBE periodically (depends on “Register Expiration” parameter) for SUBSCRIBE for message waiting indication. Default is No. Rejects incoming calls with anonymous caller ID with “486 Busy here” message. Anonymous Call Rejection Default is No. Selects Soft switch vendors’ special requirements Example of vendors: Standard, Special Feature Broadsoft, CBCOM, RNK, Huawei, China Mobile, ZTE IMS, PhonePower, TELKOM SA, Vonage, Metaswitch, CenturyLink, MTS.
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Appends “100rel” attribute to the value of the required header of the initial signaling Enable 100rel messages. Default is No. Add Auth Header Adds “Authentication” header with blank “nonce” attribute in the initial SIP REGISTER on Initial request. Default is No. REGISTER Use First Matching Includes only the first matching vocoder in its 200OK response, otherwise it will include...
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High (initial 200ms, min 40ms, max 600ms) Note: not all vocoders can meet the high requirement. Jitter Buffer Length Medium (initial 100ms, min 20ms, max 200ms). Low (initial 50ms, min 10ms, max 100ms). Selects SRTP mode to use (“Disabled”, “Enabled but not forced”, or “Enabled and forced”).
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According to customer’s choice CID information will be transferred from PSTN network to VoIP network using following rules: Relay via SIP From - PSTN CID is in the SIP From field Relay via P-Asserted-Identity - SIP From field uses the pre-configured account user Id.
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FXO Termination This value should be used in case the PSTN provider uses line power drop to indicate call completion to the end point. In this case the HT813 will search for a power drop Enable Current Disconnect for a preconfigured time frame to disconnect such calls from a VoIP extension. Default is Yes.
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GREECE ITALY JAPAN NORWAY SPAIN SWEDEN Default is USA Impedance-Based Select the Impedance used by the PSTN service provider. 600R – 600 ohms 600C – 600 ohms + 2.16uF 900R – 900 ohms ...
PSTN Idle Wait Used to customize timeout value between PSTN outgoing calls. Range is 0-10 seconds. Default is 4 seconds. Timeout between Outgoing Calls Channel Dialing DTMF Digit Length Digit length and Dial Pause are port digit dialing configurations; FXO needs to dial out (ms) digits for VoIP to PSTN 1 stage calls, and unconditional call forward to PSTN, and route to PSTN.
2. Use random SIP/RTP ports (under advanced settings webpage), this setting depends on your network settings. Generally, if you have multiple IP devices under the same network, it should be set to Yes. If using a public IP address, set this parameter to No. 3.
FIRMWARE SERVER IP ADDRESS Select voice menu option 13 to configure the IP address of the firmware server. CONFIGURATION SERVER IP ADDRESS Select voice menu option 14 to configure the IP address of the configuration server. UPGRADE PROTOCOL Select the menu option 15 to choose firmware and configuration upgrade protocol between TFTP, FTP, FTPS, HTTP and HTTPS.
Register a SIP Account The HT813 supports 2 SIP accounts. Please refer to the following steps in order to register your accounts via web user interface 1. Access your HT813 web UI by entering its IP address in your favorite browser. 2.
Figure 9: FXS Port Settings After applying your configuration, your account will register to your SIP Server, you can verify if it has been correctly registered with your SIP server from your HT813 web interface under Status Port Status Registration (If it displays Registered, it means that your account is fully registered, otherwise it will display Not Registered so in this case you must double check the settings or contact your provider).
Figure 10: Accounts Status Rebooting HT813 from Remote Press “Reboot” button at the bottom of the configuration menu to reboot the ATA remotely. The web browser will then display a message window to confirm that reboot is underway. Wait 30 seconds to log in again. P a g e HT813 Administration Guide Version 1.0.0.8...
Call Features The HT813 supports all the traditional and advanced telephony features. Table 11: HT813 Call Features Call features Forcing a Codec (per call) *027110 (PCMU), *027111 (PCMA), *02723 (G723), *02729 (G729), *027201 (iLBC). Disable LEC (per call) Dial “*03” +” number”. No dial tone is played in the middle.
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Busy Call Forward: Dial “*90” and then the forwarding number followed by “#”. Wait for dial tone then hang up. Cancel Busy Call Forward. To cancel “Busy Call Forward”, dial “*91”, wait for dial tone, then hang up. Delayed Call Forward. Dial “*92” and then the forwarding number followed by “#”. Wait for dial tone then hang up.
UPGRADING AND PROVISIONING The HT813 can be upgraded via TFTP/FTP/FTPS/HTTP/HTTPS by configuring the URL/IP Address for the TFTPFTP/FTPS/HTTP/HTTPS server and selecting a download method. Configure a valid URL for TFTP or FTP/FTPS or HTTP/HTTPS (default is HTTPS); the server name can be FQDN or IP address. Examples of valid URLs: firmware.grandstream.com or fw.ipvideotalk.com/gs Firmware Upgrade procedure...
Upgrading via Local Directory 1. Download the firmware file from Grandstream web site 2. Unzip it and copy the file in to a folder in your PC 3. From the HT813 web interface (Advanced Settings page) you can browse your hard drive and select the folder you previously saved the file (HT8xfw.bin) 4.
Managing Firmware and Configuration File Download When “Automatic Upgrade” is set “Yes, every” the auto check will be done in the minute specified in this field. If set to “daily at hour (0-23)”, Service Provider can use P193 (Auto Check Interval) to have the devices do a daily check at the hour set in this field with either Firmware Server or Config Server.
RESTORE FACTORY DEFAULT SETTINGS Warning: Restoring the Factory Default Settings will delete all configuration information on the phone. Please backup or print all the settings before you restore to the factory default settings. Grandstream is not responsible for restoring lost parameters and cannot connect your device to your VoIP service provider. There are three (3) methods for resetting your unit: Using the Reset Button To reset default factory settings using the reset button please follow the steps above:...
2222 33 (press the “3” key twice, “D” will show on the LCD) 3333 For example: if the MAC address is 000b8200e395, it should be keyed in as “0002228200333395” Reset from Web Interface (Reset Type) 1. Access your HT813 UI by entering its IP address in your favorite browser. 2.
EXPERIENCING THE HT813 Please visit our website: http://www.grandstream.com to receive the most up- to-date updates on firmware releases, additional features, FAQs, documentation and news on new products. We encourage you to browse our product related documentation, FAQs User and Developer Forum for answers to your general questions.
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