Grandstream Networks, Inc. HT–386 Dual FXS Port Analog Telephone Adaptor HT–386 User Manual Firmware Version 1.0.3.64 www.grandstream.com support@grandstream.
TABLE OF CONTENTS HT–386 USER MANUAL WELCOME ................................................................................................................................................... 4 Safety Compliances.................................................................................................................................. 4 Warranty ...................................................................................................................................................
TABLE OF FIGURES HT–386 USER MANUAL FIGURE 1: FIGURE 2: FIGURE 3: FIGURE 4: FIGURE 5: CONNECTING THE HT–386 ............................................................................................................ 5 INTERCONNECTION DIAGRAM OF THE HT–386 ................................................................................ 6 SCREENSHOT OF CONFIGURATION LOG-IN PAGE ............................................................................ 18 SCREENSHOT OF CONFIGURATION UPDATE MODE ...........
WELCOME Grandstream HandyTone Analog Telephone Adapters/IAD series offers a comprehensive line of affordable VoIP access devices based on Grandstream’s innovative technology platform. The HandyTone series offers the entry-level IP Telephony user superb audio quality, rich functionalities, interoperability with the leading 3rd party VoIP providers, and compatibility with most service providers.
INSTALLATION EQUIPMENT PACKAGING The HT–386 ATA package contains: • • • One HT–386 Main Case One Universal Power Adaptor One Ethernet Cable CONNECTING YOUR ATA HT-386 Analog Telephone Adaptor is an all-in-one VoIP integrated device designed to be a total solution for networks providing VoIP services. The HT-386 VoIP features are available using a regular analog telephone.
FIVE EASY STEPS TO INSTALL THE HT–386 Following are the steps to install a HT–386: 1. Connect a standard touch-tone analog telephone (or fax machine) to FXS port 1. 2. Connect another standard touch-tone analog telephone (or fax machine) to FXS port 2. 3. Insert a standard telephone cable into the LINE port of HT–386. and connect the other end of the telephone cable to a wall jack. 4. Insert the Ethernet cable into the LAN port of HT–386.
PRODUCT OVERVIEW The HT–386 is a next generation dual-port SIP IAD for Internet data, voice, and fax. It supports two (2) FXS ports, each with an independent SIP account or SIP server platform, and a PSTN pass through line for toggling operations between SIP and PSTN networks. The HT–386 offers the entry-level IP telephony user superb audio quality, rich functionalities, interoperability with the leading 3rd party VoIP providers, and compatibility with most service providers.
TABLE 2: HT–386 TECHNICAL SPECIFICATIONS Lines/SIP Accounts Protocol Support Feature Keys LAN/WAN Interface Device Management 2 lines / 2 SIP accounts SIP 2.0 (RFC 3261), TCP/UDP/IP, RTP/RTCP, HTTP, ARP/RARP, ICMP, DNS, DHCP, NTP, TFTP, PPPoE protocols 1 button RJ-45 10 Mbps Web interface or via secure (AES encrypted) central configuration file for mass deployment Support device configuration via built-in IVR, Web browser or central configuration file through TFTP or HTTP Support Layer 2 (802.
HARDWARE SPECIFICATION TABLE 3: HT-386 HARDWARE SPECIFICATION LAN interface FXS telephone port PSTN Port Button LED Universal Switching Power Adaptor Dimension Weight Temperature Humidity 1xRJ45 10Base-T 2 x FXS 1x PSTN pass-through or life line port 1 Green and Red color Input: 100-240VAC 50-60 Hz Output: +5VDC, 1200mA UL certified 70mm (W) x 130mm (D) x 27mm (H) 0.6lbs (0.3kg) 40 - 130oF / 5 – 45oC 10% - 90% (non-condensing) Compliance Grandstream Networks, Inc. HT-386 User Manual Firmware 1.0.3.
BASIC OPERATIONS GET FAMILIAR WITH VOICE PROMPT The HT–386 has a stored voice prompt menu for quick browsing and simple configuration. Currently, the voice prompt menu and the LED button is designed for FXS Port 1 ONLY. To enter this voice prompt menu, press the LED button or press “***” from the analog phone.
47 “Direct IP Calling” Enter a 12 digit IP address to make a direct IP call, after dial tone. (See “Make a Direct IP Call”.) 99 “RESET” Press “9” to reboot the device; or Enter encoded MAC address to restore factory default setting (See “Restoring Factory Settings”) “Invalid Entry” Automatically returns to main menu NOTE: • Once the button is pressed, you will hear the voice prompt main menu.
DIRECT IP CALLS Direct IP calling allows two parties, that is, a HT with an analog phone and another VoIP Device, to talk to each other in an ad hoc fashion without a SIP proxy. This kind of VoIP calls can be made between two parties if: • • • Both the HT–386 and other VoIP Device(i.e.
NOTE: “Enable Call Feature” must be set to “Yes” in web configuration page. Caller A can place a call on hold and wait for one of three situations: 1. A quick confirmation tone (similar to call waiting tone) followed by a dial tone. This indicates the transfer is successful (transferee has received a 200 OK from transfer target). At this point, Caller A can either hang up or make another call. 2. A quick busy tone followed by a restored call (on supported platforms only).
PSTN PASS THROUGH HT-386 supports PSTN pass through on FXS port 1. User can make and receive PSTN calls with attached analog phone in Phone 1 port. Phone 2 port (or FXS port 2) does NOT have this feature. • • To receive PSTN calls, simply make phone off hook when the analog phone rings. To make a PSTN call, simply press the PSTN access code (*00 is default, or any number configured in web configuration page) to switch to the PSTN line and get dial tone, then dial the number.
CALL FEATURES Following table shows the call features (* code) of HT-386. TABLE 5: HT–386 CALL FEATURE DEFINITIONS Key *23 *50 *51 Call Features 3 way Conferencing Refer section above above for procedure to perform 3 way Calling. Block Caller ID (for all subsequent calls) Send Caller ID (for all subsequent calls) Block Caller ID (per call). Dial “*67” + ” number ”. No dial tone will be played in the middle. Send Caller ID (per call). Dial “*82” + ” number ”. No dial tone will be played in the middle.
LED LIGHT PATTERN INDICATION Following tables show the LED light pattern indication. The LED shows PHONE1 status only. TABLE 6: HT–386 LED DEFINITIONS RED LED indicates not normal status Button flashes every 2 seconds. (if DHCP is configured) DHCP Failed or WAN No Cable Button flashes every 2 seconds. (if SIP server is configured) HT–496fails to register Button flashes every 2 seconds. Red light steady.
CONFIGURATION GUIDE CONFIGURING HT–386 THROUGH VOICE PROMPT DHCP MODE Follow Table 3 with voice menu option 01 to enable HT-386 to use DHCP. STATIC IP MODE Follow Table 3 with voice menu option 01 to enable HT-386 to use STATIC IP mode, then use option 02, 03, 04 to set up HT-386’s IP, Subnet Mask, Gateway respectively. TFTP SERVER ADDRESS Follow Table 3 with voice menu option 06 to configure the IP address of the TFTP server.
CONFIGURING HT-386 WITH WEB BROWSER The HT–386 has an embedded Web server that will respond to HTTP GET/POST requests. It also has embedded HTML pages that allow users to configure the HT–386 through a Web browser such as Microsoft’s IE, AOL’s Netscape or Mozilla Firefox installed on Windows or Unix OS. (Macintosh OS does not included). Access the Web Configuration Menu First, get the IP address of the HT-386 through section “Configuration” with menu option 02.
The password is case sensitive with maximum length of 25 characters. The factory default password for End User and administrator is “123” and “admin” respectively. Only administrator can get access to the “ADVANCED SETTING” configuration page. NOTE: 1. If you CAN NOT log into the configuration page by using default password, please check with the VoIP service provider. Most likely the VoIP service provider has provisioned the device and configured for you therefore the password has already been changed.
TABLE 8: HT-386 BASIC SETTINGS PAGE DEFINITIONS End User Password Web Port IP Address This contains the password for end user to access the Web Configuration Menu. User can put new password here. This field is case sensitive with maximum of 25 characters This is the device’s internal HTTP server port. Default is 80. - If DHCP mode is enabled, then all the field values for the Static IP mode are not used (even though they are still saved in the Flash memory.
PSTN Access Code US/Canada where daylight saving time is applicable: 04,01,7,02,00;10,-1,7,02,00;60 This means the daylight saving time starts from the first Sunday of April at 2AM and ends the last Sunday of October at 2AM. The saving is 60 minutes (1hour). Default is “*00”, user can change it. By pressing the code user can switch the phone to PSTN line connected to the Line port of ATA and make PSTN outgoing calls. This is called PSTN Pass Through.
Onhook Voltage Polarity Reversal NTP server Syslog Server Syslog Level • ETSI-DTMF (Finland, Sweden) • ETSI-FSK (France, Germany, Norway, Taiwan, UK-CCA) Select the onhook voltage to suit different area or PBX. Default is No. If set to Yes, polarity will be reversed upon call establishment and termination. URI or IP address of the NTP (Network Time Protocol) server, which the HT386 will use to synchronize the date/time.
TABLE 10: HT-386 FXS PORT1/FXS PORT2 SETTINGS PAGES DEFINITIONS SIP Server Outbound Proxy SIP User ID Authenticate ID Authentication Password Name Use DNS SRV: User ID is Phone Number SIP Registration Unregister On Reboot Register Expiration Local SIP port Local RTP port Use Random Port DTMF Payload Type Send DTMF Grandstream Networks, Inc. This field contains the URI string or the IP address (and port, if different from 5060) of the SIP proxy server. e.g.
Send Flash Event Enable Call Features Use Bell-style 3-way Conference Offhook Auto-Dial Proxy-Require Disable Call Waiting NAT Traversal Preferred Vocoder Voice Frames per TX G723 Rate: iLBC frame size: iLBC payload type: Grandstream Networks, Inc. INFO. Default is NO. If set to yes, flash will be sent as DTMF event. Default is Yes. Advance call features and feature codes functions are supported locally If this parameter is set to “Yes”, user will be able to make Bellcore style 3-way conference.
127. This controls the silence suppression/VAD feature of G723 and G729. If set to Silence Suppression “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. T.38 (Auto Detect) FoIP by default, or Pass-Through (must use codec Fax Mode PCMU/PCMA) Default is No.
TABLE 11: HT-386 CALL PROGRESS TONES SETTINGS PAGE DEFINITIONS Call Progress Tones Using these settings, user can configure ring or tone frequencies according to their preference. By default they 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.
FIGURE 5: SCREENSHOT OF REBOOTING SCREEN Grandstream Device Configuration The device is rebooting now... You may relogin by clicking on the link below in 30 seconds. Click to relogin All Rights Reserved Grandstream Networks, Inc. 2004 CONFIGURATION THROUGH A CENTRAL SERVER User can automatically configure the HT–386 from a central provisioning system. Download the configuration files via TFTP or HTTP from the central server.
SOFTWARE UPGRADE Software upgrades are performed via TFTP or HTTP. The corresponding configuration settings are in the ADVANCED SETTINGS configuration page. FIRMWARE UPGRADE THROUGH TFTP/HTTP Our latest official release can be downloaded from: http://www.grandstream.com/y-firmware.htm. To upgrade your unit firmware, follow these steps: 1. Under Advanced Settings webpage, enter your TFTP or HTTP Server IP address (or FQDN) next to the “Firmware Upgrade: Upgrade Server” field. 2.
TFTP Server Downloading Directions: 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 HT–386 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.
RESTORE FACTORY DEFAULT SETTING WARNING! 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. There are two ways to reset the device. RESET VIA THE RESET BUTTON 1.
GLOSSARY OF TERMS ADSL Asymmetric Digital Subscriber Line: Modems attached to twisted pair copper wiring that transmit from 1.5 Mbps to 9 Mbps downstream (to the subscriber) and from 16 kbps to 800 kbps upstream, depending on line distance. AGC Automatic Gain Control is an electronic system found in many types of devices. Its purpose is to control the gain of a system in order to maintain some measure of performance over a changing range of real world conditions.
FQDN Fully Qualified Domain Name. A FQDN consists of a host and domain name, including top-level domain. For example, www.grandstream.com is a fully qualified domain name. www is the host, Grandstream is the second-level domain, and and.com is the top level domain. FXS Foreign eXchange Office. An FXS device can be an analog phone, answering machine, fax, or anything that handles a call from the telephone company like AT&T. They should also operate the same way when connected to an FXS interface.
IVR IVR is a software application that accepts a combination of voice telephone input and touch-tone keypad selection and provides appropriate responses in the form of voice, fax, callback, e-mail and perhaps other media. MTU A Maximum Transmission Unit (MTU) is the largest size packet or frame, specified in octets (eight-bit bytes), that can be sent in a packet- or frame-based network such as the Internet. The maximum for Ethernet is 1500 byte.
TCP Transmission Control Protocol is one of the core protocols of the Internet protocol suite. Using TCP, applications on networked hosts can create connections to one another, over which they can exchange data or packets. The protocol guarantees reliable and in-order delivery of sender to receiver data. TFTP Trivial File Transfer Protocol, is a very simple file transfer protocol, with the functionality of a very basic form of FTP; It uses UDP (port 69) as its transport protocol.
