Grandstream Networks, Inc.
COPYRIGHT ©2019 Grandstream Networks, Inc. http://www.grandstream.com All rights reserved. Information in this document is subject to change without notice. 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. The latest electronic version of this user manual is available for download here: http://www.grandstream.
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.
Table of Content DOCUMENT PURPOSE ................................................................................................. 6 CHANGE LOG ................................................................................................................ 7 Firmware Version 1.0.3.12 ..................................................................................................................... 7 Firmware Version 1.0.1.2 .................................................................................
Changing Viewer Password.......................................................................................................... 21 Changing HTTP/HTTPS Web Port ............................................................................................... 22 Web Configuration Pages Definitions .................................................................................................. 23 Status Page Definitions ........................................................................................
Table of Tables Table 1: HT813 Features at a Glance ......................................................................................................... 10 Table 2: HT813 Technical Specifications .................................................................................................... 11 Table 3: HT813 Connectors Definitions ...................................................................................................... 14 Table 4: HT813 LEDs Pattern Description .......................
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. Please visit http://www.grandstream.com/support to download the latest “HT813 User Guide”.
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 1.0.3.12 • Added support for T.38 Fax mode under FXO Port. [Fax Mode] • Updated "São Paulo" time zone to UTC-3. [Time Zone] • Added feature “Allow SIP Factory Reset” for FXS port Settings.
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 HT813 Administration Guide Version 1.0.3.
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 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 Mbps ports (RJ45) with integrated NAT router LED Indicators POWER, FXO, FXS, WAN, LAN.
Physical Universal Power Input: 100-240VAC, 50/60Hz Supply Output: 12V/0.5A Environmental Operational: 32° – 104°F or 0º – 40ºC 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 Compliance FCC/CE/C-TICK/ITU-K.21 HT813 Administration Guide Version 1.0.3.
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 FXS Connects the analog phone / fax machine to the ATA using an RJ-11 telephone cable. FXO 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. co Connects the ATA to your PC or switch using an Ethernet RJ45 network cable. DC 12V Reset Connects the ATA to PSU (Output: 12V/0.5A ) Factory reset button. Press for 7 seconds to reset factory default. settings.
1. 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. 2. Connect a computer or switch to the LAN port of the HT813 using an Ethernet Cable. 3. 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. 4. Power, LAN and FXS LED will be solidly lit when the HT813 is ready for use.
Figure 4: HT813 LEDs Pattern Table 4: HT813 LEDs Pattern Description LED Lights Power LED Status 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. LAN LED The LAN LED lights up when the HT813 is connected to your network through the LAN 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” of your adapter via the connected phone and check its IP address mode.
02 “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.
47 “Direct IP Calling” Enter the target IP address to make a direct IP call, after dial tone. (See “Make a Direct IP Call”.) 86 Voice Mail Access to your voice mails messages.
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. Enter the HT813’s IP address in the address bar of the browser. 4. Enter the administrator’s password to access the Web Configuration Menu. Note: The computer must be connected to the same sub-network as the HT813.
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 Admin Level Password 1. Access your HT813 web UI by entering its IP address in your favorite browser. 2. Enter your admin password (default: admin). 3.
3. Press Login to access your settings. 4. Go to Basic Settings → New Viewer Password and enter the new viewer password. 5. Confirm the new viewer password. 6. Press Apply at the bottom of the page to save your new settings. Figure 7: Viewer Level Password Changing HTTP/HTTPS Web Port 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 → HTTP(S) Web Port. 5.
Web Configuration Pages Definitions This section describes the options in the HT813 Web UI. • STATUS: Displays the system info, network status, account status, and line options. • BASIC SETTINGS: Configures the end user level password, IP address modes, web access, time zone settings and language. • ADVANCED SETTINGS: Configures networks, upgrading and provisioning, TR-069, security settings, date and time, syslog, audio settings, call settings and call progress tones.
Software Status Indicates the current software status of the HT (Running or Stopped). System Up Time Indicates actual system time and uptime since last reboot. PPPoE Link Up Indicates PPPoE connection status. NAT Indicates type of NAT when it is configured. Port Status Displays relevant information regarding the FXS and FXO ports about their registration, current status and their appropriate User ID.
Lockout Time Interval If login attempt failed 5 times, login would be locked out for the time length. Default 15 mins. Range 1-15 min. Web Access Mode Allows users to choose the Web Access Mode between “HTTPS” and “HTTP”. If “HTTPS” is selected, web UI will be accessed using HTTPS. Default is “HTTP”. HTTP Web Port Customizes HTTP port used to access the HT813 web UI. Default is 80. HTTPS Web Port Customizes HTTPS port used to access the HT813 web UI. Default is 443.
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 Note: If both blacklist and whitelist are not empty, the blacklist is processed first, followed by the whitelist. Internet Protocol Selects one of the following IP protocol modes: • IPv4 Only: Enforce IPv4 protocol only. • IPv6 Only: Enforce IPv6 protocol only. • Both, Prefer IPv4: Enable both IPv4 and IPv6 and prefer IPv4.
Preferred DNS server Specifies preferred DNS server to use when DHCP or PPPoE are set. Statically configured as IP Configure IP address, subnet Mask, default router IP address, 1 st preferred address DNS server, 2nd preferred DNS server. These fields are set to zero by default. IPv6 Address Allows users to configure the appropriate network settings on the HT813 to obtain IPv6 address. Users could select "DHCP", "Static IP”. By default, it is set to "DHCP".
• Bridge: In this mode, WAN port acts as DHCP client and passthrough to LAN port; devices connected behind LAN port will get an IP from your network DHCP server (same as WAN port). • WAN Only: In this mode, only WAN port is active. LAN port is not used. Default mode is NAT Router. Save the setting and reboot prior to configuring the HT813. NAT Maximum Ports Defines the number of ports that can be managed while in NAT router mode. Range: 0 – 4096, default is 1024.
Cloned WAN MAC Address This allows the user to change/set a specific MAC address on the WAN interface. Note: Set in Hex format. Enable LAN DHCP When set to “Yes”, device will function as a simple router and LAN port will provide IP addresses to internal network. Connect the WAN port to ADSL/Cable modem or any other equipment that provides access to public Internet LAN DHCP Base IP Base IP Address for a LAN port. Default factory setting is 192.168.2.1.
There are 3 types of factory reset: • ISP Data Reset: All VoIP related configuration (mainly everything located on FXS page). • VoIP Data Reset: All ISP (Internet Service Provider) configuration which may affect the IP address. • Full Reset: Both VoIP and ISP related configuration at the same time. Note: After choosing reset type, you will have to click the reset button for it to take effect. PSTN Access Code Key pattern to use PSTN line. Maximum 5 digits.
Black List for WAN Side Port It could be either port range or single port separated by a “,” Example: “5000-6000, 7000 “. STUN Server Configures IP address or domain name of STUN server. Only non-symmetric NAT routers work with STUN. Keep-alive Interval Sends periodically a blank UDP packet to SIP server in order to keep the "ping hole" on the NAT router open. Default is 20 seconds. Use STUN to detect network Uses STUN keep-alive to detect WAN side network problems.
Firmware File Postfix Checks if firmware file is with matching postfix before downloading it. This field enables user to store different versions of firmware files in one directory on the firmware server. Config File Prefix Checks if configuration files are with matching prefix before downloading them. It allows user to store different configuration files in one directory on the provisioning server. Config File Postfix Checks if configuration files are with matching postfix before downloading them.
Always Skip the Firmware Configures the HT813 to skip the firmware check, when this option is selected Check the HT813 will always skip searching for a new firmware. Disable SIP NOTIFY Disables the SIP NOTIFY Authentication on the ATA adapter. If set to “Yes”, Authentication the ATA adapter will not challenge NOTIFY with 401. Default is No Authenticate Conf File Authenticates configuration before being accepted. This protects the configuration from unauthorized modifications. Default is No.
CPE SSL Certificate Configures the Cert File for the ATA to connect to the ACS via SSL. CPE SSL Private Key Specifies the Cert Key for the ATA to connect to the ACS via SSL. Enable SNMP Default is No. SNMP Version Choose between (Version 1, Version 2c, or Version 3). SNMP Port Listening Port of SNMP daemon (Default 161). SNMP Trap IP Address IP address of trap destination.
SNMPv3 Trap Security Level • noAuthUser: Users with security level noAuthnoPriv and context name as noAuth. • authUser: Users with security level authNoPriv and context name as auth. • privUser: Users with security level authPriv and context name as priv. SNMPv3 Trap Authentication Select the Authentication Protocol: “None” or “MD5” or “SHA”. Protocol SNMPv3 Trap Privacy Select the Privacy Protocol: “None” or “AES/AES128” or “DES”.
DDNS Password Enter DDNS Password. 64 characters as Max String Length. DDNS Hostname Enter DDNS Hostname. 64 characters as Max String Length. DDNS Hash Enter DDNS Hash. 64 characters as Max String Length. System Ring Cadence Configuration option is set ring cadence on FXS port for all incoming calls.
• Always Disconnected: User can only make/receive VoIP calls. PSTN calls will not be possible. Default setting is Auto. Blacklist for Incoming Calls Allow users to block incoming calls from specific list of numbers. Maximum allow 10 SIP numbers and each number should be separated by a comma (‘,’) in web UI. Other allowed characters are 0-9, comma (“,”), asterisk (‘*’), pound sign (‘#’) and plus sign (‘+’). NTP Server Defines the URL or IP address of the NTP server.
Maximum TLS Version The Feature allows users to choose the Maximum TLS Version. Choices are: 1- Unlimited. 2- TLS 1.0. 3- TLS 1.1. 4- TLS 1.2. Default is Unlimited. Syslog Protocol The Feature allow users to customize the Syslog Protocol. The Syslog protocol can be either UDP or SSL/TLS. Default us UDP. Syslog Server URL or IP address of syslog server. Syslog Level Select the HT813 to report the log level. Default is NONE. The level is one of EXTRA DEBUG, DEBUG, INFO, WARNING or ERROR.
Download Device XML Allows user to download and save an XML file containing all the P values of Configuration each setting as configured at that point on the unit. For Security Reasons, Passwords won’t be Downloaded. Upload Firmware Allows the user to upgrade the firmware with a single firmware file by browsing and loading the file from your computer (local directory). Upload Configuration Allows to upload configuration file to the device. Export Backup Download backup file to local computer.
Prefer Primary Outbound If the user configures this option to “Yes”, when registration expires, the Proxy device will re-register via primary outbound proxy. By default, this option is disabled. Allow DHCP Option 120 Configures the HT813 to collect SIP server address from DHCP option 120. (override SIP Server) Default is No. Selects transport protocol for SIP packets; UDP or TCP or TLS.
By default, this option is disabled and the DNS SRV will use first SRV instead of the registered IP. Indicates E.164 number in “From” header by adding “User=Phone” parameter or using “Tel:” in SIP packets, if the HT813 has an assigned PSTN Number. • Disabled: Use “SIP User ID” information in the Request-Line and “From” header. Tel URI • User=Phone: “User=Phone” parameter will be attached to the Request-Line and “From” header in the SIP request to indicate the E.164 number. If set to "Enable".
Default is 1200 seconds. Enable SIP OPTIONS Keep Alive 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 server. Default setting is No. Configures the time interval when the phone adapter sends OPTIONS SIP OPTIONS Keep Alive message to SIP server. The default setting is 30 seconds, which means the Interval phone adapter will send an OPTIONS message to the server every 30 seconds.
Header Support SIP Instance ID Validate Incoming SIP Messages Check SIP User ID for Incoming INVITE Default is No. Includes “SIP Instance ID” attribute to “Contact” header in REGISTER request as defined in IETF SIP outbound draft. Default is No. Validates incoming messages. Default is No. Checks SIP User ID in the Request URI of incoming INVITE; if it doesn't match the HT813 SIP User ID, the call will be rejected. Direct IP calling will also be disabled. Default is No.
Use P-Emergency-Info This feature support of IEEE-48-addr and IEEE-EUI-64 in SIP header for Header emergency calls. Specifies which address (LAN or WAN address) the device will detect to use it in SIP Register Contact Header. When set to LAN, Contact header will SIP REGISTER Contact include local IP from ATA in REGISTER messages, while if set to WAN, Header Uses host/port/contact will be updated from SIP 401/403/404/407 Via header "received"/"rport" parameters in REGISTER messages.
order) Disable DTMF Negotiation priority. Uses above DTMF order without negotiation. Default is No. Generate Continuous When enabled the RFC2833 events are generated until key is released. RFC2833 Events Default is No. Send Hook Flash Event Default is No. If set to yes, flash will be sent as DTMF event. When it set to YES it allows the user to perform some call setting when both channels are used while pressing: • "Flash + 1" in order to hang up the current call and resume a call that was held.
Customizes the Ring Tone 1 to 3 with associate caller ID: when selected, if caller ID is configured, then the device will ONLY use this ring tone when the incoming call is from the Caller ID. System Ring Tone is used for all other calls. When selected but no Caller ID is configured, the selected ring tone will be used for all incoming calls using the FXS port. Distinctive ring tones can be configured not only for matching a whole number, but also for matching prefixes. In this case symbol “x+” will be used.
SDP Ring Timeout Delayed Call Forward Wait Timeout No Key Entry Timeout incoming SDP. Default is No. Stops ringing when incoming call if not answered within a specific period of time. Default is 60 seconds. Forwards incoming call if not answered within a specific period of time when delayed call forward is activated locally (using *92 code). Default value is 20 seconds. Initiates the call within this time interval if no additional key entry during dialing stage. Default is 4 seconds.
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; a. xx+ - at least 2 digits number; b. xx – exactly 2 digits number; c. ^ - exclude; d. . – wildcard, matches one or more characters e. [3-5] - any digit of 3, 4, or 5; f. [147] - any digit 1, 4, or 7; g. <2=011> - replace digit 2 with 011 when dialing h. < =1> - add a leading 1 to all numbers dialed, vice versa will remove a 1 from the number dialed | - or i.
voice mail or other application provided by service provider. In this case * should be predefined inside dial plan feature. As an example { *x+ } will allow to dial * followed by any length of numbers. SUBSCRIBE for MWI Send Anonymous Anonymous Call Rejection Sends SUBSCRIBE periodically (depends on “Register Expiration” parameter) for message waiting indication. Default is No. Sets “From”, “Privacy” and “P_Asserted_Identity” headers in outgoing INVITE message to “anonymous”, blocking caller ID.
Default is Omit. Specifies which end will act as refresher for incoming calls: UAS Specify Refresher • UAS: The handy tone acts as the refresher. • UAC: Callee or proxy server act as the refresher. Default is Omit. Force INVITE Enable 100rel Add Auth Header on Initial REGISTER Conference URI Use First Matching Vocoder in 200OK SDP Uses INVITE message to refresh the session timer. Default is No. Appends “100rel” attribute to the value of the required header of the initial signaling messages.
Allows detecting the absence of audio and conserves bandwidth by VAD preventing the transmission of "silent packets" over the network. Default is No. Symmetric RTP Fax Mode Changes the destination to send RTP packets to the source IP address and port of the inbound RTP packet last received by the device. Default is No. Specifies the fax mode: T.38 (Auto Detect) FoIP by default, or Pass-Through (must use codec PCMU/PCMA) Re-INVITE after Fax Tone Allows the unit to send out the re-INVITE for T.
Selects the caller ID scheme.
2000 ms. Default values are 300 minimum and 1100 maximum. Specifies the on-hook time for an on-hook event to be validated. HT813 On Hook Timing supports a range from 40 to 2000 ms. Default value is 400. Adjusts the voice path volume. • Rx is a gain level for signals transmitted by FXS • Tx is a gain level for signals received by FXS. Default = 0dB for both parameters. Loudest volume: +6dB Lowest volume: 6dB.
disabled, although the port will not be operational, in this state, there will be no dial tone when picking up the analog phone and making/receiving calls will not be possible. Primary SIP Server Failover SIP Server Configures SIP server IP address or domain name provided by VoIP service provider. This is the primary SIP server used to send/receive SIP messages from/to HT813. Specifies failover SIP server IP address or domain name provided by VoIP service provider.
• SRV: DNS SRV resource records indicate how to find services for various protocols. • NAPTR/SRV: Naming Authority Pointer according to RFC 2915. Default is A Record. 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. Registered IP By default, this option is disabled and the DNS SRV will use first SRV instead of the registered IP. Indicates E.
SIP Registration Failure Retry Wait Time upon 403 Forbidden Sends re-register request after specific time (in seconds) when registration process fails with error 403 Forbidden. Maximum interval is 3600 seconds (1 hour). Default is 1220 seconds. Enable SIP Enables SIP OPTIONS to track account registration status so the phone adapter will OPTIONS Keep send periodic OPTIONS message to server to track the connection status with the Alive server. Default setting is No.
INVITE Default is No. Authenticate Challenges the incoming INVITE for authentication with SIP 401 Unauthorized Incoming INVITE message. Default is No. Authenticate Configures whether to validate the domain certificate when download the server certificate firmware/config file. If it is set to "Yes", the phone will download the firmware/config file domain only from the legitimate server. The default setting is "No".
Reset Defines T1 timeout value. It is an estimate of the round-trip time between the client and server transactions. For example, the HT813 will attempt to send a request to a SIP server. SIP T1 Timeout The time it takes between sending out the request to the point of getting a response is the SIP T1 timer. If no response is received the timeout is increased to (2*T1) and then (4*T1). Request re-transmit retries would continue until a maximum amount of time defined by T2. Default is 0.5 seconds.
Use SIP UserAgent Header Do Not Escape '#' as %23 in SIP URI Configures the SIP User-Agent Header. Replaces # by %23 in some special situations. Default is No. Disable Multiple m Sends only one m line in SDP, regardless of how many m fields are in the incoming Line in SDP Ring Timeout SDP. Default is No. Stops ringing when incoming call if not answered within a specific period of time. Default is 60 seconds. Sends an early INVITE each time a key is pressed when a user dials a number.
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 with 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. 3.
transaction occurs beforehand. Default is 180 seconds. Min-SE Defines Minimum session expiration (in seconds). Default is 90 seconds. Caller Request Uses session timer when making outbound calls if remote party supports it. Timer Default is No. Callee Request Uses session timer when receiving inbound calls with session timer request. Timer Default is No. Uses session timer even if the remote party does not support this feature.
Voice Frames per Transmits a specific number of voice frames per packet. Default is 2; increases to TX 10/20/32/64 for G711/G726/G723/other codecs respectively. G723 Rate iLBC Frame Size Disable OPUS Stereo in SDP iLBC Payload type Operates at specified encoding rate for G.723 vocoder. Available encoding rates are 6.3kbps or 5.3kbps. Default is 6.3kbps. Specifies iLBC packet frame size (20ms or 30ms). Default is 20ms. Disables OPUS stereo in SDP. Default is No. Determines payload type for iLBC.
DTMF Caller ID FSK Caller ID Minimum RX Level (dB) FSK Caller ID Seizure Bits FSK Caller ID Mark Bits • ETSI-FSK prior to ringing with DTAS • ETSI-FSK prior to ringing with LR+DTAS • ETSI-FSK prior to ringing with RP • ETSI-DTMF during ringing • ETSI-DTMF prior to ringing with DTAS • ETSI-DTMF prior to ringing with LR+DTAS • ETSI-DTMF prior to ringing with RP • SIN 227 – BT • NTT JAPAN • DTMF Denmark prior to ringing with no DTAS no LR • DTMF Denmark prior to ringing with LR • DTMF S
to PSTN Hook Flash Duration (ms) INFO. The time period when the cradle is pressed (Hook Flash) to simulate a FLASH. Adjust this time value to prevent unwanted activation of the Flash/Hold and automatic phone ring-back. Voice path volume adjustment. • RX is a gain level for signals transmitted by FXO (FXO-To-VoIP volume) • TX is a gain level for signals received by FXO (FXO-To-PSTN volume). Default = 0dB for both parameters. Loudest volume: +6dB; Lowest volume: -6dB.
In certain countries, the central office will send a special busy tone to indicate when a call is disconnected from the remote side. User can pre-configure this tone on the ATA. The user should know the frequency values and cadences of these tones. PSTN Disconnect Here is an example for the syntax for a busy tone in the U.S.
• COMPLEX3 – 370 ohms + (620 ohms || 310nF) • COMPLEX4 – 600R, 270 ohms + (750 ohms || 150nF) • COMPLEX5 – 320 ohms + (1050 ohms || 230nF) • COMPLEX6 – 350 ohms + (1000 ohms || 210nF) • COMPLEX7 – 200 ohms + (680 ohms || 100nF) • COMPLEX8 – 370 ohms + (820 ohms || 110nF) • COMPLEX9 – 275 ohms + (780 ohms || 115nF) • COMPLEX10 – 120 ohms + (820 ohms || 110nF) Default is 600R – 600 ohms Number of Rings This is the number of rings the HT813 will wait to send the call to the VoIP side in case t
Inter-Digit Timeout When dialing from the PSTN to VoIP, subsequent digits have to be input within the (sec) period of inter-digit timeout. Otherwise the dial plan thinks it is the end of the digit input. Wait for Dial Tone Wait for Dial tone is used for one stage VoIP to PSTN calls. If set to Yes, the device will first obtain a PSTN line and a dial tone from a central office. After obtaining the dial tone, the digits dialed will be sent to the central office.
Preferred Vocoder (Codec) The HT813 supports following voice codecs. On FXS/FXO page, choose the order of your favorite codecs: • PCMU/A (or G711µ/a) • G729 A/B • G723.1 • G726 • iLBC • OPUS Configuring HT813 Through Voice Prompts As mentioned previously, The HT813 has a built-in voice prompt menu for simple device configuration. Please refer to “Understanding HT813 Interactive Voice Prompt Response Menu” for more information about IVR and how to access its menu.
Configuration through a Central Server The HT813 can be automatically configured from a central provisioning system. When HT813 boots up, it will send TFTP, FTP/FTPS or HTTP/HTTPS requests to download configuration files, “cfg000b82xxxxxx” and “cfg00082xxxxxx.xml”, where “000b82xxxxxx” is the LAN MAC address of the HT813. If the download of “cfgxxxxxxxxxxxx.xml” is not successful, the provision program will issue request a generic configuration file “cfg.xml”.
d. Prefer Primary SIP Server to No or Yes depending on your configuration. Set to No if no Failover SIP Server is defined. If “Yes”, account will register to Primary SIP Server when failover registration expires. e. Outbound Proxy: Set your Outbound Proxy IP Address or FQDN. Leave empty if not available. f. SIP User ID: User account information, provided by VoIP service provider (ITSP). Usually in the form of digit similar to phone number or actually a phone number. g.
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).
Call Features The HT813 supports all the traditional and advanced telephony features. Table 11: HT813 Call Features Key *02 *03 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. *16 Enable SRTP *17 Disable SRTP *30 Block Caller ID (for all subsequent calls) *31 Send Caller ID (for all subsequent calls) *47 Direct IP Calling.
*79 Disable Do Not Disturb (DND): When disabled, incoming calls are accepted. *87 Blind Transfer *90 *91 *92 *93 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. Cancel Delayed Call Forward.
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 Please follow below steps in order to upgrade the firmware version of your HT813: 1.
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. Click “Upload Firmware” and wait few minutes until the new program is loaded. Note: Always check the status page to see that the program version has changed.
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. If set to “weekly on day (0-6)” the auto check will be done on the day specified in this field.
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.
C 2222 D 33 (press the “3” key twice, “D” will show on the LCD) E 333 F 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. Enter your admin password (default: admin). 3. Press Login to access your settings. 4. Go to Basic Settings → Reset Type 5. Press Reset button (after selecting the reset type).
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 and User and Developer Forum for answers to your general questions. If you has purchased our products through a Grandstream Certified Partner or Reseller, please contact them directly for immediate support.
